2012-03-15 Asterisk Development Team * Asterisk 1.6.2.23 Released. * AST-2012-002 2012-03-15 18:32 +0000 [r359645] Matthew Jordan * apps/app_milliwatt.c, /: Fix remotely exploitable stack overrun in Milliwatt Milliwatt is vulnerable to a remotely exploitable stack overrun when using the 'o' option. This occurs due to the milliwatt_generate function not accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of samples it can put in the output buffer. This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET when determining the maximum number of samples allowed. Note that at no point is remote code execution possible. The data that is written into the buffer is the pre-defined Milliwatt data, and not custom data. (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283) Note that this patch was written by Russell, even though Matt uploaded it 2011-12-19 Asterisk Development Team * Asterisk 1.6.2.22 Released 2011-12-18 18:25 +0000 [r348515] Kevin P. Fleming * configs/sip.conf.sample: Correct two flaws in sip.conf.sample related to AST-2011-013. * The sample file listed *two* values for the 'nat' option as being the default. Only 'yes' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. 2011-12-08 Asterisk Development Team * Asterisk 1.6.2.21 Released. * AST-2011-013, AST-2011-014 2011-12-08 21:03 +0000 [r347659] Leif Madsen * /: Update svn:externals to use menuselect from 1.6.2.20 and not later. This change is required because when making security releases, if you pull from menuselect/trunk you'll get changes meant for later versions of Asterisk. 2011-12-08 16:17 +0000 [r347530] Terry Wilson * channels/chan_sip.c: Don't crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. (closes issue ASTERISK-18805) 2011-11-21 20:33 +0000 [r345800-345827] Terry Wilson * channels/chan_sip.c: Don't set the nat default twice. Cleaning up a small merge issue ASTERISK-18862 * configs/sip.conf.sample, CHANGES, /, channels/chan_sip.c: Default to nat=yes; warn when nat in general and peer differ It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 2011-08-05 Leif Madsen * Asterisk 1.6.2.20 Released. 2011-08-01 21:19 +0000 [r330490-330505] Jonathan Rose * main/features.c: fixes reference leak pointed out by rmudgett in https://reviewboard.asterisk.org/r/1337/ * main/features.c: Asterisk 18103 - Fix reload crash caused by destroying default parking lot Default parking lot was being destroyed in reload and was not being rebuilt properly. This patch keeps features.c reload from destroying the default parking lot in 1.6.2. Bug was caused by a hasty backport which didn't test reload enough times to catch the problem. (closes issue ASTERISK-18103) Reported by: 808blogger Review: https://reviewboard.asterisk.org/r/1337/ 2011-07-08 22:26 +0000 [r327255] Jason Parker * cdr, formats, codecs/gsm/src, funcs, bridges, codecs/lpc10, main/db1-ast/btree, codecs/g722, main, main/db1-ast/recno, res, pbx, res/ael, channels, main/stdtime, codecs, agi, utils, main/db1-ast/hash, apps, main/db1-ast/db, main/db1-ast/mpool: Add .o files to svn:ignore property, since it's only ignored if locally configured to do so. 2011-06-28 Leif Madsen * Asterisk 1.6.2.19 Released. 2011-06-28 20:06 +0000 [r325277] Terry Wilson * /, channels/chan_sip.c: Merged revisions 325275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r325275 | twilson | 2011-06-28 15:03:19 -0500 (Tue, 28 Jun 2011) | 2 lines Don't leak SIP username information ........ 2011-06-23 18:21 +0000 [r324643] Kinsey Moore * channels/chan_sip.c: Addresses AST-2011-008, memory corruption and remote crash in SIP driver. AST-2011-008 2011-06-23 18:18 +0000 [r324634] David Vossel * channels/chan_iax2.c, /, main/features.c: Merged revisions 324627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 ........ 2011-06-21 16:10 +0000 [r324306] Kinsey Moore * apps/app_confbridge.c: ConfBridge does not handle hangup properly When playing back a prompt to a channel, confbridge neglects to check for hangup events causing lockup condititions for hangups that occur before actually joining the conference. This change ensures that the user is removed from the conference in the event of a premature hangup. Review: https://reviewboard.asterisk.org/r/1277/ 2011-06-15 18:13 +0000 [r323733] Terry Wilson * /, main/features.c: Merged revisions 323732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes sure that dynamic features are also checked when deciding whether or not to pass DTMF through or store it for interpreting. (closes issue ASTERISK-17914) Reported by: vrban ........ 2011-06-15 15:22 +0000 [r323579] Sean Bright * main/manager.c, /: Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ 2011-06-10 19:15 +0000 [r323039] Matthew Nicholson * channels/chan_sip.c: Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop. (closes issue ASTERISK-17798) tested by mnicholson 2011-06-09 15:37 +0000 [r322668-322699] Matthew Nicholson * channels/chan_sip.c: unlock pvt when we drop voice frames received in early media when in t.38 mode * channels/chan_sip.c: fix for previous commit * /, channels/chan_sip.c: Merged revisions 322646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r322646 | mnicholson | 2011-06-09 10:10:30 -0500 (Thu, 09 Jun 2011) | 5 lines don't drop any voice frames when checking for T.38 during early media (closes issue ASTERISK-17705) Review: https://reviewboard.asterisk.org/r/1186/ patch by oej reported by oej ........ 2011-05-27 08:24 +0000 [r321210] Alec L Davis * main/features.c: Fix *8 directed pickup locks system during pickupsound play out move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2 threads trying to write audio to same channel. In addition fixes choppy audio beep in issue 19177. (issue #18654) (issue #19177) Reported by: Docent Patches: review1232-1.6.2.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1232/ 2011-05-23 16:15 +0000 [r320506-320562] David Vossel * main/tcptls.c: Adds missing part to the ast_tcptls_server_start fails second attempt to bind patch. (closes issue #19289) Reported by: wdoekes Patches: issue19289_delay_old_address_setting_tcptls_2.patch uploaded by wdoekes (license 717) * apps/app_chanspy.c: Fixes chanspy enforced mode lacking a channel_unlock. (closes issue #19348) Reported by: wdoekes Patches: issue19348_chanspy_missing_channel_unlock.patch uploaded by wdoekes (license 717) 2011-05-22 23:25 +0000 [r320444] Tilghman Lesher * res/res_odbc.c: Don't crash when the connection fails. (closes issue #19250) Reported by: seadweller Patches: 20110514__issue19250.diff.txt uploaded by tilghman (license 14) Tested by: seadweller, sum 2011-05-20 21:24 +0000 [r320271] David Vossel * main/tcptls.c: Fixes issue with ast_tcptls_server_start failing on second attempt to bind. (closes issue #19289) Reported by: wdoekes Patches: issue19289_delay_old_address_setting_tcptls.patch uploaded by wdoekes (license 717) 2011-05-20 20:44 +0000 [r320236] Richard Mudgett * /, apps/app_meetme.c: Merged revisions 320235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) | 13 lines The meetme CLI command completion leaves conferences mutex locked. When issuing a meetme kick CLI command and an invalid (non-existent) conference number is specified, pressing Tab leaves the conferences mutex locked and, therefore, all conferences deadlock. Add missing unlock. (closes issue #19336) Reported by: zvision Patches: app_meetme.diff uploaded by zvision (license 798) ........ 2011-05-20 18:45 +0000 [r320179] Matthew Nicholson * channels/chan_sip.c: This commit modifies the way polling is done on TLS sockets. Because of the buffering the TLS layer does, polling is unreliable. If poll is called while there is data waiting to be read in the TLS layer but not at the network layer, the messaging processing engine will not proceed until something else writes data to the socket, which may not occur. This change modifies the logic around TLS sockets to only poll after a failed read on a non-blocking socket. This way we know that there is no data waiting to be read from the buffering layer. (closes issue #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by mnicholson (license 96) Tested by: mnicholson 2011-05-18 23:11 +0000 [r319528-319653] Terry Wilson * /, channels/chan_sip.c: Merged revisions 319652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines Make sure everyone gets an unhold when a transfer succeeds Some phones, like the Snom phones, send a hold to the transfer target after before sending the REFER. We need to make sure that we unhold the parties that are being connected after the masquerade. If Local channels with the /nm option are used when dialing the parties, hold music would still be playing on the transfer target, even after being connected with the transferee. ........ * apps/app_dial.c, /: Merged revisions 319527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines Fix app_dial ring groups Revert part of r315643. We need to remove the datastore here as well. The code in bridging code will catch anything that app_dial might miss. (closes issue #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff uploaded by elguero (license 37) ........ 2011-05-16 18:00 +0000 [r319202] Terry Wilson * channels/chan_sip.c: Unlink a peer from peers_by_ip when expiring a registration Review: https://reviewboard.asterisk.org/r/1218/ 2011-05-16 15:56 +0000 [r319144] David Vossel * channels/chan_sip.c: Fixes issue with peer ref-counting during handle_request_subscribe. (closes issue #19293) Reported by: irroot 2011-05-16 15:51 +0000 [r319141] Matthew Nicholson * channels/chan_sip.c: Make sure tcptls_session exists before dereferencing it. (closes issue #19192) Reported by: stknob Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723) Tested by: vois, Chainsaw 2011-05-13 01:14 +0000 [r318636-318735] Richard Mudgett * include/asterisk/features.h, /, channels/chan_sip.c, apps/app_directed_pickup.c, main/features.c: Merged revisions 318734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r318734 | rmudgett | 2011-05-12 20:09:40 -0500 (Thu, 12 May 2011) | 43 lines Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 * The applicable fixes for v1.4 are the SIP deadlock and the in progress masquerade check for multiple parties trying to pickup the same call. issue18654_v1.4.patch uploaded by rmudgett (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch uploaded by rmudgett (license 664) ........ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ ................ * channels/chan_sip.c: Merged revision 222981 from https://origsvn.digium.com/svn/asterisk/branches/1.8 Similar deadlock possible when running the Pickup application internally. ------------------------------------------------------------------------ r222981 | dvossel | 2009-10-08 17:04:41 -0500 (Thu, 08 Oct 2009) | 13 lines Deadlock between ast_cel_report_event and ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner channel while only the pvt lock is held. Since pbx_exec calls ast_cel_report_event which attempts to lock the channel, invalid locking order occurs. Channels should be locked before pvt's. (closes issue #15512) Reported by: lmsteffan Patches: ast_cel_deadlock_15512.diff uploaded by dvossel (license 671) 2011-05-11 17:15 +0000 [r318548] Terry Wilson * channels/chan_sip.c: Clean up several chan_sip reference leaks Several situations in the code could lead to peers or sip_pvt references being leaked. This would cause RTP ports to never be destroyed (leading to exhaustion of all available RTP ports) and memory leaks. The original patch for this issue from rgagnon was the result of an obscene amount of testing and hard work, for which I am very grateful. I did some cleanup and added a few additional refcount fixes that I found. (closes issue #17255) Reported by: kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ Review: https://reviewboard.asterisk.org/r/1207/ 2011-05-09 20:04 +0000 [r318331] Terry Wilson * channels/chan_sip.c: Don't offer video to directmedia callee unless caller offered it as well Make sure that when directmedia is enabled, that video is not offered to the callee even if it supports it. p->vrtp will not exist since the caller didn't offer video. (closes issue #19195) Reported by: one47 Patches: sip_cant_add_video_rtp uploaded by one47 (license 23) 2011-05-09 16:51 +0000 [r318230] David Vossel * channels/chan_sip.c: Fixes cases where sip_set_rtp_peer can return too early during media path reset. (closes issue #19225) Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) 2011-05-06 19:34 +0000 [r317859] Matthew Nicholson * pbx/pbx_lua.c: pbx_lua autoservice fixes Don't start an autoservice in pbx_lua if pbx_lua already started one and don't stop one if we didn't start one. Also start and stop the autoservice when transferring control from and to the pbx. 2011-05-06 18:03 +0000 [r317720] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 317719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317719 | rmudgett | 2011-05-06 12:59:05 -0500 (Fri, 06 May 2011) | 11 lines Regression after r297603 (Improve handling of REGISTER requests with multiple contact headers.) Uninitialized variable. (issue #18640) (closes issue #18785) Reported by: pnlarsson Patches: issue18785_enegaard.patch uploaded by enegaard (license 1197) ........ 2011-05-06 15:18 +0000 [r317666] Matthew Nicholson * pbx/pbx_lua.c: Add a datastore fixup to fix a pbx_lua crash. (closes issue #19055) Reported by: jamhed Patches: lua_datastore_fixup1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, jamhed 2011-05-06 08:04 +0000 [r317575] Terry Wilson * apps/app_queue.c, /: Merged revisions 317574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines Re-fix queue round-robin This part of the change for r315596 was incorrect. No bridge occurs when doing a roundrobin dial and no one answers, so this code shouldn't have been removed. ........ 2011-05-05 18:29 +0000 [r317255] Russell Bryant * /, channels/chan_sip.c: Merged revisions 317211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines chan_sip: fix broken realtime peer count, fix memory leak This patch addresses two bugs in chan_sip: 1) The count of realtime peers and users was off. The increment checked the value of the caching option, while the decrement did not. 2) Add a missing regfree() for a regex. (closes issue #19108) Reported by: vrban Patches: missing_regfree.patch uploaded by vrban (license 756) sip_object_counter.patch uploaded by vrban (license 756) ........ 2011-05-05 17:59 +0000 [r317195] Matthew Nicholson * channels/chan_sip.c: Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer abruptly disappears. This mostly occurs after a successful registration. (closes issue #17544) Reported by: marcelloceschia Patches: (modified) tcptls.patch uploaded by st (license 907) 2011-05-05 14:56 +0000 [r317103] Leif Madsen * contrib/scripts/safe_asterisk, /: Merged revisions 317102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011) | 8 lines Disable console colourization inside safe_asterisk checks. (closes issue #19213) Reported by: lefoyer Patches: issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by wdoekes (license 717) Tested by: wdoekes, lefoyer ........ 2011-05-04 16:10 +0000 [r316708] Sean Bright * apps/app_voicemail.c, /: Merged revisions 316707 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May 2011) | 8 lines If sox fails when processing a voicemail, don't delete the original file. (closes issue #18111) Reported by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright (license 71) Tested by: seanbright ........ 2011-05-04 14:23 +0000 [r316616-316644] David Vossel * apps/app_chanspy.c: Fixes one-way-audio when chanspy activated with the 'o' option (closes issue #18382) Reported by: jkister Patches: 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt uploaded by malin (license ) Tested by: firstsip, Greenlightcrm, malin, wdoekes, boroda, dvossel * channels/chan_sip.c: Fixes session-timers=refuse not being enforced for *caller* During handle_request_invite, the session timer mode was retrieved from a cached variable. This patch forces a peer lookup of the session timer mode in the case of an incoming invite. (closes issue #18804) Reported by: wdoekes Patches: issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717) issue_18804_v2.diff uploaded by dvossel (license 671) 2011-05-04 02:23 +0000 [r316475] Sean Bright * apps/app_meetme.c: Honor the C option to MeetMe when L is passed. This fixes a case that r304773 and friends missed. (closes issue #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff uploaded by var (license 1227) Tested by: seanbright 2011-05-03 21:29 +0000 [r316329] David Vossel * channels/chan_local.c, /: Merged revisions 316328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011) | 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (closes issue #19053) Reported by: oej Tested by: oej Review: https://reviewboard.asterisk.org/r/1158/ ........ 2011-05-02 19:04 +0000 [r316093] Tilghman Lesher * funcs/func_curl.c: More possible crashes based upon invalid inputs. (closes issue #18161) Reported by: wdoekes Patches: 20110301__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: wdoekes 2011-04-27 19:03 +0000 [r315893] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 315891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2. This change optimizes the free_via() function and removes some redundant null checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using the port specified in the Via header for routing responses (even when maddr is not set). Also the htons() function is now used when setting the port. Additional documentation comments have been added in various places to make the logic in the code clearer. (closes issue #18951) Reported by: jmls Patches: issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified) ........ 2011-04-26 22:52 +0000 [r315643-315672] Terry Wilson * /, channels/chan_sip.c: Merged revisions 315671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) | 11 lines Make sure unregistering a peer unlinks it from the peer container Instead of mostly copying the code from expire_register, just use the function that "does the right thing". (closes issue #16033) Reported by: kkm Patches: 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888) Tested by: kkm, tilghman, twilson ........ * apps/app_queue.c, apps/app_dial.c, /, main/features.c: Merged revisions 315596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ ........ 2011-04-26 19:22 +0000 [r315502] Tilghman Lesher * include/asterisk/select.h, /: Merged revisions 315501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines Fix the bounds-checking code. The code that set the bit within the select bitfield was correct, but the bounds-checking code was not. The change to that line uses the new _bitsize macro for clarity. Also, FD_ZERO macro did not zero-out anything but the first word of the bitfield, so this could have caused problems with modules using that macro with the expanded bitfield. (closes issue #18773) Reported by: jamicque Patches: 20110423__issue18773.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac ........ 2011-04-26 02:17 +0000 [r315393] Paul Belanger * pbx/pbx_config.c: Add back CLI command 'dialplan save' (closes issue #19140) Reported by: lmadsen Patches: __20110419_dialplan_save.patch.txt uploaded by lmadsen (license 10) 2011-04-25 19:31 +0000 [r315212-315258] Russell Bryant * /, formats/format_wav.c: Merged revisions 315257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011) | 10 lines Be more flexible with unknown chunks in wav files. This patch makes format_wav ignore unknown chunks instead of erroring out on them. (closes issue #18306) Reported by: jhirsch Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch (license 1156) ........ * channels/chan_sip.c: Don't link non-cached realtime peers into the peers_by_ip container. (closes issue #18924) Reported by: wdoekes Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded by wdoekes (license 717) 2011-04-25 07:11 +0000 [r315052] Alec L Davis * channels/chan_local.c, /: Merged revisions 315051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr 2011) | 11 lines chan_local:check_bridge() misplaced misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't followed, brigde remains locked. (closes issue #19176) Reported by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis (license 585) ........ 2011-04-22 20:49 +0000 [r314958] Matthew Nicholson * /, channels/chan_agent.c: Merged revisions 311203,314908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar 2011) | 4 lines Don't hold the pvt lock while streaming a file. ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48 -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and the app threads from using the asterisk channel at the same time. ABE-2756 ........ 2011-04-22 14:35 +0000 [r314776-314823] Russell Bryant * /: Recorded merge of revisions 314822 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314822 | russell | 2011-04-22 09:34:23 -0500 (Fri, 22 Apr 2011) | 11 lines Initialize buffers in getvar and getvarfull. Initialize the buffers used to hold the result from GET VARIABLE or GET VARIABLE FULL. The bug report shows func_read returning garbage in the result. It assumed that the buffer passed in was initialized, like many other functions do. In the more common code path (through the dialplan), it is initialized, so just initialize it here too. (closes issue #19050) Reported by: johnz ........ * res/res_agi.c: Initialize buffers in getvar and getvarfull. Initialize the buffers used to hold the result from GET VARIABLE or GET VARIABLE FULL. The bug report shows func_read returning garbage in the result. It assumed that the buffer passed in was initialized, like many other functions do. In the more common code path (through the dialplan), it is initialized, so just initialize it here too. (closes issue #19050) Reported by: johnz * main/features.c: Fix handling of some call parking config options. This patch adjusts the handling of some call parking config options to fix some issues that have already been addressed in 1.8 and trunk. (closes issue #19167) Reported by: bluecrow76 Patches: asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff uploaded by bluecrow76 (license 270) 2011-04-21 18:22 +0000 [r314620] Matthew Nicholson * configs/sip.conf.sample, configs/skinny.conf.sample, configs/http.conf.sample, main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c, main/http.c: Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ 2011-04-21 00:17 +0000 [r314549] Terry Wilson * channels/chan_sip.c: Don't allocate more space than necessary for a sip_pkt This extra allocation is a hold-over from when pkt->data was a character array. Now that it is an allocated string, just allocate enough for the sip_pkt. 2011-04-19 14:27 +0000 [r314202-314205] Leif Madsen * funcs/func_channel.c: Remove duplicate documentation from func_channel.c (closes issue #18970) Reported by: IgorG Patches: func_channel.c.doc.diff uploaded by IgorG (license 20) * apps/app_dial.c: Update seconds to milliseconds in ast_verb output. (closes issue #19084) Reported by: smurfix Patches: app_dial.patch uploaded by smurfix (license 547) Tested by: lmadsen, smurfix 2011-04-15 14:58 +0000 [r313859] Jonathan Rose * main/cli.c: Fix a Tab Completion bug that occurs due to multiple matches on a substring. Makes word_match function in cli.c repeat a search for a command string until a proper match is found or the string is searched to the last point. (closes issue #17494) Reported by: ffossard Review: https://reviewboard.asterisk.org/r/1180/ 2011-04-13 16:29 +0000 [r313579] Richard Mudgett * main/channel.c, /, res/res_agi.c: Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ 2011-04-12 18:44 +0000 [r313432-313435] Jonathan Rose * channels/chan_dahdi.c: fixing stupid mistake with putting code before variable declaration ........ Merged revisions 313433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines reload Chan_dahdi memory leak caused by variables chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would stay in the dahdi_pvt structs for individual channels (causing them to just continue adding the new ones to the list) and also there was a memory leak causes by the conf objects. This patch resolves both of these by using ast_variables_destroy during the loading process. (closes issue #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by jrose (license 1225) Tested by: tilghman, jrose Review: https://reviewboard.asterisk.org/r/1170/ ........ ........ * channels/chan_dahdi.c: white space change ........ reload Chan_dahdi memory leak caused by variables chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would stay in the dahdi_pvt structs for individual channels (causing them to just continue adding the new ones to the list) and also there was a memory leak causes by the conf objects. This patch resolves both of these by using ast_variables_destroy during the loading process. (closes issue #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by jrose (license 1225) Tested by: tilghman, jrose Review: https://reviewboard.asterisk.org/r/1170/ ........ * channels/chan_dahdi.c: fixes reload Chan_dahdi memory leak caused by variables chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would stay in the dahdi_pvt structs for individual channels (causing them to just continue adding the new ones to the list) and also there was a memory leak causes by the conf objects. This patch resolves both of these by using ast_variables_destroy during the loading process. (closes issue #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by jrose (license 1225) Tested by: tilghman, jrose Review: https://reviewboard.asterisk.org/r/1170/ 2011-04-11 19:33 +0000 [r313278] Leif Madsen * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 313277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011) | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093) Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by tzafrir (license 46) ........ 2011-04-11 15:32 +0000 [r313189] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 313188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) | 25 lines Stuck channel using FEATD_MF if caller hangs up at the right time. The cause was actually a caller hanging up just at the end of the Feature Group D DTMF tones that setup the call. The reason for this is a "guard timer" that's implemented using ast_safe_sleep(100). If the caller happens to hang up AFTER the final tone of the DTMF string but BEFORE the end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero. This causes the code to bounce to the end of ss_thread(), but it does NOT tear down the call properly. This should be a rare occurrence because the caller has to hang up at EXACTLY the right time. Nonetheless, it was happening quite regularly on the reporter's system. It's not easily reproducible, unless you purposely increase the guard-time to 2000 or more. Once you do that, you can reproduce it every time by watching the DTMF debug and hanging up just as it ends. Simply add an ast_hangup() before goto quit. (closes issue #15671) Reported by: jcromes Patches: issue15671.patch uploaded by pabelanger (license 224) Tested by: jcromes ........ 2011-04-07 13:23 +0000 [r313047] Jonathan Rose * main/features.c: Makes parking lots clear and rebuild properly when features reload is invoked from CLI Before, default parkinglot in context parkedcalls with ext 700 would always be present and when reload was invoked, the previous parkinglots would not be cleared. (closes issue #18801) Reported by: mickecarlsson Review: https://reviewboard.asterisk.org/r/1161/ 2011-04-07 10:26 +0000 [r313004] Alec L Davis * apps/app_voicemail.c: app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICE 2011-04-05 14:13 +0000 [r312764] Matthew Nicholson * main/manager.c, /, configs/manager.conf.sample: Merged revisions 312761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) | 8 lines Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson ........ 2011-04-05 14:11 +0000 [r312762] Jonathan Rose * apps/app_meetme.c: Backporting trunk change to add verbosity to 'L' option in meetme 2011-04-04 16:00 +0000 [r312574] Richard Mudgett * channels/chan_dahdi.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 312573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines Issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed. (The same for PROGRESS.) The call was also not hungup if the new B channel is not provisioned or is busy. * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are using the correct structure and B channel. If there is any problem with the operations then the call is now hungup with an appropriate cause code. * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the correct structure by looking for the call and not using the channel ID. NOTIFY is an exception with versions of libpri before v1.4.11 because a call pointer is not available for Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct structure by looking for the call and not using the channel ID. (closes issue #18313) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) ........ 2011-04-01 10:51 +0000 [r312287] Tilghman Lesher * main/asterisk.c, include/asterisk/select.h, /: Merged revisions 312285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman (license 14) ........ 2011-04-01 09:16 +0000 [r312103-312213] Alec L Davis * apps/app_voicemail.c: fix up bad merge46 extra 2 yuck: labels * apps/app_voicemail.c, /: Merged revisions 312174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines voicemail: get real last_message_index and count_messages, ODBC resequence change last_message_index to read the max msgnum stored in the database change count_messages to actually count the number of messages. last_message_index change: This fixed overwriting of the last message if msgnum=0 was missing. Previously every incoming message would overwrite msgnum=1. count_messages change: allows us to detect when requencing is required in opneA_mailbox. resequence enabled for ODBC storage: Assists with fixing up corrupt databases with gaps, but only when a user actively opens there mailboxes. (closes issue #18692,#18582,#19032) Reported by: elguero Patches: based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37) Tested by: elguero, nivek, alecdavis Review: https://reviewboard.asterisk.org/r/1153/ ........ * apps/app_voicemail.c, /: Merged revisions 312070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines app_voicemail: close_mailbox needs to respect additional messages while mailbox is open. close_mailbox leave gaps in message sequence if messages are deleted and new messages arrive during this time, this is because the shuffle down to slot 0, only shuffles the number of pre-existing messages when mailbox is opened, ignoring new arrivals. Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals. Happens on filebased or ODBC storage. (issues #19032,#18582,#18692,#18998) Reported by: alecdavis,tootai,afosorio Review: https://reviewboard.asterisk.org/r/1153/ ........ 2011-03-29 13:17 +0000 [r311844] Jonathan Rose * main/features.c: When comebacktoorigin=no, Asterisk no longer tries to dial extension @parkedcalltimeout and instead dials s without going through fallback. (closes issue #18650) Reported by: davidw Patches: patch.diff uploaded by jrose (license 1225) https://reviewboard.asterisk.org/r/1150/ 2011-03-22 15:24 +0000 [r311496] David Vossel * apps/app_meetme.c: Fixes memory leak in MeetMe AMI action 2011-03-17 14:58 +0000 [r311140] Matthew Nicholson * main/manager.c: Don't write items to the manager socket twice. AST-2011-003 (closes issue 0018987) Reported by: ks-steven 2011-03-17 10:45 +0000 [r311049] Alec L Davis * /, configs/indications.conf.sample: Merged revisions 311048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar 2011) | 12 lines Remove extra quote in indications.conf Picking low hanging fruit. (closes issue #18971) Reported by: IgorG Patches: based on indications.conf.sample.diff uploaded by IgorG (license 20) Tested by: IgorG ........ 2011-03-16 19:46 +0000 [r310889-310998] Terry Wilson * main/tcptls.c: Fix crash on fdopen failure See security advisory AST-2011-004 (closes issue #18845) Reported by: cmaj Patches: patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt uploaded by cmaj (license 830) patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt uploaded by cmaj (license 830) Tested by: cmaj, twilson * main/tcptls.c, main/manager.c: Revert patch with accidental reversion of a previous patch * main/tcptls.c, main/manager.c: Fix crash on fdopen failure See security advisory AST-2011-004 (closes issue #18845) Reported by: cmaj Patches: patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt uploaded by cmaj (license 830) patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt uploaded by cmaj (license 830) Tested by: cmaj, twilson * main/manager.c: Don't keep trying to write to a closed connection See security advisory AST-2011-003. * /, main/features.c: Merged revisions 310888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines Don't delay DTMF in core bridge while listening for DTMF features This patch is mostly the work of Olle Johansson. I did some cleanup and added the silence generating code if transmit_silence is set. When a channel listens for DTMF in the core bridge, the outbound DTMF is not sent until we have received DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds. Some products see this delay and the time skew on RTP packets that results and start ignoring the audio that is sent afterward. With this change, the DTMF_BEGIN frame is inspected and checked. If it matches a feature code, we wait for DTMF_END and activate the feature as before. If transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a multi-digit feature. If it doesn't match a feature, the frame is forwarded along with the DTMF_END without delay. By doing it this way, DTMF is not delayed. (closes issue #15642) Reported by: jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396) Tested by: globalnetinc, jde (closes issue #16625) Reported by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ Review: https://reviewboard.asterisk.org/r/1125/ ........ 2011-03-15 00:31 +0000 [r310780] Alec L Davis * /, main/utils.c: Merged revisions 310779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310779 | alecdavis | 2011-03-15 13:26:09 +1300 (Tue, 15 Mar 2011) | 10 lines core show locks: display ThreadID in hexadecimal Allow easier cross referencing of thread ID's with GDB backtraces (closes issue #18968) Reported by: alecdavis Patches: bug18968.diff.txt uploaded by alecdavis (license 585) ........ 2011-03-14 16:47 +0000 [r310635] Richard Mudgett * /, main/callerid.c: Merged revisions 310633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and TDM410 The last character in the caller id message is getting a framing error. The checksum is the last character in the message. A framing error in the checksum could be because: 1) The sender did not send a full stop bit. 2) The sender cut off the FSK carrier too soon. 3) The sender opted to send zero of the specified zero to 10 trailing mark bits and round-off errors in the code resulted in the code not being where it thought it was in the demodulated bit stream. Bit 8 of 'b' is set when parity error. Bit 9 of 'b' is set when framing error. Made ignore the framing and parity error bits if the errored character is the checksum. We can tolerate a framing/parity error there. The checksum character validates the message. (closes issue #18474) Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek (license 636) (with modifications) Tested by: nivek ........ 2011-03-14 13:56 +0000 [r310585] Jonathan Rose * funcs/func_volume.c: Adds 'p' as an option to func_volume. When it is on, the old behavior with DTMF controlling volume adjustment will be enforced. When it is off, DTMF will not be processed by the function. Programmed by Jonathan Rose Reviewed by David Vossel, Leif Madsen, and Russell Bryant http://reviewboard.digium.internal/r/93/ 2011-03-12 20:24 +0000 [r310414-310448] Tilghman Lesher * /, pbx/pbx_ael.c: Recorded merge of revisions 310435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011) | 31 lines Add AELSub, which provides a stable entry point into AEL subroutines. This commit needs some explanation, given that we're adding a new application into an existing release branch. This is generally a violation of our release policy, except in very limited circumstances, and I believe this is one of those circumstances. The problem that this solves is one of the sanity of using multiple dialplan languages to define a dialplan. In the case of the reporter, he or she is using AEL is define subroutines, while using Realtime extensions to invoke those subroutines. While you can do this, it's based upon the reality of AEL using actual dialplan extensions; however, there is no guarantee that the details of _how_ AEL is compiled into extensions will remain stable. In fact, at the time of this commit, it has already changed twice, once in a fundamental way. Now normally, a new application would only be added to trunk. However, this application is explicitly to create a stable user-level API between versions, and adding it to trunk only will not solve the user's problem of switching between 1.6.2 and 1.8, nor will it help anybody switching from 1.8 to 1.10. Therefore, it needs to go into existing release branches. For the sake of consistency, and also because one of the changes was between 1.4 and 1.6.x, I am also electing to commit this to 1.4. (closes issue #18910) Reported by: alexandrekeller Patches: 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman (license 14) 20110304__issue18919__1.4.diff.txt uploaded by tilghman (license 14) Tested by: alexandrekeller ........ * funcs/func_odbc.c: Transactional handles should be used for the insertbuf, if available. Also, fix a possible resource leak. (closes issue #18943) Reported by: irroot 2011-03-10 05:51 +0000 [r310141] Tilghman Lesher * apps/app_voicemail.c, res/res_config_odbc.c, /, funcs/func_odbc.c: Merged revisions 310140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. (closes issue #18295) Reported by: pruiz ........ 2011-03-07 22:04 +0000 [r309857] Jonathan Rose * apps/app_mixmonitor.c, /: Merged revisions 309856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | 8 lines Bug fix for MixMonitor involving filenames with '.' not in the extension Closes issue #18391) Reported by: pabelanger Patches: bugfix.patch uploaded by jrose (license 1225) Tested by: jrose ........ 2011-03-05 10:28 +0000 [r309677] Tilghman Lesher * main/asterisk.c: Missed part of the conversion when we started passing ppid to astcanary. (closes issue #18850) Reported by: viraptor Patches: canary_ppid.patch uploaded by viraptor (license 543) 2011-03-04 19:37 +0000 [r309494-309584] Matthew Nicholson * pbx/pbx_lua.c: Restore mysterious lua_pushvalue() call removed in r309494. The mystery has been solved. * pbx/pbx_lua.c: Check for errors from fseek() when loading config file, properly abort on errors from fread(), and supply a traceback for errors generated when loading the config file. Also, prepend a newline to traceback output so that the main error message is on it's own line. * pbx/pbx_lua.c: remove mysterious lua_pushvalue() that is never used 2011-03-04 00:42 +0000 [r309356] David Ruggles * apps/app_externalivr.c, /: Merged revisions 309355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines fix small memory leak fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) ........ 2011-03-03 20:13 +0000 [r309348] Leif Madsen * apps/app_directed_pickup.c: Update PickupChan documentation. The PickupChan uses the ampersand as the argument separator. (closes issue #18905) Reported by: vmikhnevych Tested by: vmikhnevych 2011-03-02 19:53 +0000 [r309255] Jason Parker * channels/chan_sip.c: Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP. Since it's a duplicate, nothing is going to be done, so delme doesn't need to be set at all. Strangely, when this was added, this was being set to 1 in 1.6, and 0 in trunk. (issue AST-439) 2011-03-02 01:06 +0000 [r309251] Tilghman Lesher * main/ast_expr2.fl, configure, include/asterisk/autoconfig.h.in, main/ast_expr2f.c, configure.ac: Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. 2011-03-01 16:05 +0000 [r309083] David Vossel * channels/chan_sip.c: Fixes thread blocking issue in the sip TCP/TLS implementation. (closes issue #18497) Reported by: vois Patches: issues_18497.diff uploaded by dvossel (license 671) Tested by: vois, rossbeer, kowalma, Freddi_Fonet 2011-02-28 11:07 +0000 [r308990-309034] Tilghman Lesher * configure, configure.ac: Clarify meaning, removing double negative (stupid!) * main/ast_expr2.fl, configure, include/asterisk/autoconfig.h.in, main/ast_expr2f.c, configure.ac: A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error. Detect whether Flex will compile without the workaround; if so, suppress our workaround code. * funcs/func_odbc.c: Statements updating zero rows may return SQL_NO_DATA. This is fine; it's handled. (closes issue #18815) Reported by: irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot (license 52) 2011-02-24 17:54 +0000 [r308814] Terry Wilson * main/manager.c, /: Merged revisions 308813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines Don't broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (closes issue #18168) Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre (license 1142) Tested by: FeyFre, twilson ........ 2011-02-24 14:59 +0000 [r308722] Matthew Nicholson * main/udptl.c, /: Merged revisions 308721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........ 2011-02-24 03:38 +0000 [r308677-308678] Terry Wilson * configs/sip.conf.sample, channels/chan_sip.c: Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ * include/asterisk/features.h, apps/app_dial.c, main/features.c: Merge missing bugfix for issue #11583 This is the combination of two commits that made it into 1.4, 1.6.0, 1.6.1, and trunk (and therefor 1.8) but that was missed for 1.6.2. ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ 2011-04-25 Leif Madsen * Asterisk 1.6.2.18 Released. * AST-2011-005, AST-2011-006 2011-02-23 Leif Madsen * Asterisk 1.6.2.18-rc1 Released. 2011-02-22 15:37 +0000 [r308528] Andrew Latham * main/http.c: Add HTTP URI log, use ast_debug for console logging Guessed the log levels based on info that level 3 is the soft roof. Can we create a page / document to define the levels? 2011-02-21 15:00 +0000 [r308414] Matthew Nicholson * main/udptl.c, /: Merged revisions 308413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway. AST-2011-002 FAX-281 ........ 2011-02-19 14:03 +0000 [r308329] Andrew Latham * main/http.c: Add CSS MIME Type Modern browsers are checking for the MIME Type of pages and in some cases will not load a file if the type is wrong. 2011-02-15 23:33 +0000 [r308007] Jason Parker * apps/app_queue.c, /: Merged revisions 308002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban ........ 2011-02-15 07:01 +0000 [r307792-307836] Tilghman Lesher * funcs/func_odbc.c: Need to retrieve the rows affected before using the associated variable. (closes issue #18795) Reported by: irroot Patches: 20110211__issue18795.diff.txt uploaded by tilghman (license 14) Tested by: tilghman * res/res_odbc.c: Increment usage count at first reference, to avoid a race condition with many threads creating connections all at once. (issue #18156) Reported by: asgaroth Patches: 20110214__issue18156.diff.txt uploaded by tilghman (license 14) Tested by: tilghman 2011-02-11 01:02 +0000 [r307624] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 307623 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307623 | rmudgett | 2011-02-10 18:29:17 -0600 (Thu, 10 Feb 2011) | 13 lines Reentrancy problem if outgoing call gets different B channel than requested. The chan_dahdi pri_fixup_principle() routine needs to protect the Asterisk channel with the channel lock when it changes the technology private pointer to a new private structure. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. Partial backport from v1.8 -r300714. ........ 2011-02-10 22:35 +0000 [r307535] Jason Parker * main/asterisk.c, contrib/init.d/rc.debian.asterisk, /: Merged revisions 307534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines Remove color when executing commands via a remote console. Essentially this makes '-x' imply '-n' on rasterisk. This was done in a different and incomplete way previously, which I'm reverting here. (issue #18776) Reported by: alecdavis ........ 2011-02-09 21:48 +0000 [r307316] Andrew Latham * contrib/init.d/rc.debian.asterisk: Disable color during running test (closes issue #18776) Reported by: alecdavis Patches: ast_deb_init.diff uploaded by lathama (license 1028) Tested by: andrel, lathama 2011-02-09 19:52 +0000 [r307227] Jeff Peeler * main/features.c: Make sure to set parking dial context for non-default parking lots. Since parking_con_dial isn't settable, set all parking lots to "park-dial". (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270) modified by me 2011-02-08 20:14 +0000 [r306973] Terry Wilson * /, channels/chan_sip.c: Merged revisions 306972 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with pedantic=yes ........ 2011-02-08 19:41 +0000 [r306865-306966] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 306965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) | 1 line fix this line again ........ * apps/app_voicemail.c, /: Merged revisions 306960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines Backup file storing message duration is not used with IMAP_STORAGE, remove code. The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil ........ * apps/app_voicemail.c, /: Merged revisions 306864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) | 1 line make this safer and fully correct, pointed out by Steve Davis ........ 2011-02-07 22:40 +0000 [r306618-306673] Terry Wilson * /, main/features.c: Merged revisions 306672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) | 10 lines Don't try to pickup a call in the middle of a masquerade If A calls B which doesn't answer and C & D both try to do a call pickup, it is possible for ast_pickup_call to answer the call, then fail to masquerade one of the calls because the other one is already in the process of masquerading. This patch checks to see if the channel is in the process of masquerading before call before selecting it for a pickup. Review: https://reviewboard.asterisk.org/r/1094/ ........ * /, channels/chan_sip.c: Merged revisions 306617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines Don't allow a REFER w/replaces to replace its own dialog Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ ........ 2011-02-04 19:21 +0000 [r306346] Jason Parker * apps/app_queue.c: Don't fallthrough to 'unknown' in the 'ringing' case. This could cause improper exits from the queue. (closes issue #18499) Reported by: zaltar Patches: app_queue.patch uploaded by zaltar (license 1148) 2011-02-03 20:56 +0000 [r306126] Terry Wilson * channels/chan_local.c, /: Merged revisions 306119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines Set hangup cause in local_hangup When a call involves a local channel (like SIP -> Local -> SIP), the hangup cause was not being set. This resulted in SIP channels sometimes getting a 503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+ this also can cause issues with CCSS that involve a local channel. This patch sets the hangupcause for one side of the local channel to the other in local_hangup for outbound calls. ........ 2011-02-03 20:49 +0000 [r306123] Jeff Peeler * main/features.c: Set exception on channel in parking thread when POLLPRI event detected. This is done just to make the code be equivalent to the old select code. As noted in 303106 the same issue was already fixed in this branch, but the exception was not set on the channel in the case of POLLPRI. The reason that this did not cause a problem here is because in 122923 the check in __ast_read to check the exception flag was removed. (related to #18637) 2011-02-03 15:41 +0000 [r305985] Andrew Latham * phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample: res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support (issue #18713) Reported by: lathama Patches: snom_dir.diff uploaded by lathama (license 1028) Tested by: lathama 2011-02-03 00:15 +0000 [r305889] Richard Mudgett * main/channel.c, main/manager.c, /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ 2011-02-02 14:40 +0000 [r305648-305752] Andrew Latham * channels/chan_sip.c: Replace link to old doc with new wiki page. Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions * configs/sip.conf.sample: SIP Configuration Documentation sip show settings reports qualifyfreq in milliseconds. sip.conf configures qualifyfreg in seconds. 2011-02-01 17:02 +0000 [r305472] Jason Parker * res/res_musiconhold.c, /: Merged revisions 305471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) | 9 lines Close file descriptor for timing source when a MOH class gets destroyed. (closes issue #18457) Reported by: mcallist Patches: 18457-closetimer.diff uploaded by qwell (license 4) 18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested by: qwell, loloski ........ 2011-01-31 23:50 +0000 [r305342] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 305341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) | 7 lines Obtain the pri lock for PRI queue counters. Need to obtain the pri lock when calling pri_dump_info_str() to avoid a reentrancy problem when calculating the Q.921 Q count statistic. JIRA AST-484 ........ 2011-01-31 22:59 +0000 [r305130-305253] Jason Parker * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) ........ * res/res_musiconhold.c, /: Merged revisions 305129 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan 2011) | 2 lines Set file descriptors to -1 on creation, so that we don't see weirdness later. ........ 2011-01-31 13:52 +0000 [r305082] Andrew Latham * main/http.c: Asterisk HTTP response Content-type Address content type for BSD and other platforms (closes issue #18456) Reported by: alexo Patches: asterisk18_http.patch uploaded by alexo (license 1175) Tested by: alexo 2011-01-31 07:25 +0000 [r304978] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 304952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31 Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined. ........ 2011-01-29 23:05 +0000 [r304659-304865] Sean Bright * res/res_config_ldap.c: Plug some memory leaks in the LDAP realtime driver. (closes issue #18435) Reported by: zaltar Patches: res_config_ldap.patch uploaded by zaltar (license 1148) * apps/app_meetme.c: If we fail to allocate our announcement objects, make sure we don't leak objects. The majority of this patch was committed already in r304726 and r304729. (issue #18225) Reported by: kenji (issue #18444) Reported by: junky (closes issue #18343) Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz (license 834) * apps/app_meetme.c: When we pass the S() or L() options to MeetMe, make sure that we honor C as well. Without this patch, if the user was kicked from the conference via the S() or L() mechanism, we would just hang up on them even if we also passed C (continue in dialplan when kicked). With this patch we honor the C flag in those cases. (closes issue #17317) Reported by: var * apps/app_meetme.c: Make sure that we unref the correct object when ejecting the most recent caller. Currently, when we kick the last user to enter, we decrement our own reference count which results in a crash when we kick another user or when we exit the conference ourselves. This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in 1.6.2. (closes issue #18225) Reported by: kenji Patches: issue18225.patch uploaded by seanbright (license 71) Tested by: seanbright * apps/app_meetme.c: Fix user reference leak in MeetMe. We were unlinking the user from the conferences user container, but not decrementing the reference count of the user as well, resulting in a leak. (closes issue #18444) Reported by: junky Tested by: seanbright * apps/app_meetme.c: Revert part of the previous commit that snuck in. * apps/app_meetme.c: Don't leak references if we can't create a pseudo channel for mixing in MeetMe. If there was a problem allocating a pseudo channel when building our meetme, we weren't destroying our user container or destroying the mutexes that we created. 2011-01-27 17:01 +0000 [r304461-304465] Jason Parker * /, configure, configure.ac: Merged revisions 304464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) | 9 lines Fix default prefix=/usr regression on non-Linux systems. This partially reverts a change made in branches/1.4/ r267759, which will cause issue #17013 to be reopened. This issue was pointed out by a user on #asterisk, who helpfully discovered that paths were being set incorrectly. To truly understand what was wrong, one should run: svn diff --force -c configure ........ * /, configure: Merged revisions 304460 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan 2011) | 1 line Rerun bootstrap.sh with no changes, so that it is more obvious what my next commit changes. ........ 2011-01-26 22:26 +0000 [r304338] Jeff Peeler * main/features.c: Change delimiter used internally for GOTO_ON_BLINDXFR to commas to match 76703. 2011-01-26 21:02 +0000 [r304250] Mark Michelson * main/udptl.c, /: Merged revisions 304242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed, 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in ast_udptl ........ 2011-01-26 21:01 +0000 [r304244-304249] Matthew Nicholson * /: Merged revisions 304247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304247 | mnicholson | 2011-01-26 15:00:15 -0600 (Wed, 26 Jan 2011) | 2 lines Convert from network to host byte ordering before checking if an IP is a multicast address. ........ * /, channels/chan_sip.c: Merged revisions 304241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ 2011-01-26 20:22 +0000 [r304181] Sean Bright * /, configs/queues.conf.sample: Merged revisions 304159 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed, 26 Jan 2011) | 1 line Make sure the sample queues.conf is properly commented. ........ 2011-01-26 19:38 +0000 [r304149] Richard Mudgett * channels/chan_dahdi.c: Merged revisions 304148 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) | 2 lines Update documentation for DAHDISendCallreroutingFacility() application. .......... 2011-01-26 01:24 +0000 [r304096] Sean Bright * main/file.c: Per the man page, setvbuf() must be called before any other operation on an open file. We use setvbuf() to associate a buffer with a stream, but we have already written to the open file. This works (by chance) on Linux, but fails on other platforms, such as OpenSolaris. (closes issue #16610) Reported by: bklang Patches: setvbuf.patch uploaded by crjw (license 963) Tested by: bklang, asgaroth, efutch 2011-01-25 23:25 +0000 [r304006] Richard Mudgett * /, main/features.c: Merged revisions 304005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) | 8 lines DTMF attended transfers sometimes fail for no apparent reason. The loop in feature_request_and_dial() can exit when Party C has answered without processing an AST_CONTROL_ANSWER. Also sometimes an AST_CONTROL_ANSWER never happens even though Party C has answered. Don't hangup Party C if he is up or we receive an AST_CONTROL_ANSWER. ........ 2011-01-25 22:02 +0000 [r303960] Terry Wilson * /, channels/chan_sip.c: Merged revisions 303906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines Guard against retransmitting BYEs indefinitely In the case of an attended transfer (A calls B, A atxfers to C) where A becomes unreachable before replying to Asterisk's BYE, Asterisk can sometimes retransmit the BYE indefinitely. This is because __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out, it will not ever be marked as ALREADYGONE, so when __sip_autodestruct is called again, we end up starting the cycle over. This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt in the case of a BYE that has timed out. This should prevent Asterisk from trying to transmit new BYE messages in the future. Review: https://reviewboard.asterisk.org/r/1077/ ........ 2011-01-25 18:41 +0000 [r303858] Tilghman Lesher * channels/chan_sip.c: Fix "sip show user ", so that it actually shows results, instead of just completing the last entry. (closes issue #16675) Reported by: pj 2011-01-25 17:42 +0000 [r303769] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ 2011-01-25 16:59 +0000 [r303677] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 303676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines Fix voicemail sequencing for file based storage. A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) ........ 2011-01-24 20:49 +0000 [r303548] Russell Bryant * main/channel.c, main/pbx.c, /, apps/app_meetme.c, main/features.c, include/asterisk/channel.h: Merged revisions 303546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ 2011-01-21 21:48 +0000 [r303285] Jason Parker * channels/chan_dahdi.c, /: Merged revisions 303284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ 2011-01-21 16:12 +0000 [r303273] Leif Madsen * apps/app_dial.c: Fix changes to L() flag in Dial(). Tony Mountifield pointed out an error I had in my patch. I was a bit too aggressive on changing 'seconds' to 'milliseconds'. So I decided to do some additioanl testing and have no changed just the appropriate lines. One line says milliseconds, and the other says seconds. Probably should change this to be either just seconds or milliseconds, but I've spent too much time on this already :) (issue #18264) 2011-01-20 19:56 +0000 [r303106] Shaun Ruffell * main/features.c: main/features: Use POLLPRI when waiting for events on parked channels. This change resolves a regression in the 1.6.2 when converting from select to poll. The DAHDI timers use POLLPRI to indicate that the timer fired, but features was not waiting for that flag. The result was no audio for MOH when a call was parked and res_timing_dahdi was in use. This patch is slightly modified from the one on the mantis issue. It does not set an exception on the channel if the POLLPRI flag is set. (closes issue #18262) Reported by: francesco_r Patches: patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029) Tested by: francesco_r, rfrantik, one47 2011-01-20 17:07 +0000 [r303008] Jeff Peeler * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions 303007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 ........ 2011-01-20 16:11 +0000 [r302920] Russell Bryant * apps/app_privacy.c: Resolve a compiler warning. 2011-01-20 15:42 +0000 [r302917] Leif Madsen * apps/app_dial.c, /: Option L() is milliseconds, not seconds. > Change the verbose output of option L() to say milliseconds and not seconds > as the value is in milliseconds. > > (closes issue #18264) > Reported by: jacco > Patches: > app_dial_patch.txt uploaded by lmadsen (license 10) 2011-01-19 23:47 +0000 [r302833] Sean Bright * apps/app_voicemail.c: Support greetingsfolder as documented in voicemail.conf.sample. (closes issue #17870) Reported by: edhorton Patches: __20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10) 2011-01-19 23:06 +0000 [r302788] Russell Bryant * main/manager.c: Turn a noisy verbose message into a debug message. This can drown your console if you're using the AMI over HTTP. 2011-01-19 21:25 +0000 [r302693] Richard Mudgett * /, main/features.c: Merged revisions 302671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines DTMF transfer plays the wrong sounds for wrong number or other call failure. * Set the default for features.conf.sample xferfailsound option to "beeperr" as documented instead of "pbx-invalid" and corrected the use of it in DTMF blind transfer (#1). * Improved DTMF blind transfer handling of wrong numbers. Most of the concerns in this issue were taken care of by the patch for issue 17999: Issues with DTMF triggered attended transfers. (closes issue #18379) Reported by: gincantalupo Tested by: rmudgett ........ 2011-01-19 21:22 +0000 [r302599-302675] Tilghman Lesher * include/asterisk/astdb.h, /: Merged revisions 302663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 Jan 2011) | 2 lines Add some API documentation ........ * main/app.c: Kill zombies. When we ast_safe_fork() with a non-zero argument, we're expected to reap our own zombies. On a zero argument, however, the zombies are only reaped when there aren't any non-zero forked children alive. At other times, we accumulate zombies. This code is forward ported from res_agi in 1.4, so that forked children are always reaped, thus preventing an accumulation of zombie processes. (closes issue #18515) Reported by: ernied Patches: 20101221__issue18515.diff.txt uploaded by tilghman (license 14) Tested by: ernied 2011-01-19 19:02 +0000 [r302504-302554] Sean Bright * main/utils.c: Don't call strlen() when we only need to look at the next character or two. (closes issue #18042) Reported by: wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded by wdoekes (license 717) * main/features.c: Remove an extraneous \r\n at the end of a parking manager events. (closes issue #18363) Reported by: clegall_proformatique Patches: asterisk_1.8_295998_parking_manager_events_format.patch uploaded by clegall proformatique (license 1139) * res/res_agi.c: Properly handle partial reads from fgets() when handling AGIs. When fgets() failed with EAGAIN, we were continually decrementing the available space left in our buffer, resulting in botched command handling. (closes issue #16032) Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by fnordian (license 110) * main/utils.c: Make sure that h_length is set when we short-circuit out of ast_gethostbyname. (closes issue #16135) Reported by: thedavidfactor Patches: utils.patch uploaded by thedavidfactor (license 903) 2011-01-19 17:08 +0000 [r302461] Paul Belanger * res/res_timing_timerfd.c: Handle 'Resource temporarily unavailable' error more gracefully. 2011-01-19 15:52 +0000 [r302416] Sean Bright * configs/extensions.conf.sample: Remove references to priorityjumping from the sample extensions.conf. Priority jumping was removed from pbx_config in r68970. (closes issue #18622) Reported by: kshumard Patches: extensions.conf.sample.patch uploaded by kshumard (license 92) 2011-01-18 21:40 +0000 [r302313] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 302311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines URI encode the user part of the contact header. ABE-2705 ........ 2011-01-18 20:13 +0000 [r302265] Jeff Peeler * main/pbx.c: Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip. Lock scenario presented here: Thread 1 holds ast_rdlock_contexts &conlock holds handle_statechange hints holds handle_statechange hint waiting for cb_extensionstate Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds handle_request_do &netlock holds find_call sip_pvt_ptr waiting for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911 (ast_rdlock_contexts) Chan_sip has an established locking order of locking the sip_pvt and then getting the context lock. So the as stated by the summary, the operations in thread 2 have been modified to no longer require the context lock. (closes issue #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch uploaded by one47 (license 23), modified by me Review: https://reviewboard.asterisk.org/r/1072/ 2011-01-18 18:07 +0000 [r302173] Richard Mudgett * /, main/features.c: Merged revisions 302172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines Issues with DTMF triggered attended transfers. Issue #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in features.conf for attended transfer). 3) A hears MOH. B dial number C 4) C ringing. A hears MOH. 5) B hangup. A still hears MOH. C ringing. 6) A hangup. C still ringing until "atxfernoanswertimeout" expires. For v1.4 C will ring forever until C answers the dead line. (Issue #17096) Problem: When A and B hangup, C is still ringing. Issue #18395 SIP call limit of B is 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C ringing 4. Timeout waiting for C to answer 5. Recall to B fails because B has reached its call limit. Because B reached its call limit, it cannot do anything until the transfer it started completes. Issue #17273 Same scenario as issue 18395 but party B is an FXS port. Party B cannot do anything until the transfer it started completes. If B goes back off hook before C answers, B hears ringback instead of the expected dialtone. ********** Note for the issue #17273 and #18395 fix: DTMF attended transfer works within the channel bridge. Unfortunately, when either party A or B in the channel bridge hangs up, that channel is not completely hung up until the transfer completes. This is a real problem depending upon the channel technology involved. For chan_dahdi, the channel is crippled until the hangup is complete. Either the channel is not useable (analog) or the protocol disconnect messages are held up (PRI/BRI/SS7) and the media is not released. For chan_sip, a call limit of one is going to block that endpoint from any further calls until the hangup is complete. For party A this is a minor problem. The party A channel will only be in this condition while party B is dialing and when party B and C are conferring. The conversation between party B and C is expected to be a short one. Party B is either asking a question of party C or announcing party A. Also party A does not have much incentive to hangup at this point. For party B this can be a major problem during a blonde transfer. (A blonde transfer is our term for an attended transfer that is converted into a blind transfer. :)) Party B could be the operator. When party B hangs up, he assumes that he is out of the original call entirely. The party B channel will be in this condition while party C is ringing, while attempting to recall party B, and while waiting between call attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will replace the party B channel technology with a NULL channel driver to complete hanging up the party B channel technology. The consequences of this code is that the 'h' extension will not be able to access any channel technology specific information like SIP statistics for the call. ATXFER_NULL_TECH is not defined by default. ********** (closes issue #17999) Reported by: iskatel Tested by: rmudgett JIRA SWP-2246 (closes issue #17096) Reported by: gelo Tested by: rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: shihchuan Tested by: rmudgett (closes issue #17273) Reported by: grecco Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1047/ ........ 2011-01-17 16:53 +0000 [r302049] Terry Wilson * channels/chan_sip.c: Merged revisions 293493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 [^] ........ r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue 0017403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ [^] ........ Backporting a bugfix that should have been included. 2011-02-22 Leif Madsen * Asterisk 1.6.2.17 Released. * Merged in changes related to AST-2011-002 2011-02-16 Leif Madsen * Asterisk 1.6.2.17-rc3 Released. ------------------------------------------------------------------------ r308002 | qwell | 2011-02-15 17:32:21 -0600 (Tue, 15 Feb 2011) | 10 lines Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue 0018747) Reported by: vrban ------------------------------------------------------------------------ 2011-01-20 Leif Madsen * Asterisk 1.6.2.17-rc2 Released. ------------------------------------------------------------------------ r302172 | rmudgett | 2011-01-18 12:04:37 -0600 (Tue, 18 Jan 2011) | 88 lines Issues with DTMF triggered attended transfers. Issue 0017999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in features.conf for attended transfer). 3) A hears MOH. B dial number C 4) C ringing. A hears MOH. 5) B hangup. A still hears MOH. C ringing. 6) A hangup. C still ringing until "atxfernoanswertimeout" expires. For v1.4 C will ring forever until C answers the dead line. (Issue 0017096) Problem: When A and B hangup, C is still ringing. Issue 0018395 SIP call limit of B is 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C ringing 4. Timeout waiting for C to answer 5. Recall to B fails because B has reached its call limit. Because B reached its call limit, it cannot do anything until the transfer it started completes. Issue 0017273 Same scenario as issue 18395 but party B is an FXS port. Party B cannot do anything until the transfer it started completes. If B goes back off hook before C answers, B hears ringback instead of the expected dialtone. ********** Note for the issue 0017273 and 0018395 fix: DTMF attended transfer works within the channel bridge. Unfortunately, when either party A or B in the channel bridge hangs up, that channel is not completely hung up until the transfer completes. This is a real problem depending upon the channel technology involved. For chan_dahdi, the channel is crippled until the hangup is complete. Either the channel is not useable (analog) or the protocol disconnect messages are held up (PRI/BRI/SS7) and the media is not released. For chan_sip, a call limit of one is going to block that endpoint from any further calls until the hangup is complete. For party A this is a minor problem. The party A channel will only be in this condition while party B is dialing and when party B and C are conferring. The conversation between party B and C is expected to be a short one. Party B is either asking a question of party C or announcing party A. Also party A does not have much incentive to hangup at this point. For party B this can be a major problem during a blonde transfer. (A blonde transfer is our term for an attended transfer that is converted into a blind transfer. :)) Party B could be the operator. When party B hangs up, he assumes that he is out of the original call entirely. The party B channel will be in this condition while party C is ringing, while attempting to recall party B, and while waiting between call attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will replace the party B channel technology with a NULL channel driver to complete hanging up the party B channel technology. The consequences of this code is that the 'h' extension will not be able to access any channel technology specific information like SIP statistics for the call. ATXFER_NULL_TECH is not defined by default. ********** (closes issue 0017999) Reported by: iskatel Tested by: rmudgett JIRA SWP-2246 (closes issue 0017096) Reported by: gelo Tested by: rmudgett JIRA SWP-1192 (closes issue 0018395) Reported by: shihchuan Tested by: rmudgett (closes issue 0017273) Reported by: grecco Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1047/ [^] ------------------------------------------------------------------------ ------------------------------------------------------------------------ r303106 | sruffell | 2011-01-20 13:56:35 -0600 (Thu, 20 Jan 2011) | 15 lines main/features: Use POLLPRI when waiting for events on parked channels. This change resolves a regression in the 1.6.2 when converting from select to poll. The DAHDI timers use POLLPRI to indicate that the timer fired, but features was not waiting for that flag. The result was no audio for MOH when a call was parked and res_timing_dahdi was in use. This patch is slightly modified from the one on the mantis issue. It does not set an exception on the channel if the POLLPRI flag is set. (closes issue 0018262) Reported by: francesco_r Patches: patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029) Tested by: francesco_r, rfrantik, one47 ------------------------------------------------------------------------ 2011-01-14 Leif Madsen * Asterisk 1.6.2.17-rc1 Released. 2011-01-14 20:03 +0000 [r301842-301848] lathama : * funcs/func_base64.c, funcs/func_aes.c: Add relationships to function documentation. Fix amatuer type mistake * funcs/func_base64.c, funcs/func_aes.c: Add relationships to function documentation. 2011-01-13 17:01 +0000 [r301730] Leif Madsen * configs/phoneprov.conf.sample: Add static entry for split Polycom 332 firmware. (closes issue #18607) Reported by: cjacobsen Patches: polycom_331.diff uploaded by cjacobsen (license 1029) Tested by: lathama 2011-01-12 21:05 +0000 [r301682] Terry Wilson * channels/chan_sip.c: Don't reject all SUBSCRIBE auth requests When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh 2011-01-12 18:50 +0000 [r301594] Matthew Nicholson * main/manager.c, /: Removed a usleep(1) that shouldn't be necessary in session_do, and removed the ms_t member from the mansession_session structure. Merged revisions 301591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines Don't store the thread id for the manager session in the structure we pass to the thread for the manager session. ABE-2543 ........ 2011-01-12 18:11 +0000 [r301503] Jeff Peeler * main/channel.c, /: Merged revisions 301502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines Fix CPU spike when pressing DTMF after agent login. The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj ........ 2011-01-11 19:14 +0000 [r301310] Paul Belanger * configs/extensions.conf.sample: Fix a logic issue when passing context ARG 2011-01-11 18:42 +0000 [r301307] Matthew Nicholson * /, main/utils.c: Merged revisions 301305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan 2011) | 4 lines Prevent buffer overflows in ast_uri_encode() ABE-2705 ........ 2011-01-09 21:38 +0000 [r301176-301220] Paul Belanger * autoconf/ast_ext_lib.m4, configure, configure.ac: SOUND_CACHE_DIR now defaults to empty Sounds files included in the Asterisk tarball were being ignored and re-downloaded. Users wanting to cache the files can still override the setting using the --with-sounds-cache option. (closes issue #18589) Reported by: pabelanger Patches: issue18589.patch uploaded by pabelanger (license 224) Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/1074/ * apps/app_verbose.c: Indicate log level argument for Log() is not optional (closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) 2011-01-07 20:52 +0000 [r301089] Jason Parker * apps/app_meetme.c: Initialize useropts/adminopts in case there is no column in the realtime DB. (closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas 2011-01-07 19:57 +0000 [r300951-301046] Jeff Peeler * apps/app_voicemail.c: Fix regression causing forwarding voicemails to not work with file storage. I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 * apps/app_voicemail.c, /: Merged revisions 300918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played at the correct time. Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 ........ 2011-01-05 18:54 +0000 [r300622] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 300621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011) | 10 lines Use the sanity check in place of the disconnect/connect cycle. The disconnect/connect cycle has the potential to cause random crashes. (closes issue #18243) Reported by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147) Tested by: ks3 ........ 2011-01-05 16:28 +0000 [r300574] Paul Belanger * cdr/cdr_sqlite.c: Change deprecated message to LOG_WARNING Also removed latter part of message Discussed on #asterisk-dev 2011-01-04 21:52 +0000 [r300431-300520] Leif Madsen * channels/chan_iax2.c, main/xmldoc.c, channels/chan_sip.c, channels/chan_agent.c: Fix backwards and broken XML documentation. (closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) * configs/users.conf.sample: Add some documentation to users.conf.sample. (closes issue #18531) Reported by: lathama Patches: users.conf.sample2.diff uploaded by lathama (license 1028) Tested by: lathama 2011-01-04 20:59 +0000 [r300429] Russell Bryant * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8: Merged revisions 300428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011) | 4 lines Update the autosupport script from Digium support. (closes AST-395) ........ 2011-01-04 17:37 +0000 [r300298] Terry Wilson * /, channels/chan_sip.c: Merged revisions 300216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines Don't authenticate SUBSCRIBE re-transmissions This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ ........ 2011-01-03 23:02 +0000 [r300165] Richard Mudgett * main/features.c: Use correct variable for atxfercallbackretries config option. * Misc formatting changes. 2010-12-28 18:51 +0000 [r299864] Paul Belanger * apps/app_chanspy.c: Documentation typo 2010-12-25 10:05 +0000 [r299625] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 299624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) | 5 lines Move check for extension existence below variable inheritance, due to the possible use of an eswitch. (closes issue #16228) Reported by: jlaguilar ........ 2010-12-23 03:02 +0000 [r299530-299533] Moises Silva * channels/chan_dahdi.c: do not use progress which is for PRI and SS7, add mfcr2_progress member * channels/chan_dahdi.c: Enqueue AST_CONTROL_PROGRESS after AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue #18438) Reported by: mariner7 Tested by: moy 2010-12-22 20:03 +0000 [r299448] Tilghman Lesher * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest25, pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-test3: Resolve warnings by disambiguating the "s" extension as used by chan_dahdi from the "s" extension as used by the AEL macros. (closes issue #18480) Reported by: nivek Patches: 20101215__issue18480__2.diff.txt uploaded by tilghman (license 14) Tested by: nivek 2010-12-20 21:25 +0000 [r299242] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 299194,299198,299220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines Respond as soon as possible with a 202 Accepted to refer requests. This change also plugs a few memory leaks that can occur when parking sip calls. ABE-2656 ........ r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines Remove changes to via processing that were not supposed to go into the last commit. ........ r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use ast_free() instead of free() ABE-2656 ........ 2010-12-20 18:16 +0000 [r299130-299136] Tilghman Lesher * sample.call: Documentation fix * cdr/cdr_pgsql.c: If a call was not answered, then the billsec was calculated unusually large. Also, due to a copy and paste error, a request for the answer field would have given the start value, instead. (closes issue #18460) Reported by: joscas Patches: 20101215__issue18460.diff.txt uploaded by tilghman (license 14) Tested by: joscas 2010-12-20 16:18 +0000 [r299087] Leif Madsen * main/features.c: Note that Park() timeout is milliseconds. (closes issue #15758) Reported by: mmurdock Tested by: mmurdock, seanbright 2010-12-20 09:13 +0000 [r299003] Tzafrir Cohen * channels/chan_sip.c: Typos: recieved => received 2010-12-18 00:08 +0000 [r298817-298962] Tilghman Lesher * main/say.c: Remove backtrace used for testing merge process * main/astobj2.c, utils/conf2ael.c, include/asterisk/logger.h, configure, build_tools/menuselect-deps.in, main/logger.c, utils/ael_main.c, utils/hashtest2.c, makeopts.in, utils/check_expr.c, utils/refcounter.c, include/asterisk/utils.h, build_tools/cflags-devmode.xml, /, main/Makefile, include/asterisk/autoconfig.h.in, main/say.c, configure.ac, utils/hashtest.c, main/utils.c: Merged revisions 298905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines Let Asterisk find better backtrace information with libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. ........ * configure, configure.ac: Also include PTHREAD_LIBS and PTHREAD_CFLAGS for SQLite 3, as it's needed on some platforms. (closes issue #18493) Reported by: pprindeville Patches: asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347) Tested by: pprindeville 2010-12-16 23:30 +0000 [r298597-298684] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 298683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 Dec 2010) | 2 lines After recording only silence for a voicemail prepending, restore backup files. ........ * apps/app_queue.c, /: Merged revisions 298596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines Fix improper hangup when doing an attended transfer to queue. Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 ........ 2010-12-16 09:04 +0000 [r298393-298481] Tilghman Lesher * res/res_config_odbc.c, /: Merged revisions 298480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010) | 14 lines Only increment the pointer once per loop, otherwise we corrupt the value. (closes issue #18251) Reported by: bcnit Patches: 20101110__issue18251.diff.txt uploaded by tilghman (license 14) Tested by: trev, jthurman, elguero (closes issue #18279) Reported by: zerohalo Patches: 20101109__issue18279.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ * funcs/func_dialgroup.c: Eliminate duplicates from container. (closes issue #18091) Reported by: bunny Patches: 20101006__issue18091.diff.txt uploaded by tilghman (license 14) Tested by: bunny * /, cdr/cdr_sqlite.c: Merged revisions 298392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010) | 8 lines Unregister before shutting down the connection, to avoid a race. (closes issue #18481) Reported by: pabelanger Patches: 20101215__issue18481.diff.txt uploaded by tilghman (license 14) Tested by: pabelanger ........ 2010-12-15 21:31 +0000 [r298346] Sean Bright * main/astobj2.c, /: Merged revisions 298345 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298345 | seanbright | 2010-12-15 16:28:29 -0500 (Wed, 15 Dec 2010) | 6 lines Fix reference and container leaks when running 'astobj2 test.' We need to make sure that ao2_iterator_destroy is called once for each time that ao2_iterator_init is called. Also make sure to unref a newly allocated object that we've linked into a container. ........ 2010-12-13 17:04 +0000 [r298194] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 298193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) | 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered transfers. Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING message is not received. The debug output shows that the DTMF begin event is seen, but the DTMF end event is missing. When the DTMF begin happens, the call is muted so we now have one way audio (until a DTMF end event is somehow seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin and DTMF end events if we are overlap dialing and have not seen a PROCEEDING message. * Added a debug message when absorbing a DTMF event. JIRA SWP-2690 JIRA ABE-2697 ........ 2011-01-12 Leif Madsen * Asterisk 1.6.2.16 Released. 2011-01-12 Leif Madsen * Merge in changes for configure script to resolve issue for Debian package builders. ------------------------------------------------------------------------ r301220 | pabelanger | 2011-01-09 15:38:25 -0600 (Sun, 09 Jan 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds files included in the Asterisk tarball were being ignored and re-downloaded. Users wanting to cache the files can still override the setting using the --with-sounds-cache option. (closes issue 0018589) Reported by: pabelanger Patches: issue18589.patch uploaded by pabelanger (license 224) Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/1074/ [^] ------------------------------------------------------------------------ 2010-12-13 Leif Madsen * Asterisk 1.6.2.16-rc1 Released. 2010-12-10 16:24 +0000 [r298050] Tilghman Lesher * main/netsock.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Portability issue on OpenSolaris. Also detect the required structure element, because OpenSolaris defines SIOCGIFHWADDR, but without support for IP sockets. (closes issue #18442) Reported by: ranjtech Patches: 20101209__issue18442.diff.txt uploaded by tilghman (license 14) Tested by: ranjtech 2010-12-09 22:10 +0000 [r297960] Terry Wilson * /, channels/chan_sip.c: Merged revisions 297959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines Ignore spurious REGISTER requests If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ 2010-12-08 18:04 +0000 [r297908] Tilghman Lesher * configs/extensions.conf.sample: Use inheritance to get correct results for SIPFROMDOMAIN. (from an internal Digium discussion) 2010-12-07 22:58 +0000 [r297824] Jeff Peeler * main/channel.c, /: Merged revisions 297823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) | 12 lines Revert code that changed SSRC for DTMF. Some previous behavior was attempted to be restored, but mistakingly I did not realize that the previous behavior was incorrect. This fixes DTMF not being detected since DTMF shouldn't cause the SSRC to change. (related to issue #17404) (closes issue #18189) (closes issue #18352) Reported by: marcbou Tested by: cmbaker82 ........ 2010-12-07 22:40 +0000 [r297713-297819] Tilghman Lesher * contrib/init.d/org.asterisk.muted.plist (added), Makefile, utils/muted.c, /: Merged revisions 297818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010) | 4 lines Use non-deprecated APIs for CoreAudio Review: https://reviewboard.asterisk.org/r/1040/ ........ * apps/app_followme.c, /: Merged revisions 297689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) ........ 2010-12-06 22:03 +0000 [r297605] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 297603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) | 12 lines Improve handling of REGISTER requests with multiple contact headers. The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 ........ 2010-12-03 17:40 +0000 [r297534] Sean Bright * channels/chan_console.c: The CLI command should not contain s, these are for descriptions. 2010-12-02 20:06 +0000 [r297405] Paul Belanger * Makefile, /: Merged revisions 297404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec 2010) | 7 lines Resolve compile error under FreeBSD We now set _ASTCFLAGS+=-march=i686 for i386 processors, still allowing ASTCFLAGS to override the setting. Review: https://reviewboard.asterisk.org/r/1043/ ........ 2010-12-02 18:07 +0000 [r297311] Terry Wilson * /, main/abstract_jb.c: Merged revisions 297310 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010) | 12 lines Initialize offset for adaptive jitter buffer When the adaptive jitter buffer is enabled in sip.conf, the first frame placed in the jitter buffer fails with something like: jb_warning_output: Resyncing the jb. last_delay 0, this delay -215886466, threshold 1000, new offset 215886466 This happens because the offset is not initialized before calling jb_put(). This patch modifies jb_put_first_adaptive() to set the offset to the frame's timestamp. Review: https://reviewboard.asterisk.org/r/1041/ ........ 2010-12-02 13:16 +0000 [r297229] Russell Bryant * /, apps/app_meetme.c: Merged revisions 297228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines Add "DAHDI" to a couple of app_meetme error messages. This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. ........ 2010-12-02 08:55 +0000 [r297186] Olle Johansson * /, channels/chan_sip.c: Merged revisions 297185 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5 lines If we get a NOTIFY from a non-existing subscription we should answer with 481, not bad event. If we answer 481 the subscription that we don't want will be cancelled. ........ 2010-12-01 17:52 +0000 [r297073] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 297072 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 ........ 2010-12-01 17:01 +0000 [r296950-296991] Tilghman Lesher * include/asterisk/frame.h, /: Merged revisions 296990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010) | 5 lines Clarify documentation on how we store codec preference lists. (closes issue #18397) Reported by: birgita ........ * channels/chan_iax2.c: Missed initializations caused startup errors on Mac OS X (and possibly others, too). 2010-12-01 00:24 +0000 [r296869] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 296868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) | 4 lines Properly restore backup information file when hanging up during message prepending. ABE-2654 ........ 2010-11-29 22:54 +0000 [r296671] Paul Belanger * channels/chan_iax2.c, /: Merged revisions 296670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov 2010) | 5 lines Make sure nothing else is needed before destroying the scheduler. (closes issue #18398) Reported by: pabelanger ........ 2010-11-29 07:27 +0000 [r296533] Tilghman Lesher * main/asterisk.c, configure, include/asterisk/autoconfig.h.in, configure.ac: I love standards. There are so many to choose from. Except when there isn't one. Linux and *BSD disagree on the elements within the ucred structure. Detect which one is in use on the system. (closes issue #18384) Reported by: bjm Patches: cred-diffs uploaded by bjm (license 473) 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman (license 14) 20101127__issue18384__1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, bjm 2010-11-27 10:39 +0000 [r296466] Tilghman Lesher * apps/app_meetme.c: 18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision). (closes issue #18369) Reported by: tnakonz 2010-11-26 12:23 +0000 [r296351] Olle Johansson * /, main/say.c: Merged revisions 296309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11 lines Fix bugs in saying numbers using the Swedish language syntax (closes issue #18355) Reported by: oej Patch by: oej Much help from Peter Lindahl. Testing by the ClearIT team during a coffee break. Review: https://reviewboard.asterisk.org/r/1033/ ........ 2010-11-24 23:28 +0000 [r296221] Russell Bryant * main/channel.c, /: Merged revisions 296213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines Make Asterisk less crashy. Since we might not put a new translation path on the channel, go ahead and set it to NULL right after destroying the old one to ensure we don't try to free an invalid translation path later on. ........ 2010-11-24 22:42 +0000 [r296166] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 296165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip. The FXS connected phone has to have CW/CID support to fail, as it will send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal phone with no CID never fails. Also the SIP phone does not hear MOH when the CW call is answered. The DTMF end frame is suppressed when the phone acknowledges the CW signal for CID. The problem is the DTMF begin frame needs to be suppressed as well. The DTMF begin frame is causing SIP to start sending the DTMF RTP frames. Since the DTMF end frame is suppressed, SIP will not stop sending those DTMF RTP packets. * Suppress the DTMF begin and end frames when the channel driver is looking for DTMF digits. * Fixed a couple issues caused by not cleaning up the CID spill if you answer the CW call while it is sending the CID spill. * Fixed not sending CW/CID spill to the phone when the call is natively bridged. (Fixed by not using native bridge if CW/CID is possible.) * Suppress received audio when sending CW/CID spills. The other parties involved do not need to hear the CW/CID spills and may be confused if the CW call is for them. (closes issue #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett NOTE: * v1.4 does not have the main problem fixed by suppressing the DTMF start frames. The other three items fixed are relevant. * If you really must restore native bridging between analog ports, you need to disable CW/CID either by configuring chan_dahdi.conf callwaitingcallerid=no or dialing *70 before dialing the number to temporarily disable CW. ........ 2010-11-24 20:23 +0000 [r296001-296083] Russell Bryant * main/channel.c, /: Merged revisions 296082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines Fix false reporting of an error by set_format(). In the case that the native format was able to be changed to match the new requested format, the code proceeded to attempt to build a translation path, anyway. The result would be NULL, since no translation path is necessary and resulted in this function thinking an error has occurred. This case is now specifically caught and no attempt to build a translation path is attempted. Thanks to our automated tests and bamboo.asterisk.org for catching this problem and making a whole lot of noise when things started failing. :-) ........ * apps/app_dial.c, main/channel.c, /: Merged revisions 296000 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ 2010-11-23 09:36 +0000 [r295907] Olle Johansson * /, main/say.c: Merged revisions 295906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8 lines Fix support of saynumber(1,n) in the Swedish language (closes issue #18353) Reported by: oej Review: https://reviewboard.asterisk.org/r/1031/ ........ 2010-11-22 20:02 +0000 [r295868] Sean Bright * configs/chan_dahdi.conf.sample: Change some documentation to suggest dahdi_monitor instead of ztmonitor. 2010-11-22 19:28 +0000 [r295843] Richard Mudgett * include/asterisk/frame.h, main/channel.c, main/pbx.c, /, apps/app_macro.c, include/asterisk/channel.h: Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ 2010-11-20 00:45 +0000 [r295710] Russell Bryant * include/asterisk/event.h, main/event.c: Fix cache of device state changes for multiple servers. This patch addresses a regression where device states across multiple servers were not being processing completely correctly. The code works to determine the overall state by looking at the last known state of a device on each server. However, there was a regression due to some invasive rewrites of how the cache works that led to the cache only storing the last device state change for a device, regardless of which server it was on. The code is set up to cache device state change events by ensuring that each event in the cache has a unique device name + entity ID (server ID). The code that was responsible for comparing raw information elements (which EID is) always returned a match due to a memcmp() with a length of 0. There isn't much code to fix the actual bug. This patch also introduces a new CLI command that was very useful for debugging this problem. The command allows you to dump the contents of the event cache. (closes issue #18284) Reported by: klaus3000 Patches: issue18284.rev1.txt uploaded by russell (license 2) Tested by: russell, klaus3000 (closes issue #18280) Reported by: klaus3000 Review: https://reviewboard.asterisk.org/r/1012/ 2010-11-19 21:55 +0000 [r295672] Terry Wilson * /, channels/chan_sip.c: Merged revisions 295628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) | 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ 2010-11-18 17:51 +0000 [r295440] Paul Belanger * res/res_jabber.c, include/asterisk/jabber.h: Fix compiler warnings when using openssl-dev 1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ 2010-11-16 22:57 +0000 [r295281] Richard Mudgett * main/channel.c, /: Merged revisions 295280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) | 1 line Dead code elimination in channel.c:ast_channel_bridge() variable who. ........ 2010-12-02 Leif Madsen * Asterisk 1.6.2.15 Released. 2010-11-15 Leif Madsen * Asterisk 1.6.2.15-rc1 2010-11-15 18:24 +0000 [r294988-295062] Tilghman Lesher * tests/test_expr.c (added), /: Merged revisions 295026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010) | 2 lines Create test verifying results of expression parser ........ * funcs/func_curl.c: It is possible to crash Asterisk by feeding the curl engine invalid data. (closes issue #18161) Reported by: wdoekes Patches: 20101029__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: tilghman 2010-11-12 21:14 +0000 [r294904-294910] Jeff Peeler * apps/app_voicemail.c: Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent. Reported by alecdavis in asterisk-dev. * apps/app_voicemail.c, /: Merged revisions 294903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) | 16 lines Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger ........ 2010-11-12 02:44 +0000 [r294822] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 294821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) | 11 lines Asterisk is getting a "No D-channels available!" warning message every 4 seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls ........ 2010-11-11 21:57 +0000 [r294639-294733] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ * main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged revisions 294384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) | 47 lines Fix a deadlock in device state change processing. Copied from some notes from the original author (Russell): Deadlock scenario: Thread 1: device state change thread Holds - rdlock on contexts Holds - hints lock Waiting on channels container lock Thread 2: SIP monitor thread Holds the "iflock" Holds a sip_pvt lock Holds channel container lock Waiting for a channel lock Thread 3: A channel thread (chan_local in this case) Holds 2 channel locks acquired within app_dial Holds a 3rd channel lock it got inside of chan_local Holds a local_pvt lock Waiting on a rdlock of the contexts lock A bunch of other threads waiting on a wrlock of the contexts lock To address this deadlock, some locking order rules must be put in place and enforced. Existing relevant rules: 1) channel lock before a pvt lock 2) contexts lock before hints lock 3) channels container before a channel What's missing is some enforcement of the order when you involve more than any two. To fix this problem, I put in some code that ensures that (at least in the code paths involved in this bug) the locks in (3) come before the locks in (2). To change the operation of thread 1 to comply, I converted the storage of hints to an astobj2 container. This allows processing of hints without holding the hints container lock. So, in the code path that led to thread 1's state, it no longer holds either the contexts or hints lock while it attempts to lock the channels container. (closes issue #18165) Reported by: antonio ABE-2583 ........ 2010-11-10 23:16 +0000 [r294571] Tilghman Lesher * main/features.c: Actually pay attention to documented settings in features.conf. (closes issue #16757) Reported by: voxter Patches: 20101012__issue16757.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/994/ 2010-11-10 12:41 +0000 [r294500] Russell Bryant * main/devicestate.c: Improve a debug message to be more readable and consistent. (closes issue #18282) Reported by: klaus3000 Patches: ast_devstate2str-patch.txt uploaded by klaus3000 (license 65) 2010-11-09 20:27 +0000 [r294429] Tilghman Lesher * configure, configure.ac: Detect GMime properly on systems where gmime flags and libs are configured with pkg-config. (closes issue #16155) Reported by: jcollie Patches: 20100917__issue16155.diff.txt uploaded by tilghman (license 14) Tested by: tilghman 2010-11-08 22:30 +0000 [r294277-294312] Jeff Peeler * res/res_timing_timerfd.c: add missing unlock not present in 294277 * main/timing.c, main/channel.c, res/res_timing_timerfd.c, include/asterisk/timing.h: Fix playback failure when using IAX with the timerfd module. To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! 2010-11-08 20:50 +0000 [r294242] Matthew Nicholson * channels/chan_sip.c: Go off hold when we get an empty reinvite telling us to. (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd 2010-11-05 00:06 +0000 [r293969] Shaun Ruffell * codecs/codec_dahdi.c, /: Merged revisions 293968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically commit 9034) added the capability for the wctc4xxp to return more than a single packet of data in response to a read. However, when decoding packets, codec_dahdi was still assuming that the default number of samples was in each read. In other words, each packet your provider sent you, regardless of size, would result in 20 ms of decoded data (30 ms if decoding G723). If your provider was sending 60 ms packets then codec_dahdi would end up stripping 40 ms of data from each transcoded frame resulting in "choppy" audio. This would only affect systems where G729 packets are arriving in sizes greater than 20ms or G723 packets arriving in sizes greater than 30ms. DAHDI-744. ........ 2010-11-03 18:31 +0000 [r293806] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 293805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) | 20 lines Party A in an analog 3-way call would continue to hear ringback after party C answers. All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. ........ 2010-11-02 23:07 +0000 [r293723] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 293722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ 2010-11-02 21:26 +0000 [r293647] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 293639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) | 6 lines Make warning message have more useful information in it. Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '' on channel ((), line )". ........ 2010-10-30 01:49 +0000 [r293340-293417] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 293416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) | 1 line Remove some more code that serves no purpose. ........ * channels/chan_dahdi.c, /: Merged revisions 293339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) | 1 line Remove some code that serves no purpose. ........ 2010-10-28 19:54 +0000 [r293195-293196] Tilghman Lesher * main/ast_expr2.c, main/ast_expr2.h: Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ * /, res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c, res/ael/ael.tab.h: Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ 2010-10-28 16:09 +0000 [r293158] Jeff Peeler * funcs/func_strings.c: Fix infinite loop in FILTER(). Specifically when you're using characters above \x7f or invalid character escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: issue18060_func_strings_filter_infinite_loop.patch uploaded by wdoekes (license 717) Tested by: wdoekes 2010-10-26 18:33 +0000 [r293118] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 293004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) | 29 lines Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 ........ 2010-10-25 19:06 +0000 [r292867] David Vossel * channels/chan_local.c, /: Merged revisions 292866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) | 27 lines This patch turns chan_local pvts into astobj2 objects. chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. ........ 2010-10-22 21:16 +0000 [r292786] Leif Madsen * contrib/scripts/asterisk.ldif, channels/chan_sip.c, configs/res_ldap.conf.sample: Update the LDIF file for LDAP. The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner 2010-10-21 13:11 +0000 [r292556] Leif Madsen * configs/res_ldap.conf.sample: Change res_ldap.sample.conf to match the schema. (closes issue #17376) Reported by: jcovert Patches: res_ldap.conf.sample.patch uploaded by jcovert (license 551) 2010-10-21 00:05 +0000 [r292412] Paul Belanger * apps/app_dial.c, /: Merged revisions 292411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) | 10 lines Record priv-recordintro as sln, not gsm This removes the gsm->sln step when transcoding priv-recordintro. (closes issue #18176) Reported by: pabelanger Patches: chan_sip.diff uploaded by pabelanger (license 224) ........ 2010-10-18 22:01 +0000 [r292229] Leif Madsen * sounds/Makefile: Fix typo in the sounds/Makefile. (Issue #17426) 2010-10-18 21:54 +0000 [r292226] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 292223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) | 11 lines Fix improper operator key acceptance and clean up temp recording files. This is a fix for when pressing the operator key after recording an unavailable, busy, name, or temporary message in mailbox options. The operator key should not be accepted here, but should be allowed during the message recording. If the operator key is pressed during ensure the file is saved or deleted as apporopriate. Also, ensure removal of temporary recorded files after an early hang up or when message acceptance confirmation times out. ABE-2518 ........ 2010-10-18 21:50 +0000 [r292224] Leif Madsen * sounds/Makefile, /, sounds/sounds.xml: Merged revisions 292222 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010) | 9 lines Add support for the new English (Australian Accent) sound files. (closes issue #17426) Reported by: camsown Patches: core-sounds-en_AU.txt uploaded by camsown (license 1050) add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested by: camsown, lmadsen, jtodd, qwell ........ 2010-10-16 10:03 +0000 [r292049] Tzafrir Cohen * res/res_musiconhold.c, configs/musiconhold.conf.sample: Base directory for MOH should be ASTDATADIR If the directive 'directory' is relative, make it relative to the datadir, rather than to the varlibdir. In the sample configuration it is relative ('moh'). This has no effect unless you have actively set the datadir explicitly (at build time or at run time). (closes issue #16906) Patches: moh_datadir uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/974/ 2010-10-15 19:35 +0000 [r291939] Paul Belanger * configs/gtalk.conf.sample, /: Merged revisions 291938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct 2010) | 2 lines Clean up formatting. ........ 2010-10-15 16:16 +0000 [r291904] Terry Wilson * res/res_jabber.c: Don't crash or deadlock on module unload We can't hold the lock while pthread_join is called since aji_log_hook will attempt to lock from the other therad. We reorder the pthread_join and ast_aji_disconnect so that we don't do an SSL_read() while SSL_shutdown is running, causing a crash. 2010-10-13 23:36 +0000 [r291655] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 291643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) | 20 lines Deadlock between dahdi_exception() and dahdi_indicate(). There is a deadlock between dahdi_exception() and dahdi_indicate() for analog ports. The call-waiting and three-way-calling feature can experience deadlock if these features are trying to do something and an event from the bridged channel happens at the same time. Deadlock avoidance code added to obtain necessary channel locks before attemting an operation with call-waiting and three-way-calling. (closes issue #16847) Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/971/ ........ 2010-10-13 22:58 +0000 [r291580] Terry Wilson * main/channel.c, /: Merged revisions 291577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines Don't ignore frames that have been queued when softhangup'd When an outgoing call is answered and hung up by the far end *very* quickly, we may not read any frames and therefor end up with a call that displays the wrong disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately sets the _softhangup flag on the channel and then queues the HANGUP control frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates that a hangup request has been made (which it will if _softhangup is set). So, we end up losing control frames. This change makes __ast_read continue to read frames even if a soft hangup has been requested. It queues a hangup frame to make sure that __ast_read() will still eventually return NULL. Much thanks to David Vossel for all of the reviews, discussion, and help! (closes issue #16946) Reported by: davidw Review: https://reviewboard.asterisk.org/r/740/ ........ 2010-10-13 15:29 +0000 [r291393] Russell Bryant * /, channels/chan_sip.c: Merged revisions 291392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ 2010-10-12 17:20 +0000 [r291280] Leif Madsen * configs/phoneprov.conf.sample: Add undocumented variables to phoneprov.conf.sample (closes issue #18107) Reported by: lathama Patches: phoneprov.conf.sample.diff uploaded by lathama (license 1028) 2010-10-12 17:05 +0000 [r291264] Tilghman Lesher * /, main/acl.c: Merged revisions 291263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010) | 2 lines Oops, incorrect range (although unallocated at ARIN) ........ 2010-10-12 16:07 +0000 [r291229] Leif Madsen * configs/manager.conf.sample: Add documention that mentions options are defined but not used. (Issue #18101) 2010-10-11 18:39 +0000 [r291073-291111] Richard Mudgett * channels/chan_sip.c: Make exit from handle_request_do() consistent. * /, channels/chan_sip.c: Merged revisions 291109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line Add missing unlock to an exception condition in reload_config(). ........ * main/cli.c: Fixed infinite loop in verbose/debug message output. Setting the module/filename specific message level and then changing it resulted in the linked list being looped on itself. Traversing this linked list is an infinite loop if what you are looking for is not in the list. Also plugged some CLI parsing holes in the associated CLI command: * Removing a nonexistent module from the list actually added it with a level of zero. * Setting the non-module specific level to zero is now equivalent to setting it to "off" as documented. 2010-10-08 02:45 +0000 [r290863] Jeff Peeler * main/asterisk.c, /: Merged revisions 290862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed at control console. A recent change was made to avoid a race condition on shutdown which only called the end functions from the console thread. However, when pressing Ctrl-C the quit handler is called from the signal handler thread. (closes issue #17698) Reported by: jmls ........ 2010-10-07 20:57 +0000 [r290751] Jason Parker * autoconf/ast_ext_lib.m4, /, configure, include/asterisk/autoconfig.h.in: Merged revisions 290750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | 9 lines Allow PRI to build properly when using --with-pri. Use the directories found for the parent when using lib dependencies. (closes issue #17314) Reported by: tzafrir Patches: 17314-withdeps.diff uploaded by qwell (license 4) ........ 2010-10-07 10:53 +0000 [r290712] Russell Bryant * main/pbx.c: Don't crash when Set() is called without a value. Review: https://reviewboard.asterisk.org/r/949/ 2010-10-06 13:48 +0000 [r290396-290575] Tilghman Lesher * main/file.c: Allow streaming audio from a pipe. (closes issue #18001) Reported by: jamicque Patches: 20100926__issue18001.diff.txt uploaded by tilghman (license 14) Tested by: jamicque * res/res_jabber.c, /: Merged revisions 290392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010) | 8 lines Fix a crash by ensuring that we don't alter memory after it's freed. (closes issue #17387) Reported by: jmls Patches: 20100726__issue17387.diff.txt uploaded by tilghman (license 14) Tested by: jmls ........ 2010-10-05 19:54 +0000 [r290375] David Vossel * apps/app_directed_pickup.c: Fixes PickupChan() not working with full channel name. (closes issue #18011) Reported by: schern Patches: app_directed_pickup.c.2.patch uploaded by schern (license 995) app_directed_pickup.c.trunk.patch uploaded by schern (license 995) Tested by: schern, dvossel 2010-10-05 17:42 +0000 [r290324] Richard Mudgett * contrib/valgrind.supp, /: Merged revisions 290323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r290323 | rmudgett | 2010-10-05 12:41:18 -0500 (Tue, 05 Oct 2010) | 11 lines Merged revision 258974 from https://origsvn.digium.com/svn/asterisk/trunk .......... r258974 | diruggles | 2010-04-26 14:05:47 -0500 (Mon, 26 Apr 2010) | 4 lines Line 24 missed in compatibility fix in revision 233577 added a "fun:" prefix line 24 .......... ................ 2010-10-04 23:14 +0000 [r290101-290254] Tilghman Lesher * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c, pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5: Change new pattern matcher to regard dashes the same as the old pattern matcher -- as visual candy to be ignored. Also change the AEL parser to not generate dashes within extensions, as those dashes would be ignored. Update the AEL tests to match this behavior. (closes issue #17366) Reported by: murf Patches: 20100727__issue17366.diff.txt uploaded by tilghman (license 14) Tested by: tilghman * /, configure, configure.ac: Merged revisions 290177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04 Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........ * /, configure, configure.ac: Merged revisions 290100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03 Oct 2010) | 2 lines Automatically re-run configure test for menuselect, when the relevant makeopts settings change. ........ 2010-10-02 08:52 +0000 [r289950] Olle Johansson * main/manager.c, /: Merged revisions 289949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 lines Add documentation for undocumented option to AMI action originate ........ 2010-10-02 04:45 +0000 [r289874] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 289873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) | 8 lines When forwarding a message, a prepend means that the filesystem will always have a better copy. (closes issue #17803) Reported by: dpetersen Patches: 20100923__issue17803.diff.txt uploaded by tilghman (license 14) Tested by: dpetersen ........ 2010-10-01 23:01 +0000 [r289798] Jeff Peeler * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ 2010-10-01 17:09 +0000 [r289704] Paul Belanger * res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions 289703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct 2010) | 6 lines Disable debugging by default and reformat .config file. Review: https://reviewboard.asterisk.org/r/929/ ........ 2010-10-01 16:21 +0000 [r289700] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 289699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines Ensure user portion of SIP URI matches dialplan when using encoded characters. This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes ........ 2010-10-01 09:42 +0000 [r289622] schmitds : * channels/chan_sip.c: don't iterate through all dialogs to find and delete old subscribes On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed. (closes issue #17950) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/901/ 2010-09-30 19:51 +0000 [r289553] Matthew Nicholson * channels/chan_sip.c: Properly handle channel allocation failures duing invites with replaces. ABE-2588 2010-09-30 17:09 +0000 [r289501] Brett Bryant * /, res/res_agi.c: Merged revisions 289500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289500 | bbryant | 2010-09-30 13:08:20 -0400 (Thu, 30 Sep 2010) | 11 lines res_agi.c:handle_getvariablefull() could recursively lock a channel and not release it if an argument is the current channel's name. (closes issue #17970) Reported by: mdu113 Patches: res_agi.c.diff3 uploaded by mdu113 (license 582) Tested by: mdu113 Review: https://reviewboard.asterisk.org/r/947/ ........ 2010-09-30 15:37 +0000 [r289425] Russell Bryant * /, apps/app_sms.c: Merged revisions 289424 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) | 8 lines Fix a crash in app_sms. Since the data being passed to the generator callback is on the stack of the SMS() application, we must ensure that the generator is stopped before the application exits. ABE-2587 ........ 2010-09-29 21:03 +0000 [r289339] Jason Parker * main/channel.c, /, main/features.c: Merged revisions 289338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | 8 lines Allow a manager originate to succeed on forwarded devices. The timeout to wait for an answer was being set to 0 when a device forwarded to another extension. We don't always need the timeout set like this, so make it an optional parameter, and don't use it in this case. ABE-2544 ........ 2010-09-29 20:24 +0000 [r289334] Leif Madsen * configs/res_ldap.conf.sample: Update sample documentation to note md5secret requirements. 2010-09-29 20:15 +0000 [r289332] Russell Bryant * res/res_config_ldap.c: Don't completely ignore md5secret from LDAP if the value does not begin with {md5}. This fixes a problem that lmadsen ran in to where md5secret was not working for him. 2010-09-29 15:04 +0000 [r289178] Matthew Nicholson * main/channel.c, /: Merged revisions 289177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep 2010) | 8 lines Set the caller id on CDRs when it is set on the parent channel. (closes issue #17569) Reported by: tbelder Patches: 17569.diff uploaded by tbelder (license 618) Tested by: tbelder ........ 2010-09-28 18:14 +0000 [r289095] Brett Bryant * main/channel.c, /: Merged revisions 289094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) | 14 lines Fixes an issue with the Newchannel AMI event during the Masquerading process. Fixes an issue with the Newchannel AMI event during the Masquerading process, where no Newchannel AMI event was generated for the psuedo channel used during the masquerading process. (closes issue #17987) Reported by: RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish (license 1122) Tested by: RadicAlish Review: https://reviewboard.asterisk.org/r/937/ ........ 2010-09-24 15:37 +0000 [r288747] Terry Wilson * channels/chan_local.c, /: Merged revisions 288746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) | 5 lines Don't fail a masquerade if it is already being hung up This avoids noise on some Local channel situations where we don't use /n. Thanks to Alec Davis for the suggestion. ........ 2010-09-24 13:53 +0000 [r288637-288712] Tilghman Lesher * funcs/func_strings.c: Solaris won't printf a NULL. (closes issue #18041) Reported by: asgaroth * cdr/cdr_pgsql.c, /, configure, include/asterisk/autoconfig.h.in, include/asterisk/compat.h, main/strcompat.c, configure.ac, include/asterisk/channel.h: Merged revisions 288636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines Solaris compatibility fixes ........ 2010-09-22 23:10 +0000 [r288500] Terry Wilson * channels/chan_local.c, /: Merged revisions 288499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) | 8 lines Don't let a Local channel get bridged to itself If a local channel gets bridged to itself, it becomes orphaned with no devices left to actually tell it to hang up. This patch modifies local_fixup() to detect this case and deny it. Review: https://reviewboard.asterisk.org/r/934 ........ 2010-09-22 17:49 +0000 [r288344-288417] David Vossel * /, channels/chan_sip.c: Merged revisions 288416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) | 5 lines RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response. ABE-2458 ........ * /, channels/chan_sip.c: Merged revisions 288343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) | 2 lines During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup. ........ 2010-09-22 16:44 +0000 [r288340] Russell Bryant * main/asterisk.c, /: Merged revisions 288339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) | 11 lines Fix a 100% CPU consumption problem when setting console=yes in asterisk.conf. The handling of -c and console=yes should be the same, but they were not. When you specify -c, it sets both a flag for console module and for asterisk not to fork() off into the background. The handling of console=yes only set console mode, so you would end up with a background process() trying to run the Asterisk console and freaking out since it didn't have anything to read input from. Thanks to beagles for reporting and helping debug the problem! ........ 2010-09-22 15:11 +0000 [r288267] Tilghman Lesher * cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /, UPGRADE.txt: Merged revisions 288265-288266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010) | 9 lines Allow the encoding to be set, in case local charset does not agree with database. (closes issue #16940) Reported by: jamicque Patches: 20100827__issue16940.diff.txt uploaded by tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010) | 5 lines Document addition of encoding parameter. (issue #16940) Reported by: jamicque ........ 2010-09-22 00:03 +0000 [r288193] Richard Mudgett * channels/chan_iax2.c, /: Merged revisions 288192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) | 26 lines In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel. Near the beginning of schedule_delivery(), ast_bridged_channel() is called on iaxs[fr->callno]->owner. However, the channel is not locked, which can result in ast_bridged_channel() crashing should owner->tech change to a technology that doesn't implement bridged_channel. I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since the owner lock was not held there either. Converted the existing channel deadlock avoidance to use iax2_lock_owner(). Using the new function simplified some awkward code. In the process of fixing the locking on ast_bridged_channel(), I also found a memory leak in socket_process() for v1.6.2 and v1.8. The local struct variable ies.vars is not freed on early/abnormal function exits. (closes issue #17919) Reported by: rain Patches: issue17919_v1.4.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/926/ ........ 2010-09-21 22:22 +0000 [r288147] Paul Belanger * channels/chan_iax2.c: Setup timer before set_config(). (closes issue #18019) Reported by: Netview Patches: issue_0018019.patch uploaded by pabelanger (license 224) Tested by: Netview 2010-09-21 21:59 +0000 [r288113] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 288112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) | 15 lines Try both the encoded and unencoded subscription URI for a match in hints. When a phone sends an encoded URI for a subscription, the URI is not matched with the actual hint that is in decoded format. For example, if we have an extension with a hint that is named: "#5601" or "*5601", the subscription will work fine if the phone subscribes with an already decoded URI, but when it's decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the correct hint. (closes issue #17785) Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ 2010-09-21 19:46 +0000 [r288006] Brett Bryant * main/channel.c, /: Merged revisions 288005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) | 8 lines Add a check to fix a rare segmentation fault you'd get if ast_frdup couldn't allocate memory on the first frame being queued in ast_queue_frame. (closes issue #17882) Reported by: seanbright Tested by: seanbright ........ 2010-09-21 19:07 +0000 [r287934] Tilghman Lesher * main/asterisk.c, /: Merged revisions 287933 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010) | 2 lines Less than zero is an error, not any non-zero value. ........ 2010-09-20 23:58 +0000 [r287759] Brett Bryant * /, apps/app_meetme.c: Merged revisions 287758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a user and admin pin setup for your conference, using the user pin would gain you admin priviledges. Also, when no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the user tried to enter a conference then they were still prompted for a pin and forced to hit #. (closes issue #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: kuj Review: [full review board URL with trailing slash] ........ 2010-09-20 23:16 +0000 [r287685] Alec L Davis * main/channel.c: ast_channel_masquerade: Avoid recursive masquerades. Check all 4 combinations of (original/clonechan) * (masq/masqr). Initially original->masq and clonechan->masqr were only checked. It's possible with multiple masq's planned - and not yet executed, that the 'original' chan could already have another masq'd into it - thus original->masqr would be set, that masqr would lost. Likewise for the clonechan->masq. (closes issue #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches: based on bug16057.diff4.txt uploaded by alecdavis (license 585) Tested by: ramonpeek, davidw, alecdavis 2010-09-20 21:28 +0000 [r287642] Jason Parker * channels/chan_skinny.c: Don't crash when parking a non-bridged call. (closes issue #17680) Reported by: jmhunter Patches: chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by: jmhunter, DEA 2010-11-02 Leif Madsen * Asterisk 1.6.2.14 Released. 2010-09-20 Leif Madsen * Asterisk 1.6.2.14-rc1 Released. 2010-09-20 15:56 +0000 [r287556-287558] Matthew Nicholson * main/pbx.c, /: Use ast_str when processing hint state changes Merged revisions 287555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep 2010) | 5 lines Use ast_dynamic_str when processing hint state changes (related to issue #17928) Reported by: mdu113 ........ * /: Revert r287556. * /: Use ast_str when processing hint state changes Merged revisions 287555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep 2010) | 5 lines Use ast_dynamic_str when processing hint state changes (related to issue #17928) Reported by: mdu113 ........ 2010-09-19 16:06 +0000 [r287470] Olle Johansson * main/manager.c, /: Merged revisions 287469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 lines Make sure we always free variables properly in manager originate. (closes issue #17891) reported, solved and tested by oej Review: https://reviewboard.asterisk.org/r/869/ ........ 2010-09-17 21:08 +0000 [r287387] Tilghman Lesher * apps/app_queue.c, /: Merged revisions 287386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines Blank columns should get set on reload, not ignored. (closes issue #16893) Reported by: haakon Patches: 20100818__issue16893.diff.txt uploaded by tilghman (license 14) ........ 2010-09-17 13:36 +0000 [r287308] Matthew Nicholson * main/pbx.c, /: Merged revisions 287307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while processing in ast_hint_state_changed(). (related to issue #17928) Reported by: mdu113 ........ 2010-09-16 22:12 +0000 [r287198] Jason Parker * contrib/init.d/rc.debian.asterisk, /: Merged revisions 287197 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287197 | qwell | 2010-09-16 17:12:30 -0500 (Thu, 16 Sep 2010) | 7 lines Add LSB headers for Debian init script, since Debian will complain if it isn't there. Headers were taken from trunk. (closes issue #17958) Reported by: javyer ........ 2010-09-16 20:06 +0000 [r287115-287119] Matthew Nicholson * main/pbx.c, /: Merged revisions 287118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep 2010) | 8 lines Don't limit hint processing in ast_hint_state_changed() to AST_MAX_EXTENSION length strings. (closes issue #17928) Reported by: mdu113 Patches: 20100831__issue17928.diff.txt uploaded by tilghman (license 14) Tested by: mdu113 ........ * main/cdr.c, /: Merged revisions 287114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep 2010) | 8 lines Don't stop printing cdr variables if we encounter one with a blank name or value. (closes issue #17900) Reported by: under Patches: core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ 2010-09-15 20:28 +0000 [r286998] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 286941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines Ensure mailbox is not filled to capacity before doing message forwarding. Specifically, before prompting to record a prepended message the capacity is checked first. If the mailbox is full the extension will be reprompted. ABE-2517 ........ 2010-09-14 19:27 +0000 [r286681-286757] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 286756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines Don't clear the username from a realtime database when a registration expires. Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson ........ * main/channel.c, /: Merged revisions 286679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines Only drop duplicate answer frames if the channel is bridged. Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework. ABE-2473 (related to issue #2342) ........ 2010-09-14 05:06 +0000 [r286527-286587] Tilghman Lesher * contrib/realtime/mysql/voicemail_messages.sql (added), contrib/realtime/mysql/voicemail_data.sql (added): Add documentation on missing backend tables for Voicemail * main/features.c: C precedence got me * main/features.c: Refactor conversion to ast_poll() to fix callparking regression. 2010-09-13 19:38 +0000 [r286456] Jason Parker * channels/chan_sip.c: Remove "Internal IP" from sip show settings, as it's not at all useful to display. (closes issue #17840) Reported by: oej 2010-09-11 17:05 +0000 [r286268] Olle Johansson * /, main/file.c: Merged revisions 286267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines Handle error response when we can't make file compatible Review: https://reviewboard.asterisk.org/r/911/ ........ 2010-09-10 22:56 +0000 [r286223] Terry Wilson * channels/chan_local.c, /: Merged revisions 286222 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10 Sep 2010) | 1 line Return -1 if chan_local doesn't support an option ........ 2010-09-10 20:55 +0000 [r286117] Paul Belanger * channels/chan_iax2.c, /: Merged revisions 286114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) | 4 lines Load iax.conf before registering any functions/applications/actions. Review: https://reviewboard.asterisk.org/r/914/ ........ 2010-09-10 20:42 +0000 [r286116] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 286113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected. If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. ........ 2010-09-10 20:35 +0000 [r286115] Terry Wilson * include/asterisk/pbx.h, include/asterisk/frame.h, channels/chan_local.c, /, funcs/func_channel.c, include/asterisk/channel.h: Merged revisions 286059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ 2010-09-10 18:30 +0000 [r285930-286024] Tilghman Lesher * tests/test_heap.c, /, main/test.c: Merged revisions 286023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010) | 2 lines Missing newline ........ * include/asterisk/select.h: Another fix for Mac OS X. While trying to fix this the "right" way, I wandered into dependency hell. Two hours later, I backed out, and just removed the offending code. ast_inline_api only goes one level deep and then it breaks. Ouch. * tests/test_poll.c, include/asterisk/select.h, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 285889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines Fix Mac OS X build. This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). ........ 2010-09-09 22:49 +0000 [r285818] Paul Belanger * /, codecs/gsm/Makefile: Merged revisions 285817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep 2010) | 8 lines GCC 4.2.x optimizations result in improper behavior of GSM codec (closes issue #17688) Reported by: pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347) Tested by: mkeuter, pprindeville ........ 2010-09-09 20:09 +0000 [r285744] Jason Parker * main/channel.c, /: Merged revisions 285742 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines Transmit silence when reading DTMF in ast_readstring. Otherwise, you could get issues with DTMF timeouts causing hangups. (closes issue #17370) Reported by: makoto Patches: channel-readstring-silence-generator.patch uploaded by makoto (license 38) ........ 2010-09-09 18:50 +0000 [r285639-285710] Brett Bryant * main/pbx.c: Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis * res/res_musiconhold.c, /: Merged revisions 285638 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010) | 7 lines Fixes an issue with MOH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MOH class. (closes issue #17807) Reported by: kshumard Review: https://reviewboard.asterisk.org/r/910/ ........ 2010-09-08 22:11 +0000 [r285563-285567] David Vossel * /, channels/chan_sip.c: Merged revisions 285566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure. ........ * channels/chan_sip.c: Fixes interoperability problems with session timer behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require" header. This is not to our benefit and RFC 4028 section 7.1 even warns against it. It is possible for one endpoint to perform session-timer refreshes while the other endpoint does not support them. If in this case the end point performing the refreshing puts "timer" in the Require field during a refresh, the dialog will likely get terminated by the other end. 2. Change the behavior of 'session-timer=accept' in sip.conf (which is the default behavior of Asterisk with no session timer configuration specified) to only run session-timers as result of an incoming INVITE request if the INVITE contains an "Session-Expires" header... Asterisk is currently treating having the "timer" option in the "Supported" header as a request for session timers by the UAC. I do not agree with this. Session timers should only be negotiated in "accept" mode when the incoming INVITE supplies a "Session-Expires" header, otherwise RFC 4028 says we should treat a request containing no "Session-Expires" header as a session with no expiration. Below I have outlined some situations and what Asterisk's behavior is. The table reflects the behavior changes implemented by this patch. SITUATIONS: -Asterisk as UAS 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header 5. Outgoing INVITE: HAS "Session-Expires". Active - Asterisk will have an active refresh timer regardless if the other endpoint does. Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does. XXXXXXX - Not possible for mode. ______________________________________ |SITUATIONS | 'session-timer' MODES | |___________|________________________| | | originate | accept | |-----------|------------|-----------| |1. | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX | Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX | -------------------------------------- (closes issue #17005) Reported by: alexrecarey 2010-09-08 20:56 +0000 [r285532] Brett Bryant * apps/app_meetme.c: Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart. (closes issue #17408) Reported by: sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009) Tested by: sysreq 2010-09-08 20:42 +0000 [r285526-285529] Jason Parker * res/res_musiconhold.c, include/asterisk/astobj2.h: Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble. * res/res_musiconhold.c: Fixes issue where moh files were no longer rescanned during a reload. (closes issue #16744) Reported by: pj Patches: 16744-reload.diff uploaded by qwell (license 4) Tested by: qwell 2010-09-07 20:31 +0000 [r285267-285366] Tilghman Lesher * pbx/pbx_config.c, /: Merged revisions 285365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010) | 9 lines Catch invalid extensions at the parser, instead of making the core deal with them. (closes issue #17794) Reported by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded by tilghman (license 14) 20100820__issue17794__1.4.diff.txt uploaded by tilghman (license 14) Tested by: PavelL ........ * main/poll.c, /: Merged revisions 285266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010) | 4 lines Use poll, if indicated to do so, in the ast_poll2 implementation. This fixes the unit tests on FreeBSD 8.0. ........ 2010-09-07 17:49 +0000 [r285196] Brett Bryant * apps/app_voicemail.c, /: Merged revisions 285194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages. (closes issue #15726) Reported by: 298 Patches: M15726.diff uploaded by junky (license 177) Tested by: junky Review: [full review board URL with trailing slash] ........ 2010-09-06 06:55 +0000 [r285089] Tilghman Lesher * makeopts.in, /, BSDmakefile (added): Merged revisions 285088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010) | 2 lines Silly convenience script for BSD platforms. ........ 2010-09-03 18:15 +0000 [r284958] Brett Bryant * channels/chan_iax2.c: This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response. (closes issue #17935) Reported by: alexkuklin Patches: iaxshowreg uploaded by alexkuklin (license 1115) Tested by: alexkuklin 2010-09-03 16:20 +0000 [r284897] Terry Wilson * apps/app_chanspy.c, /: Merged revisions 284881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines Properly detect when a sound file doesn't exist ast_fileexists returns -1 for error and 0 for a non-existant file. The existing code treated missing files as though they existed. ........ 2010-09-02 20:54 +0000 [r284778] Brett Bryant * main/manager.c, /: Merged revisions 284777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010) | 7 lines Fixes a bug in manager.c where the default configuration values weren't reset when the manager configuration was reloaded. (closes issue #17917) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/883/ ........ 2010-09-02 16:48 +0000 [r284704] David Vossel * /, channels/chan_sip.c: Merged revisions 284703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines Removed relatedpeer code from sip_autodestruct Handling of the relatedpeer structure associated with a sip_pvt should be done during the final sip_destruction function, not in sip_autodestruct. ........ 2010-09-02 16:07 +0000 [r284399-284665] Tilghman Lesher * channels/chan_usbradio.c: Fixing build. * apps/app_queue.c: Don't reset queue stats on a module reload. (closes issue #17535) Reported by: raarts Patches: 20100819__issue17535.diff.txt uploaded by tilghman (license 14) * configure, include/asterisk/autoconfig.h.in: Failed to rerun bootstrap.sh after last commit * res/res_jabber.c, main/rtp.c, main/poll.c, include/asterisk/select.h (added), channels/chan_usbradio.c, channels/chan_phone.c, channels/chan_misdn.c, main/features.c, include/asterisk/poll-compat.h, tests/test_poll.c (added), main/asterisk.c, utils/clicompat.c, res/res_ais.c, /, configure.ac, channels/console_video.c, include/asterisk/channel.h: Merged revisions 284478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ * res/res_config_pgsql.c: Don't warn on floats and timestamps (closes issue #17082) Reported by: coolmig * /, channels/chan_sip.c: Merged revisions 284393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines Don't send a devstate change on poke_noanswer if the state did not change. (closes issue #17741) Reported by: schmidts Patches: chan_sip.c.patch uploaded by schmidts (license 1077) ........ 2010-08-31 18:59 +0000 [r284317] Leif Madsen * configs/say.conf.sample, /: Merged revisions 284316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines Update say.conf.sample to match the rules in say.c (closes issue #17835) Reported by: RoadKill Patches: say.conf.sample.patch.rules uploaded by RoadKill (license 933) Tested by: RoadKill ........ 2010-08-30 22:27 +0000 [r284280] Tilghman Lesher * apps/app_festival.c: Fix 3 coding errors: 1) After we close FD, we should not be trying to write to it. 2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child. 3) Use endian, not processor, detection to ensure bytes are written in the correct order. (closes issue #15706) Reported by: modelnine Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865) Tested by: gmartinez 2010-08-27 22:27 +0000 [r284002] David Vossel * /, channels/chan_sip.c: Merged revisions 283960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests. (closes issue #17758) Reported by: ibc Patches: multiple_accept_headers_1.4.diff uploaded by dvossel (license 671) ........ 2010-08-27 20:30 +0000 [r283881] Jason Parker * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged revisions 283880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | 8 lines Fix issue with decoding ^-escaped characters in realtime. (closes issue #17790) Reported by: denzs Patches: 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell, denzs ........ 2010-08-26 15:24 +0000 [r283381-283691] David Vossel * /, channels/chan_sip.c: Merged revisions 283690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response. If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc compliant and results in confusion at the other endpoint. sip_pretend_ack will ack and remove all the packets in the retransmit queue. This means that the INVITE will stop retransmitting, and that any response to that INVITE that comes after the pretend_ack occurs will be ignored. Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal hangup, we should let the protocol stack process the INVITE transaction and terminate the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag is used, once the dialog proceeds to an escapable state the transaction will either be canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do this any other way is incorrect. If the endpoint is not responding to the INVITE request, the INVITE must continue to be retransmitted until it times out which will result in the dialog being destroyed. ........ * channels/chan_sip.c: Add to and from tags to NOTIFY dialog-info xml body so pickup can occur. When pedantic mode is used, the dialog-info xml generated during a ringing event must contain the to and from tag values. Otherwise if a pickup occurs using INVITE with replaces, Astrisk will not be able to locate the subscription. * channels/chan_sip.c: Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used. Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) * /, channels/chan_sip.c: Merged revisions 283380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set. When the pending bye flag is used, it is possible that the dialog will terminate and leave the sip_pvt->owner channel up. This is because we never hangup the ast_channel after sending the SIP_BYE request. When we receive the response for the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the next do_monitor loop, but this is not the case. The dialog will only be destroyed once the owner is hungup even with the need_destroy flag set. This patch sets the softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the pending bye flag. ........ 2010-08-23 21:32 +0000 [r283318] Tilghman Lesher * cdr/cdr_odbc.c, cdr/cdr_adaptive_odbc.c: CDR drivers depend upon res_odbc, not directly on the ODBC libraries 2010-09-15 Leif Madsen * Asterisk 1.6.2.13 released. * Incorrect .version and ChangeLog files updated. Re-release of Asterisk 1.6.2.12 with corrections and version number bump. 2010-09-15 Leif Madsen * Asterisk 1.6.2.12 released. 2010-08-23 Leif Madsen * Asterisk 1.6.2.12-rc1 Released. 2010-08-20 16:48 +0000 [r283049-283124] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 283123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500 (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from https://origsvn.digium.com/svn/asterisk/trunk .......... r278274 | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1 line Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR. .......... ................ * channels/chan_dahdi.c, /: Merged revisions 283048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a protocol error The PRI layer in chan_dadhi will check if a PROGRESS message has already been sent, and not allow sending another (although that is technically allowed by the Q931 spec), however it does not protect against sending an ALERTING and then sending a PROGRESS message, which is a violation of the specification. Most switches don't seem to care too deeply about this, but some do, and will disconnect the call when receiving this invalid sequence. Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview protocol control (network side) point-point (sheet 3 of 8)" (closes issue #17874) Reported by: nic_bellamy Patches: asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) ........ 2010-08-19 21:05 +0000 [r282890-282894] David Vossel * /, channels/chan_sip.c: Merged revisions 282893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue #17712) Reported by: nickb ........ * channels/chan_sip.c: fixes sip peer memory leaks in the peer_by_ip table (issue #17798) 2010-08-19 19:44 +0000 [r282859] Matthew Nicholson * channels/chan_sip.c: Merged revisions 277944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines Regression with T.38 negotiation Prior to 1.4.26.3 T.38 negotiation worked properly, in the case of the reporter. (issue #16852) Reported by: cfc (closes issue #16705) Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ 2010-08-19 02:14 +0000 [r282730] Terry Wilson * configs/sip.conf.sample, /: Merged revisions 282729 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines Add some documentation about codec negotiation to sip.conf ........ 2010-08-18 14:28 +0000 [r282668] David Vossel * channels/chan_sip.c: fixes crash with notifycid (closes issue #17868) Reported by: francesco_r Patches: issue_17868.diff uploaded by dvossel (license 671) Tested by: francesco_r 2010-08-18 07:43 +0000 [r282607] Tilghman Lesher * channels/chan_dahdi.c: Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes. (closes issue #16770) Reported by: jamicque Patches: 20100413__issue16770.diff.txt uploaded by tilghman (license 14) 20100811__issue16770.diff.txt uploaded by tilghman (license 14) Tested by: jamicque 2010-08-17 21:35 +0000 [r282576] David Vossel * channels/chan_sip.c: fixes no default transport for temp peer creation in chan_sip (closes issue #17829) Reported by: falves11 Patches: issue_17829.rev1.txt uploaded by russell (license 2) issue_17829.diff uploaded by dvossel (license 671) Tested by: falves11 2010-08-16 18:00 +0000 [r282469] Leif Madsen * doc/tex/asterisk.tex, doc/tex/sounds.tex (added): Add information about creating sounds files using the sounds tools publically available so that others can create their own sounds prompts using the same tools we use to generate sounds releases. This allows people creating their own prompts to sound consistent with the prompts available from the open source project. SWP-595 2010-08-16 17:32 +0000 [r282467] Terry Wilson * main/channel.c, /: Merged revisions 282430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines Send a SRCCHANGE indication when we masquerade Masquerading a channel means that the src of the audio is potentially changing, so send a SRCCHANGE so that RTP-based media streams can get a new SSRC generated to reflect the change. Original patch by addix (along with lots of testing--thanks!). (closes issue #17007) Reported by: addix Patches: 1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006) srcchange.diff uploaded by twilson (license 396) Tested by: addix, twilson Review: https://reviewboard.asterisk.org/r/862/ ........ 2010-08-13 18:54 +0000 [r282235] David Vossel * channels/chan_sip.c: only do magic pickup when notifycid is enabled A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set... but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup. (closes issue #17633) Reported by: urosh Patches: chan_sip.txt uploaded by urosh (license ) blf_cid_issue.diff uploaded by dvossel (license 671) Tested by: dvossel, urosh, okrief, alecdavis 2010-08-12 22:50 +0000 [r282130] Jason Parker * pbx/pbx_config.c, /: Merged revisions 282129 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) | 1 line Register CLI commands before parsing config, in case there is a config error. ........ 2010-08-12 03:01 +0000 [r281912] Jeff Peeler * main/channel.c, /: Merged revisions 281911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines Ensure SSRC is changed when media source is changed to resolve audio delay. This change causes the SSRC to change right before the channels are bridged, which is what used to happen. It seems that fixes were made to attempt limiting SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC with this change. There are two other control frames sent in ast_channel_bridge that probably should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change up to the discretion of resolving issue #17007. For reference - old review implementing new control frame SRCCHANGE: https://reviewboard.asterisk.org/r/540 (closes issue #17404) Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler (license 325) Tested by: sdolloff ........ 2010-08-11 21:09 +0000 [r281763-281873] Leif Madsen * configs/say.conf.sample, /: Merged revisions 281819 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes issue #17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk uploaded by RoadKill (license 933) ........ * configs/say.conf.sample, /: Merged revisions 281762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010) | 6 lines Allow say.conf to handle large numbers ending with multiple zeros. (closes issue #17833) Reported by: RoadKill Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill (license 933) ........ 2010-08-11 15:17 +0000 [r281722] Tilghman Lesher * apps/app_readexten.c: Only set status TIMEOUT, if we have no digits. (closes issue #15188) Reported by: jcovert Patches: app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551) 2010-08-10 18:04 +0000 [r281567-281574] Russell Bryant * main/sched.c: Don't move the time threshold for running scheduled events on every iteration. Instead, only calculate the time threshold each time ast_sched_runq() is called. (closes issue #17742) Reported by: schmidts Patches: sched.c.patch uploaded by schmidts (license 1077) * apps/app_dial.c, /: Merged revisions 281566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines Reset visible indication after answer. (closes issue #17641) Reported by: klaus3000 Patches: ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65) Tested by: schmidts ........ 2010-08-09 20:46 +0000 [r281430] David Vossel * channels/chan_sip.c: fixes SIP peers memory leak We zeroed out the peer's addr before it was removed from the peers_by_ip container. This made it impossible to be removed from the container as the addr is the key used by the container to find the peer. (closes issue #17774) Reported by: kkm Patches: 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888) 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888) 2010-08-09 20:07 +0000 [r281391] Jeff Peeler * channels/chan_local.c, /: Merged revisions 281390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) | 13 lines Prevent loss of Caller ID information set on local channel after masquerade. Caller ID set on the channel before a masquerade occurs when using a local channel would cause the information to be lost. The problem was that the information was set on a channel destined to be hung up. The somewhat confusing fix is to detect if any Caller ID has been set on the channel and if so preswap the Caller ID data so that basically the masquerade puts the data back. (closes issue #17138) Reported by: kobaz Review: https://reviewboard.asterisk.org/r/847/ ........ 2010-08-05 13:11 +0000 [r281051] Russell Bryant * main/cdr.c: Cleanup default option value handling for cdr.conf [general]. The default values would differ depending on whether or not cdr.conf exists. That is no longer the case. Apply a default value to the unanswered option. Define all default values as named constants. 2010-08-05 07:40 +0000 [r280983] Tilghman Lesher * include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280982 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines Change context lock back to a mutex, because functionality depends upon the lock being recursive. (closes issue #17643) Reported by: zerohalo Patches: 20100726__issue17643.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ 2010-08-03 20:52 +0000 [r280671-280812] Tilghman Lesher * funcs/func_callerid.c, channels/chan_dahdi.c, /: Merged revisions 280811 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010) | 9 lines Prevent DAHDI channels from overriding the callerid, once it's been set by the user. (closes issue #16661) Reported by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: jstapleton ........ * doc/asterisk.sgml, doc/asterisk.8, doc/Makefile (added): Document -B and -W flags and regenerate manpage from sgml * apps/app_voicemail.c: Allow the pipe, but also allow the comma 2010-08-02 21:14 +0000 [r280669] Jeff Peeler * channels/chan_sip.c: Change SIP NOTIFY requests to expect a response so authentication will work. This changes the request to be sent with the transmit type XMIT_RELIABLE so that sip_ack doesn't return false and cause the 401 to be ignored in cases where authentication is required. (closes issue #14255) Reported by: zktech 2010-07-29 21:07 +0000 [r280556] Tilghman Lesher * res/res_config_curl.c: Off-by-one error (closes issue #17590) Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded by tilghman (license 14) 2010-07-29 20:42 +0000 [r280449-280551] David Vossel * channels/chan_sip.c: fixes wrong SRV query for TLS connection (closes issue #17612) Reported by: marcelloceschia Patches: chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079) Tested by: marcelloceschia, st, pabelanger * main/channel.c, /: Merged revisions 280448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines fixes issue with translator frame not getting freed A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel ........ 2010-07-29 16:01 +0000 [r280345] Jean Galarneau * /, apps/app_meetme.c: Merged revisions 280341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 ........ 2010-07-29 13:45 +0000 [r280306] Matthew Nicholson * channels/chan_local.c: Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ 2010-07-28 20:02 +0000 [r280231] Jason Parker * sounds/Makefile: Work around some silly behavior on BSD. A non-zero exit from a subshell should make the build fail. (closes issue #17621) 2010-07-28 19:57 +0000 [r280229] Richard Mudgett * channels/chan_dahdi.c: Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options. 2010-07-28 19:54 +0000 [r280193-280227] Jason Parker * build_tools/sha1sum-sh (added): Add sha1sum-sh in case there is no util on the system. * sounds/Makefile: Remove unnecessary subshells. Attempt to make checksumming work. Also improves readability. (issue #17621) Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/ 2010-07-28 16:51 +0000 [r280160] Sean Bright * apps/app_queue.c: Plug a reference leak in app_queue when adding members dynamically. (closes issue #17738) Reported by: bobwienholt Patches: issue17738.patch uploaded by bobwienholt (license 950) Tested by: bobwienholt, seanbright 2010-07-28 13:51 +0000 [r280089] Leif Madsen * contrib/scripts/live_ast, /: Merged revisions 280088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 Jul 2010) | 1 line Update help text to be less confusing. ........ 2010-07-27 20:54 +0000 [r279946] David Vossel * main/audiohook.c, main/channel.c, /, include/asterisk/audiohook.h: Merged revisions 279945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines remove empty audiohook write list on channel If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ 2010-07-27 17:54 +0000 [r279849-279883] Jason Parker * makeopts.in, configure, configure.ac: Add SHA1SUM to configure, since we require it for sounds/ * sounds/Makefile: Remove aptly-named EMPTY and BS vars, since they aren't used anymore. * sounds/Makefile: Simply sounds/Makefile some more. 2010-07-27 15:13 +0000 [r279784] Mark Michelson * channels/chan_sip.c: Fix bad behavior of dynamic_exclude_static option in sip.conf. We were attempting to create a contactdeny rule based on the peer's IP address before the peer's IP address had been set. By moving the processing further down in the function, we can ensure stuff works as we expect for it to. (closes issue #17717) Reported by: mmichelson Patches: 17717.patch uploaded by mmichelson (license 60) Tested by: DennisD 2010-07-26 22:59 +0000 [r279657] Jason Parker * sounds/Makefile (added), sounds/Makefile.380 (removed), configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381 (removed), configure.ac: Really fix sounds Makefile (and make it readableish). There was a rather large syntax error that should have caused ALL versions of GNU make to fail. I don't know how it worked. 2010-07-26 21:18 +0000 [r279609] Tilghman Lesher * configure, configure.ac: Dunno why this worked on my machine, but it works better this way. 2010-07-26 20:25 +0000 [r279597] Gavin Henry * res/res_config_ldap.c: Apply all patches in: https://issues.asterisk.org/view.php?id=13573 (closes issue #13573) Reported by: navkumar Patches: res_config_ldap-category.diff uploaded by navkumar (license 580) res_config_ldap.patch uploaded by bencer (license 961) res_config_ldap uploaded by bencer (license 961) Tested by: suretec 2010-07-26 19:15 +0000 [r279561] Tilghman Lesher * sounds/Makefile (removed), configure, sounds/Makefile.380 (added), sounds/Makefile.381 (added), configure.ac: Use a special Makefile for noobs who still have GNU Make 3.80. (Closes issue #17716) Reported by: farisraouf 2010-07-26 15:41 +0000 [r279501] Sean Bright * autoconf/ast_ext_lib.m4: Expand the correct value within AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth 2010-07-24 23:58 +0000 [r279347] Bradley Latus * doc/asterisk.8: Minor update to man page 2010-07-23 22:11 +0000 [r279207] Richard Mudgett * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines SIP promiscuous redirect could fail to dial the redirect. The ast_channel was created with one variable to ast_request() but the call to ast_call() that initiates the outgoing call was using a different variable. The two variables are not equivalent if the call_forward string included a channel technology specifier. e.g., SIP/200 ........ 2010-07-23 18:29 +0000 [r279112] Mark Michelson * channels/chan_sip.c: Backport sip_uri_params_cmp() fix from trunk to 1.6.2. 2010-07-23 18:22 +0000 [r279072-279088] Russell Bryant * /: remove old properties * /: Add branch-1.4-merged and branch-1.4-blocked properties to 1.6.2 branch. 2010-07-23 17:06 +0000 [r278983-278986] Tilghman Lesher * autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged revisions 278985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010) | 12 lines Merged revisions 278984 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010) | 5 lines Establish a maximum version for openh323 (i.e. not opal), because chan_h323 will fail to load, even if it links. (issue #17679) Reported by: am ........ ................ * main/asterisk.c, /: Merged revisions 278982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010) | 15 lines Merged revisions 278981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010) | 8 lines Avoid race with consolethread on shutdown (on parallel processors). (closes issue #17080) Reported by: sybasesql Patches: 20100721__issue17080.diff.txt uploaded by tilghman (license 14) Tested by: sybasesql ........ ................ 2010-07-23 15:23 +0000 [r278934] Tzafrir Cohen * channels/chan_dahdi.c: Two more typos to cancell. 2010-07-22 19:52 +0000 [r278709] Jeff Peeler * main/xmldoc.c, /: Merged revisions 278708 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r278708 | jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines Add method for finding XML doc files for systems that don't support GLOB_BRACE. In particular, Solaris and perhaps others do not support the above mentioned GNU extension. In this case the paths are simply expanded without the braces and the calls to glob are made separately. Note: I could not explain memory allocation failures that were being reported from within libxml itself when making calls to glob without using GLOB_NOCHECK. This is the only reason why that flag is being used. (closes issue #15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded by snuffy (license 35), modified by me ........ 2010-07-22 19:32 +0000 [r278703] Richard Mudgett * channels/chan_dahdi.c: DNID does not get cleard on a new call when using immediate=yes with ISDN signaling. When you are using chan_dahdi ISDN signaling with immediate=yes and a call comes in without a DNID then you get the DNID of a previous call. Chan_dahdi does not touch the DNID field on a new call if it does not have a DNID. Made always copy the DNID from the new call. The patches backport the relevant changes from trunk -r210387. (closes issue #17568) Reported by: wuwu Patches: issue17568_v1.4.patch uploaded by rmudgett (license 664) issue17568_v1.6.2.patch uploaded by rmudgett (license 664) 2010-08-10 Leif Madsen * Asterisk 1.6.2.11 Released. 2010-07-26 Leif Madsen * Asterisk 1.6.2.11-rc2 Released. 2010-07-26 Leif Madsen * qwell, asterisk, branch-1.6.2, r279657 *** Really fix sounds Makefile (and make it readableish). There was a rather large syntax error that should have caused ALL versions of GNU make to fail. I don't know how it worked. (Closes issue #17716) 2010-07-22 Leif Madsen * Asterisk 1.6.2.11-rc1 Released. 2010-07-22 15:00 +0000 [r278621] Mark Michelson * main/channel.c, /: Merged revisions 278620 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul 2010) | 19 lines Merged revisions 278618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul 2010) | 13 lines Allow PLC to function properly when channels use SLIN for audio. If a channel involved in a bridge was using SLIN audio, then translation paths were not guaranteed to be set up properly since in all likelihood the number of translation steps was only 1. This patch enforces the transcode_via_slin behavior if transcode_via_slin or generic_plc is enabled and one of the formats to make compatible is SLIN. AST-352 ........ ................ 2010-07-21 18:22 +0000 [r278524] Tzafrir Cohen * channels/chan_dahdi.c: Fix invalid test for rxisoffhook in FXO channels This fixes some cases of no outgoing calls on FXO before an incoming call. Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO (KS/GS) channels.In some cases the bit would not be initialized properly before the first inbound call and thus prevent an outgoing call. If those tests are actually required by anybody, they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c . (closes issue #14577) Reported by: jkroon Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd (license 610) Tested by: frawd Review: https://reviewboard.asterisk.org/r/699/ 2010-07-21 16:20 +0000 [r278479] Russell Bryant * /, res/res_timing_pthread.c: Merged revisions 278465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines Use poll() instead of select() in res_timing_pthread to avoid stack corruption. This code did not properly check FD_SETSIZE to ensure that it did not try to select() on fds that were too large. Switching to poll() removes the limitation on the maximum fd value. (closes issue #15915) Reported by: keiron (closes issue #17187) Reported by: Eddie Edwards (closes issue #16494) Reported by: Hubguru (closes issue #15731) Reported by: flop (closes issue #12917) Reported by: falves11 (closes issue #14920) Reported by: vrban (closes issue #17199) Reported by: aleksey2000 (closes issue #15406) Reported by: kowalma (closes issue #17438) Reported by: dcabot (closes issue #17325) Reported by: glwgoes (closes issue #17118) Reported by: erikje possibly other issues, too ... ........ 2010-07-21 15:58 +0000 [r278025-278464] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 278463 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 | tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11 lines Ensure realtime conferences are treated the same as static conferences when trying to find an empty one. Also, parse the useropts properly, when retrieving from realtime, and add them to the existing flags. (closes issue #17502) Reported by: kenji Patches: 20100720__issue17502.diff.txt uploaded by tilghman (license 14) Tested by: kenji ........ * apps/app_voicemail.c, /: Merged revisions 278275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500 (Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) | 7 lines Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message. (closes issue #16350) Reported by: noahisaac Patches: 20100623__issue16350.diff.txt uploaded by tilghman (license 14) ........ ................ * main/autoservice.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 278272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278272 | tilghman | 2010-07-20 17:26:23 -0500 (Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines Do not queue up DTMF frames while a call is on hold. (Fixes ABE-2110) ........ ................ * main/manager.c, /: Merged revisions 278024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010) | 14 lines Merged revisions 278023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) | 7 lines Off-by-one error (closes issue #16506) Reported by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by tilghman (license 14) ........ ................ 2010-07-19 21:21 +0000 [r277966] Jean Galarneau * /, main/features.c: Merged revisions 277945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) | 15 lines Merged revisions 277906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | 7 lines Avoid trying to pickup a parked extension before the park operation is completed. A crash could occur if the extension is picked up while the parking extension is being announced. Testing pu->notquiteyet while searching for a parked extension resolves this crash. (ABE-2418) ........ ................ 2010-07-17 17:52 +0000 [r277774-277777] Tilghman Lesher * res/res_config_pgsql.c: Merge issues... * /, autoconf/ast_func_fork.m4, configure, include/asterisk/autoconfig.h.in: Merged revisions 277775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277775 | tilghman | 2010-07-17 12:42:32 -0500 (Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) | 5 lines Remove uclibc cross-compile triplet, as uclibc has a working fork()... it's only uclinux that does not. (closes issue #17616) Reported by: pprindeville ........ ................ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged revisions 277773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010) | 15 lines Merged revisions 277568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines Since we split values at the semicolon, we should store values with a semicolon as an encoded value. (closes issue #17369) Reported by: gkservice Patches: 20100625__issue17369.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ 2010-07-16 23:37 +0000 [r277666] Tim Ringenbach * /, main/features.c: Merged revisions 277657 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul 2010) | 16 lines Merged revisions 277625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer. ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer, ast_bridge_call() is called for a second bridge on the same channel, and it clears that flag, which still needs to get set for when the original ast_bridge_call() gets control back and checks it. Review: https://reviewboard.asterisk.org/r/741 ........ ................ 2010-07-16 21:31 +0000 [r277563] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 277530 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul 2010) | 11 lines Merged revisions 277497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested. FAX-128 ........ ................ 2010-07-16 21:16 +0000 [r277489] Jeff Peeler * apps/app_queue.c, /: Merged revisions 277488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 | jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines Fix reporting estimated queue hold time. Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes. (closes issue #17498) Reported by: corruptor Patches: holdesecs_bug.diff uploaded by corruptor (license 253) ........ 2010-07-16 20:35 +0000 [r277485] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 277467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500 (Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when reloading dahdi module During a reload, the priexclusive and outsignalling parameters are not read in from the config file as intended. Unfortunately, they get set to defaults as a result. This patch makes sure that they do not get set to defaults during a reload. (closes issue #17441) Reported by: mtryfoss Patches: issue17441_v1.4.patch uploaded by rmudgett (license 664) issue17441_v1.6.2.patch uploaded by rmudgett (license 664) issue17441_trunk.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ ................ 2010-07-16 20:30 +0000 [r277478] Tilghman Lesher * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql (added), /: Merged revisions 277452 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r277452 | tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines Add documentation for MOH realtime fields ........ 2010-07-16 19:24 +0000 [r277377] Jeff Peeler * apps/app_queue.c, /: Merged revisions 277366 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 | jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines Add missing handling for ringing state for use with queue empty options. (closes issue #17471) Reported by: jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056) ........ 2010-07-16 18:33 +0000 [r277338] Matthew Nicholson * main/pbx.c, /: Merged revisions 277331 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul 2010) | 15 lines Merged revisions 277327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035) Reported by: francesco_r Patches: pbx.c.patch uploaded by viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar ........ ................ 2010-07-16 18:14 +0000 [r277264] Tilghman Lesher * main/manager.c, /: Merged revisions 277263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010) | 12 lines Merged revisions 277261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines If variable gotten is not set, will segfault on Solaris. (closes issue #17636) Reported by: bklang ........ ................ 2010-07-16 17:31 +0000 [r277256] Matthew Nicholson * main/channel.c, /: Merged revisions 277250 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277250 | mnicholson | 2010-07-16 12:30:39 -0500 (Fri, 16 Jul 2010) | 11 lines Merged revisions 277247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul 2010) | 4 lines For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation. AST-362 ........ ................ 2010-07-16 17:18 +0000 [r277188] Paul Belanger * /, apps/app_amd.c: Merged revisions 277183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul 2010) | 15 lines Merged revisions 277182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines Total analysis time error with SIP and silence suppression When using app_amd with SIP providers that have silence suppression on, the iTotalTime count increases exponentially. (closes issue #17656) Reported by: juls ........ ................ 2010-07-16 15:21 +0000 [r277144] Sean Bright * /, main/translate.c: Merged revisions 277143 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r277143 | seanbright | 2010-07-16 11:20:40 -0400 (Fri, 16 Jul 2010) | 8 lines Avoid crashing when installing a duplicate translation path with a lower cost. (closes issue #17092) Reported by: moy Patches: translate.rev254273.patch uploaded by moy (license 222) Tested by: moy ........ 2010-07-15 20:42 +0000 [r276572-276809] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 276788 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r276788 | jpeeler | 2010-07-15 15:21:03 -0500 (Thu, 15 Jul 2010) | 6 lines Correct not setting the bindport before attempting to open the socket. Related to changes from 276571, I was accidentally testing with a port set in my configuration causing me to miss this. Also moved the TCP handling as well to occur before build_peer is called. ........ * main/channel.c, /: Merged revisions 276653 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r276653 | jpeeler | 2010-07-15 08:51:11 -0500 (Thu, 15 Jul 2010) | 9 lines Merged revisions 276652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is. ........ ................ * /, channels/chan_sip.c: Merged revisions 276571 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r276571 | jpeeler | 2010-07-14 17:58:24 -0500 (Wed, 14 Jul 2010) | 21 lines Fix MWI notification transmission problems over SIP. MWI updates were not being sent if no messages were found in the event cache. This was corrected since a phone may need to clear its MWI status configured previously from another mailbox. Upon module or sip reload, MWI updates could not be sent due to the sipsock socket not being set early enough in reload_config. The code handling the descriptor assignment and such has simply been moved before the call to build_peer. Issuing a sip reload cleared the IP address of the peer, but skipped checking the database for registration information. The database is now checked both for sip reload and actually reloading the module. If a transmission occurs before the do_monitor thread has started, do not attempt to send a signal to it. (closes issue #17398) Reported by: ip-rob ........ 2010-07-14 20:16 +0000 [r276442] Kevin P. Fleming * main/loader.c, /: Merged revisions 276441 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r276441 | kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4 lines Don't try to call an embedded module's backup_globals() function until after confirming it exists. ........ 2010-07-14 11:52 +0000 [r276269] Leif Madsen * /, configs/voicemail.conf.sample: Merged revisions 276268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r276268 | lmadsen | 2010-07-14 06:51:48 -0500 (Wed, 14 Jul 2010) | 9 lines Merged revisions 276267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010) | 1 line Update documentation for voicemail.conf externpass option. ........ ................ 2010-07-13 19:11 +0000 [r276125] Russell Bryant * /, main/features.c: Merged revisions 276124 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r276124 | russell | 2010-07-13 14:09:42 -0500 (Tue, 13 Jul 2010) | 9 lines Merged revisions 276123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr instead of peer_cdr in the last commit). ........ ................ 2010-07-13 19:01 +0000 [r276121] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 276074 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010) | 19 lines Merged revisions 275773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines Make user removals and traversals thread safe in meetme. Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ ........ ................ 2010-07-13 16:55 +0000 [r275996] Russell Bryant * /, main/features.c: Merged revisions 275995 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275995 | russell | 2010-07-13 11:53:44 -0500 (Tue, 13 Jul 2010) | 21 lines Merged revisions 275994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) | 14 lines Access peer->cdr directly instead of through a saved off reference. At this point in the code, it is possible that peer_cdr may be invalid. Specifically, in the blind transfer code, CDRs are swapped between channels. So, peer_cdr is no longer == peer->cdr. The scenario that exposed a crash in this code was a blind transfer that hit the system call limit, causing the transferee channel to get destroyed after the transfer attempt failed. Even if it succeeds and this code doesn't crash, this code was still trying to reset a CDR on a channel that was now owned by a different thread, which is a BadThing(tm). (ABE-2417) ........ ................ 2010-07-13 14:49 +0000 [r275911] Tilghman Lesher * contrib/realtime/mysql, contrib/realtime/oracle, contrib/scripts/sip-friends.sql (removed), contrib/realtime/mysql/sipfriends.sql, contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql (removed), contrib/realtime/mysql/meetme.sql, contrib/realtime/sqlserver, contrib/scripts/realtime_pgsql.sql (removed), contrib/scripts/iax-friends.sql (removed), /, contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql (removed), contrib/realtime (added), contrib/realtime/postgresql, contrib/realtime/postgresql/realtime.sql: Merged revisions 275910 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275910 | tilghman | 2010-07-13 09:48:40 -0500 (Tue, 13 Jul 2010) | 9 lines Merged revisions 275909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 Jul 2010) | 2 lines Move SQL scripts into their own database-specific directories. ........ ................ 2010-07-12 17:26 +0000 [r275706] Jeff Peeler * main/channel.c, /: Merged revisions 275682 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275682 | jpeeler | 2010-07-12 12:21:01 -0500 (Mon, 12 Jul 2010) | 18 lines Merged revisions 275665 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) | 11 lines Change ast_write to not stop generator when called from ast_prod. For SIP channels configured with the progressinband option on, the ringback was being immediately stopped. This problem was due to ast_prod being moved for a deadlock fix in 259858. Prodding the channel after setting up the generator triggered the check in ast_write to stop the generator. The fix here should write the frame the same as was done before the call to ast_prod was moved. (closes issue #17372) Reported by: tech_admin ........ ................ 2010-07-12 15:38 +0000 [r275627] Leif Madsen * cdr/cdr_pgsql.c, /: Merged revisions 275626 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r275626 | lmadsen | 2010-07-12 10:37:01 -0500 (Mon, 12 Jul 2010) | 11 lines cdr_pgsql does not detect when a table is found. This change adds an ERROR message to let you know when a failure exists to get the columns from the pgsql database, which typically means that the table does not exist. (closes issue #17478) Reported by: kobaz Patches: cdr_pgsql.patch uploaded by kobaz (license 834) Tested by: kobaz, russell, lmadsen ........ 2010-07-10 15:11 +0000 [r275311-275469] Russell Bryant * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 245192 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245192 | mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21 lines Remove useless sip options related to hash table size. First off, these options weren't actually doing anything. By the time the options were parsed, the peer and dialog containers had already been allocated with their default values. Second, hash table size is something that doesn't really make sense to change in a config file. If a user is that interested in changing the hashtable size, he can modify the source itself. I have removed the parsing of the hash_peer, hash_user, and hash_dialog options. I have removed the hash_user_size variable altogether since it is not used at all. I also changed hash_peer_size and hash_dialog_size to be constant, and have changed the symbols to be in all caps as constants typically are. I have also removed the entire section in sip.conf.sample regarding configurable hashtable sizes. ........ (merge to 1.6.2 inspired by issue #17553) * /: unblock a rev * configs/features.conf.sample, /, main/features.c: Merged revisions 275424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r275424 | russell | 2010-07-09 16:57:21 -0500 (Fri, 09 Jul 2010) | 27 lines Fix some issues related to dynamic feature groups in features.conf. The bridge handling code did not properly consider feature groups when setting parameters that would affect whether or not a native bridge would be attempted. If DYNAMIC_FEATURES only include a feature group, a native bridge would occur that may prevent features from working. Fix a bug in verbose output that would show the key mapping as empty if it was using the default mapping and not a custom mapping in the feature group. Add feature groups to the output of "features show". Adjust the feature execution logic to match that of the logic when executing a feature that was not configured through a feature group. Update features.conf.sample to show that an '=' is still required if using the default key mapping from [applicationmap]. Finally, clean up a little bit of formatting to better coform to coding guidelines while in the area. (closes issue #17589) Reported by: lmadsen Patches: issue_17589.rev4.txt uploaded by russell (license 2) Tested by: russell, lmadsen ........ * /, main/features.c: Merged revisions 275310 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r275310 | russell | 2010-07-09 14:58:06 -0500 (Fri, 09 Jul 2010) | 2 lines Add missing ao2_iterator_destroy(). ........ 2010-07-09 19:23 +0000 [r275260] Paul Belanger * /, channels/chan_sip.c: Merged revisions 275249 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275249 | pabelanger | 2010-07-09 15:21:27 -0400 (Fri, 09 Jul 2010) | 15 lines Merged revisions 275241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul 2010) | 8 lines Fix logging message for stale nonce. (closes issue #17582) Reported by: kenner Patches: chan_sip.c.diff uploaded by kenner (license 1040) Tested by: lmadsen ........ ................ 2010-07-09 18:24 +0000 [r275191] Matthew Nicholson * main/loader.c, /: Merged revisions 275186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275186 | mnicholson | 2010-07-09 13:24:03 -0500 (Fri, 09 Jul 2010) | 9 lines Merged revisions 275182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul 2010) | 2 lines give a better error message when attempting to unload a module that is not loaded ........ ................ 2010-07-09 18:11 +0000 [r275148] Russell Bryant * configs/features.conf.sample, /: Merged revisions 275147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r275147 | russell | 2010-07-09 13:11:13 -0500 (Fri, 09 Jul 2010) | 2 lines Move parking lot sample config out from the middle of dynamic features sample config. ........ 2010-07-09 17:51 +0000 [r275029-275145] Matthew Nicholson * main/loader.c, /: Merged revisions 275144 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul 2010) | 9 lines Merged revisions 275143 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul 2010) | 2 lines don't unload modules that returned AST_MODULE_LOAD_DECLINE when they were loaded ........ ................ * apps/app_dial.c, /: Merged revisions 275028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul 2010) | 15 lines Merged revisions 275027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial (closes issue #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson ........ ................ 2010-07-09 15:39 +0000 [r275023] Russell Bryant * include/asterisk/test.h, /, main/test.c: Merged revisions 275022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275022 | russell | 2010-07-09 10:35:53 -0500 (Fri, 09 Jul 2010) | 11 lines Merged revisions 275021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines Document that a leading and trailing slash is expected for test categories. Also, emit a warning if a test is registered without one of these. ........ ................ 2010-07-07 18:34 +0000 [r274627-274640] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 274639 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r274639 | rmudgett | 2010-07-07 13:32:35 -0500 (Wed, 07 Jul 2010) | 1 line Add missing conditional around chan_dahdi mfcr2_skip_category config parameter. ........ * channels/chan_dahdi.c, /: Merged revisions 274595 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274595 | rmudgett | 2010-07-07 13:20:00 -0500 (Wed, 07 Jul 2010) | 9 lines Merged revisions 274579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter. ........ ................ 2010-07-07 06:16 +0000 [r274419] Tilghman Lesher * configs/say.conf.sample, /: Merged revisions 274418 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274418 | tilghman | 2010-07-07 01:15:43 -0500 (Wed, 07 Jul 2010) | 15 lines Merged revisions 274417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010) | 8 lines Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers. (closes issue #16102) Reported by: Delvar Patches: say.conf.fix.patch uploaded by Delvar (license 908) (plus a few additional fixes and simplifications by me) ........ ................ 2010-07-06 23:06 +0000 [r274360] Terry Wilson * configs/sip.conf.sample, channels/chan_sip.c: Merged revisions 274284 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010) | 18 lines Merged revisions 274280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines Add option to not do a call forward on 482 Loop Detected Asterisk has always set up a forwarded call when receiving a 482 Loop Detected. This prevents handling the call failure by just continuing on in the dialplan. Since this would be a change in behavior, the new option to disable this behavior is forwardloopdetected which defaults to 'yes'. Review: https://reviewboard.asterisk.org/r/764/ ........ ................ 2010-07-06 22:30 +0000 [r274347] Jeff Peeler * configs/sip.conf.sample, /: Merged revisions 274316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274316 | jpeeler | 2010-07-06 17:23:35 -0500 (Tue, 06 Jul 2010) | 14 lines Merged revisions 274283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines Correct sip.conf.sample comments for prematuremedia option. (closes issue #17513) Reported by: festr Patches: patch uploaded by festr (license 443) ........ ................ 2010-07-06 22:10 +0000 [r274282] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 274281 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r274281 | tilghman | 2010-07-06 17:09:23 -0500 (Tue, 06 Jul 2010) | 2 lines Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the bitfield. ........ 2010-07-06 14:33 +0000 [r274168] Mark Michelson * main/rtp.c, /: Merged revisions 274164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274164 | mmichelson | 2010-07-06 09:31:13 -0500 (Tue, 06 Jul 2010) | 22 lines Merged revisions 274157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines Fix problem with RFC 2833 DTMF not being accepted. A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order. The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the seqno rolling over will not cause us to stop accepting DTMF. (closes issue #17571) Reported by: mdeneen Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license 60) Tested by: richardf, maxochoa, JJCinAZ ........ ................ 2010-07-05 13:55 +0000 [r273888] Paul Belanger * main/config.c, /: Merged revisions 273886 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273886 | pabelanger | 2010-07-05 09:53:44 -0400 (Mon, 05 Jul 2010) | 15 lines Merged revisions 273884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul 2010) | 8 lines Remove extra line breaks from 'core show config mappings' (closes issue #17583) Reported by: pabelanger Patches: issue17583.patch uploaded by pabelanger (license 224) Tested by: lmadsen ........ ................ 2010-07-03 02:43 +0000 [r273716-273831] Tilghman Lesher * channels/chan_local.c, /, channels/chan_agent.c, channels/chan_h323.c, include/asterisk/lock.h: Merged revisions 273830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010) | 16 lines Merged revisions 273793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ ................ * main/autoservice.c, /: Merged revisions 273718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273718 | tilghman | 2010-07-02 12:10:59 -0500 (Fri, 02 Jul 2010) | 15 lines Merged revisions 273717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup. (closes issue #17564) Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ ................ * apps/app_queue.c, /: Merged revisions 273714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273714 | tilghman | 2010-07-02 11:57:28 -0500 (Fri, 02 Jul 2010) | 2 lines The switch fallthrough could create some errorneous situations, so best to force directly to the default case. ........ 2010-07-02 15:59 +0000 [r273642] Tzafrir Cohen * channels/chan_iax2.c, apps/app_voicemail.c, channels/chan_dahdi.c, channels/chan_sip.c, res/res_agi.c: Fix typos reported by Lintian 2010-07-01 22:17 +0000 [r273571] Russell Bryant * main/datastore.c, /: Merged revisions 273566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273566 | russell | 2010-07-01 17:16:23 -0500 (Thu, 01 Jul 2010) | 14 lines Merged revisions 273565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines Don't return a partially initialized datastore. If memory allocation fails in ast_strdup(), don't return a partially initialized datastore. Bad things may happen. (related to ABE-2415) ........ ................ 2010-07-01 20:29 +0000 [r273356-273529] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 273522 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273522 | jpeeler | 2010-07-01 15:28:15 -0500 (Thu, 01 Jul 2010) | 21 lines Merged revisions 273474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines Allow admin user to join conference without using admin mode and no user pin. Configuring the conference in meetme.conf like the following: conf => 2345,,6666 did not prompt for pin when used without admin mode. This meant that the conference could not be joined as an admin even if the user knew the correct pin. The original bug report was submitted claiming that the blank user pin should deny entry into the conference. I think a better way to handle this would be with a feature enhancement that used the following syntax: conf => 2345,X,6666 - where X denotes no acceptable pin allowed (closes issue #15704) Reported by: modelnine ........ ................ * /, apps/app_meetme.c: Merged revisions 273355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273355 | jpeeler | 2010-07-01 10:12:31 -0500 (Thu, 01 Jul 2010) | 19 lines Merged revisions 273354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. Specifically checking for this scenario by looking at the appropriate control frames resolves the issue. (closes issue #15871) Reported by: Ivan Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229) ........ ................ 2010-07-01 14:39 +0000 [r273271-273353] Matthew Nicholson * main/manager.c, /: Merged revisions 273352 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273352 | mnicholson | 2010-07-01 09:37:37 -0500 (Thu, 01 Jul 2010) | 2 lines Fixed whitespace problems ........ * main/manager.c, /: Merged revisions 273350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273350 | mnicholson | 2010-07-01 09:34:31 -0500 (Thu, 01 Jul 2010) | 2 lines Altered my comment about TCP_NODELAY ........ * main/manager.c, /: Merged revisions 273270 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273270 | mnicholson | 2010-06-30 13:48:21 -0500 (Wed, 30 Jun 2010) | 2 lines Set TCP_NODELAY on manager TCP sockets to prevent delays on outgoing packets. This regression was introduced in r48338. AST-359 ........ 2010-06-30 17:32 +0000 [r273193-273234] Paul Belanger * main/rtp.c, /: Merged revisions 273233 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273233 | pabelanger | 2010-06-30 13:28:04 -0400 (Wed, 30 Jun 2010) | 11 lines Fix rt(c)p set debug ip taking wrong argument Also clean up some coding errors. (closes issue #17469) Reported by: wdoekes Patches: astsvn-rtp-set-debug-ip.patch uploaded by wdoekes (license 717) Tested by: wdoekes, pabelanger ........ * /: Revert previous commit; res_rtp_asterisk.c does not exist. * /: Unblock revisions 218107 ........ r218107 | mvanbaak | 2009-09-12 15:08:16 +0200 (Sat, 12 Sep 2009) | 8 lines use the actual given ip address for 'rtp set debug ip ' instead of the word 'ip' (closes issue 0015711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) Tested by: davidw ........ 2010-06-30 01:07 +0000 [r273056-273145] Tilghman Lesher * main/manager.c, /: Merged revisions 273144 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273144 | tilghman | 2010-06-29 20:07:02 -0500 (Tue, 29 Jun 2010) | 8 lines Permission checking for the system application is backwards. (closes issue #17550) Reported by: kenner Patches: manager.c.diff uploaded by kenner (license 1040) Tested by: kenner ........ * main/config.c, /: Merged revisions 273142 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273142 | tilghman | 2010-06-29 20:01:14 -0500 (Tue, 29 Jun 2010) | 5 lines Don't attempt to proceed if our internal parser indicates an invalid file. (closes issue #17560) Reported by: Nick_Lewis ........ * /, channels/chan_sip.c: Merged revisions 273078 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273078 | tilghman | 2010-06-29 18:20:40 -0500 (Tue, 29 Jun 2010) | 17 lines Merged revisions 273060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines Allow the "useragent" value to be restored into memory from the realtime backend. This value is purely informational. It does not alter configuration at all. (closes issue #16029) Reported by: Guggemand Patches: realtime-useragent.patch uploaded by Guggemand (license 897) Tested by: Guggemand ........ ................ * main/channel.c, /: Merged revisions 273058 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273058 | tilghman | 2010-06-29 17:59:51 -0500 (Tue, 29 Jun 2010) | 11 lines Recorded merge of revisions 273057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) | 4 lines _Really_ skip the channel... don't just retry for another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........ ................ * main/pbx.c, /: Merged revisions 273054 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273054 | tilghman | 2010-06-29 17:39:22 -0500 (Tue, 29 Jun 2010) | 11 lines Send DialPlanComplete as a response, not as a separate event. Otherwise, it goes to all manager sessions and may exclude the current session, if the Events mask excludes it. (closes issue #17504) Reported by: rrb3942 Patches: showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested by: rrb3942 ........ 2010-06-29 16:43 +0000 [r272972] Russell Bryant * main/asterisk.c, /: Merged revisions 253357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253357 | russell | 2010-03-18 13:18:43 -0500 (Thu, 18 Mar 2010) | 8 lines Increase CLI command output timeout for asterisk -rx to 60 seconds. (closes issue #17049) Reported by: russell Tested by: russell Review: https://reviewboard.asterisk.org/r/573/ ........ 2010-07-22 Leif Madsen * Release Asterisk 1.6.2.10 * Included a fix for res_timing_pthread per the description below: r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines Use poll() instead of select() in res_timing_pthread to avoid stack corruption. This code did not properly check FD_SETSIZE to ensure that it did not try to select() on fds that were too large. Switching to poll() removes the limitation on the maximum fd value. 2010-07-07 Leif Madsen * Release Asterisk 1.6.2.10-rc2 * Fix problem with RFC 2833 DTMF not being accepted. A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order. The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the seqno rolling over will not cause us to stop accepting DTMF. (closes issue 0017571) Reported by: mdeneen Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license 60) Tested by: richardf, maxochoa, JJCinAZ * Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial (closes issue 0017592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson 2010-06-29 Leif Madsen * Release Asterisk 1.6.2.10-rc1 2010-06-28 21:51 +0000 [r272924-272927] Tilghman Lesher * main/asterisk.c, /: Merged revisions 272926 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272926 | tilghman | 2010-06-28 16:50:57 -0500 (Mon, 28 Jun 2010) | 15 lines Merged revisions 272925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) | 8 lines Don't change ownership/group/permissions on run directory, if it already exists. (closes issue #17076) Reported by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by tilghman (license 14) Tested by: stuarth ........ ................ * main/config.c, /: Merged revisions 272923 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272923 | tilghman | 2010-06-28 16:42:52 -0500 (Mon, 28 Jun 2010) | 19 lines Merged revisions 272921-272922 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010) | 8 lines Change the way that we read include files, to accommodate for changes in GCC 4.4. (closes issue #17472) Reported by: seandarcy Patches: config2.patch uploaded by nivan (license 1066) Tested by: nivan ........ r272922 | tilghman | 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim trailing blanks on #includes ........ ................ 2010-06-28 18:50 +0000 [r272882] Russell Bryant * tests/test_astobj2.c (added): Backport applicable parts of test_astobj2 from trunk. 2010-06-28 17:37 +0000 [r272806] Mark Michelson * /, channels/chan_sip.c: Merged revisions 272805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272805 | mmichelson | 2010-06-28 12:33:12 -0500 (Mon, 28 Jun 2010) | 11 lines Merged revisions 272804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun 2010) | 5 lines Decode URI in contact header of 302 response. ABE-2352 ........ ................ 2010-06-28 15:36 +0000 [r272685-272686] Russell Bryant * doc/tex/chan-mobile.tex (removed): remove accidentally added file. * doc/tex/cdrdriver.tex, doc/tex/asterisk.tex, /, doc/tex/chan-mobile.tex (added): Merged revisions 272684 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272684 | russell | 2010-06-28 10:33:32 -0500 (Mon, 28 Jun 2010) | 2 lines Use the underscore package so that underscores do not need to be escaped. ........ 2010-06-25 20:20 +0000 [r272556-272577] Tilghman Lesher * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272568 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272568 | tilghman | 2010-06-25 15:18:47 -0500 (Fri, 25 Jun 2010) | 12 lines Merged revisions 272562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines Make the structure of the table specified before match the queries and results. (closes issue #17557) Reported by: cmaj ........ ................ * sounds/Makefile, /: Merged revisions 272533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272533 | tilghman | 2010-06-25 14:17:16 -0500 (Fri, 25 Jun 2010) | 2 lines Symlink sounds files, to save disk space, when multiple tarballs/checkouts are on the same system. ........ 2010-06-25 18:58 +0000 [r272531] Russell Bryant * include/asterisk/_private.h, tests/test_sched.c, main/asterisk.c, include/asterisk/test.h (added), build_tools/cflags-devmode.xml, tests/test_heap.c, tests/test_skel.c, main/Makefile, main/test.c (added): Backport unit test API from trunk. Also, update existing test modules that were already in this branch but had been converted to the unit test API in trunk. Review: https://reviewboard.asterisk.org/r/748/ 2010-06-24 22:19 +0000 [r272459] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 272447 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272447 | rmudgett | 2010-06-24 17:11:26 -0500 (Thu, 24 Jun 2010) | 17 lines Merged revisions 272446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines ss_thread calls pri_grab without lock during overlap dial Recent changes to chan_dahdi with relation to overlap dialing call pri_grab without first obtaining a lock. (closes issue #17414) Reported by: pdf Patches: bug17414.patch uploaded by jpeeler (license 325) ........ ................ 2010-06-23 23:40 +0000 [r272440] Terry Wilson * autoconf/ast_ext_tool_check.m4, /, configure: Merged revisions 272254,272256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272254 | twilson | 2010-06-23 15:53:48 -0500 (Wed, 23 Jun 2010) | 10 lines Honor the --with-${library}=path for AST_EXT_TOOL_CHECK (closes issue #16991) Reported by: pprindeville Patches: with_netsnmp.patch.txt uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/739/ ........ r272256 | twilson | 2010-06-23 15:59:17 -0500 (Wed, 23 Jun 2010) | 2 lines Update configure when changing autconf m4 files... ........ 2010-06-23 23:14 +0000 [r272371] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 272370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272370 | russell | 2010-06-23 18:09:28 -0500 (Wed, 23 Jun 2010) | 23 lines Resolve some errors produced during module unload of chan_iax2. The external test suite stops Asterisk using the "core stop gracefully" command. The logs from the tests show that there are a number of problems with Asterisk trying to cleanly shut down. This patch addresses the following type of error that comes from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy: chan_iax2.c line 11371 (iax2_process_thread_cleanup): Error destroying mutex &thread->lock: Device or resource busy For an example in the context of a build, see: http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary purpose of this patch is to change the thread pool shutdown procedure to be more explicit to ensure that the thread exits from a point where it is not holding a lock. While testing that, I encountered various crashes due to the order of operations in unload_module() being problematic. I reordered some things there, as well. Review: https://reviewboard.asterisk.org/r/736/ ........ 2010-06-23 22:37 +0000 [r272369] Matthew Nicholson * apps/app_queue.c, /: Merged revisions 272368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272368 | mnicholson | 2010-06-23 17:36:49 -0500 (Wed, 23 Jun 2010) | 16 lines Merged revisions 272367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers. ........ r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines Send AgentComplete manager events in the event of blind and attended transfers. (closes issue #16819) Reported by: elbriga Patches: app_queue.diff uploaded by elbriga (license 482) ........ ................ 2010-06-23 21:54 +0000 [r272333] Tilghman Lesher * res/res_musiconhold.c, /: Merged revisions 272332 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272332 | tilghman | 2010-06-23 16:53:49 -0500 (Wed, 23 Jun 2010) | 8 lines If there is realtime configuration, it does not get re-read on reload unless the config file also changes. (closes issue #16982) Reported by: dmitri Patches: res_musiconhold.patch uploaded by dmitri (license 1001) Tested by: atis ........ 2010-06-23 21:15 +0000 [r272263] Paul Belanger * apps/app_meetme.c: Revert previous commit, ast_test_flag64 does not exist in 1.6.2 2010-06-23 21:09 +0000 [r272262] Tilghman Lesher * res/ael/ael.flex, /, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael_lex.c: Merged revisions 272260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272260 | tilghman | 2010-06-23 16:06:40 -0500 (Wed, 23 Jun 2010) | 8 lines Ensure a NULL file while debugging cannot crash AEL. (closes issue #17215) Reported by: vazir Patches: 20100518__issue17215.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ 2010-06-23 21:07 +0000 [r272253-272261] Paul Belanger * /, apps/app_meetme.c: Merged revisions 272259 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272259 | pabelanger | 2010-06-23 17:06:15 -0400 (Wed, 23 Jun 2010) | 2 lines Fix previous merge. ast_test_flag != ast_test_flag64 ........ * /, apps/app_meetme.c: Merged revisions 272257 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272257 | pabelanger | 2010-06-23 17:00:00 -0400 (Wed, 23 Jun 2010) | 19 lines Merged revisions 272255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines First caller into a dynamic conference now enter pin once. If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ ................ * main/manager.c, /: Merged revisions 272252 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272252 | pabelanger | 2010-06-23 16:35:45 -0400 (Wed, 23 Jun 2010) | 8 lines Correct manager variable 'EventList' case. (closes issue #17520) Reported by: kobaz Patches: manager.patch uploaded by kobaz (license 834) Tested by: lmadsen ........ 2010-06-23 18:41 +0000 [r272124-272149] Terry Wilson * /, apps/app_meetme.c: Merged revisions 272146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272146 | twilson | 2010-06-23 13:39:20 -0500 (Wed, 23 Jun 2010) | 2 lines Don't start the sla thread unless we realy need it ........ * /, apps/app_meetme.c: Merged revisions 272109 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272109 | twilson | 2010-06-23 12:21:40 -0500 (Wed, 23 Jun 2010) | 12 lines Make sure reload updates SLA config Even if there are no stations or trunks defined, we need to start the sla thread to make sure we get the reload event. Also, when doing a reload we need to remove the existing trunks and stations or they end up hanging around. (closes issue #16818) Reported by: mbonin Patches: sla_reload.patch uploaded by twilson (license 396) Tested by: twilson ........ 2010-06-22 22:14 +0000 [r272015] David Vossel * pbx/pbx_config.c, /: Merged revisions 272014 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272014 | dvossel | 2010-06-22 17:11:50 -0500 (Tue, 22 Jun 2010) | 5 lines fixes issue with 'dialplan remove extension blah' segfaulting with tab completion (closes issue #17440) Reported by: kobaz ........ 2010-06-22 17:37 +0000 [r271904] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 271903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun 2010) | 15 lines Merged revisions 271902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct. (closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain ........ ................ 2010-06-22 16:30 +0000 [r271869] Russell Bryant * /, res/ais/clm.c, res/ais/evt.c: Merged revisions 271867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r271867 | russell | 2010-06-22 11:28:03 -0500 (Tue, 22 Jun 2010) | 7 lines Resolve some errors that occur on a graceful shutdown. Don't Finalize() if Initialize() did not succeed. This resulted in an error about trying to Finalize() an invalid handle. Also trim some trailing whitespace while in the area. ........ 2010-06-22 15:49 +0000 [r271832] David Vossel * /, main/features.c: Merged revisions 271831 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r271831 | dvossel | 2010-06-22 10:46:22 -0500 (Tue, 22 Jun 2010) | 10 lines fixes attended transfer behavior when both transferee and transferer hung up If both the transferer and transferee of a attended transfer hangup before the new channel picks up, the new channel should be hung up as well as it has no endpoint to talk to. This mirrors the expected behavior used in 1.4. (closes issue #17444) Reported by: corruptor ........ 2010-06-22 15:00 +0000 [r271691-271763] Matthew Nicholson * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions 271762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271762 | mnicholson | 2010-06-22 09:54:58 -0500 (Tue, 22 Jun 2010) | 15 lines Merged revisions 271761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines Allow users to specify a port for dundi peers. (closes issue #17056) Reported by: klaus3000 Patches: dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ * include/asterisk/strings.h, configs/sip_notify.conf.sample, /, channels/chan_sip.c: Merged revisions 271690 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun 2010) | 18 lines Merged revisions 271689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct. (closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ ........ This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls. ................ 2010-06-21 20:48 +0000 [r271555] Jeff Peeler * res/ael/pval.c, /: Merged revisions 271554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271554 | jpeeler | 2010-06-21 15:46:53 -0500 (Mon, 21 Jun 2010) | 14 lines Merged revisions 271552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines Do not use sizeof to calculate size of a heap allocated character array. Change left out from 271399. (closes issue #16053) Reported by: diLLec ........ ................ 2010-06-18 21:33 +0000 [r271338-271484] Jeff Peeler * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged revisions 271483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010) | 18 lines Merged revisions 271399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines Fix crash when parsing some heavily nested statements in AEL on reload. Due to the recursion used when compiling AEL in gen_prios, all the stack space was being consumed when parsing some AEL that contained nesting 13 levels deep. Changing a few large buffers to be heap allocated fixed the crash, although I did not test how many more levels can now be safely used. (closes issue #16053) Reported by: diLLec Tested by: jpeeler ........ ................ * channels/chan_dahdi.c, /: Merged revisions 269307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010) | 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are in a potentially inconsistent state is a potentially bad thing in principle. However, calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. ........ 2010-06-17 Leif Madsen * Asterisk 1.6.2.9 Released. 2010-06-10 Leif Madsen * Asterisk 1.6.2.9-rc3 Released. 2010-06-10 Tilghman Lesher * Ensure signals are not blocked inside other signal handlers. This eliminates the annoying on the console. (closes issue 0017477) Reported by: jvandal Patches: 20100610__issue17477.diff.txt uploaded by tilghman (license 14 2010-06-09 Paul Belanger * Fix Debian init script to not use -c. When using the init script as-is currently, it could cause issues on Debian such as high CPU usage. This fix has worked for several people so I'm implementing the change. We now handle color displays properly. (closes issue 0016784) Reported by: pabelanger Patches: 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14) Tested by: pabelanger, tilghman 2010-06-07 Leif Madsen * Asterisk 1.6.2.9-rc2 Released. 2010-06-07 Tilghman Lesher * Fix crash in DTMF detection. What I did not originally see in my previous commit was that even though the next digit could be detected before the previous was considered ended, the detection of the next digit effectively ends the detection of the previous. Therefore, the length moves in lockstep with the digit, and no separate counter is needed for the length alone. (closes issue 0017371) Reported by: alecdavis (closes issue 0017474) Reported by: kenner 2010-06-01 Leif Madsen * Asterisk 1.6.2.9-rc1 Released. 2010-06-01 15:20 +0000 [r266598] Tilghman Lesher * main/asterisk.c, /: Merged revisions 266592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010) | 18 lines Merged revisions 266585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) | 11 lines Prevent CLI prompt from distorting output of lines shorter than the prompt. Uses the VT100 method of clearing the line from the cursor position to the end of the line: Esc-0K (closes issue #17160) Reported by: coolmig Patches: 20100531__issue17160.diff.txt uploaded by tilghman (license 14) Tested by: coolmig ........ ................ 2010-05-31 16:07 +0000 [r266570] Paul Belanger * res/res_agi.c: Fix typo in documentation (closes issue #17395) Reported by: pabelanger Patches: res_agi.c.patch uploaded by pabelanger (license 224) 2010-05-30 04:45 +0000 [r266439] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r266438 | tilghman | 2010-05-29 23:44:28 -0500 (Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 May 2010) | 2 lines Reverting patch and reopening issue #16784, as patch breaks color display. ........ ................ 2010-05-28 20:55 +0000 [r266338] Tilghman Lesher * main/asterisk.c, /: Merged revisions 266337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 | tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line Only report swap on platforms which can examine those statistics ........ 2010-05-28 17:57 +0000 [r266293] David Vossel * /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 | dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines fixes crash when creation of UDPTL fails (closes issue #17264) Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff uploaded by dvossel (license 671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel (license 671) Tested by: falves11 ........ 2010-05-26 21:19 +0000 [r266154] Tilghman Lesher * utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged revisions 266146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010) | 21 lines Merged revisions 266142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) | 14 lines Use sigaction for signals which should persist past the initial trigger, not signal. If you call signal() in a Solaris signal handler, instead of just resetting the signal handler, it causes the signal to refire, because the signal is not marked as handled prior to the signal handler being called. This effectively causes Solaris to immediately exceed the threadstack in recursive signal handlers and crash. (closes issue #17000) Reported by: rmcgilvr Patches: 20100526__issue17000.diff.txt uploaded by tilghman (license 14) Tested by: rmcgilvr ........ ................ 2010-05-26 18:37 +0000 [r266007] David Vossel * /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 | dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines fixes failed SIP Directed pickup resulting in dead channel (closes issue #17339) Reported by: one47 Patches: sip_magic_pickup2 uploaded by one47 (license 23) Tested by: one47, dvossel ........ 2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher * res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r265923 | tilghman | 2010-05-26 11:23:28 -0500 (Wed, 26 May 2010) | 14 lines Merged revisions 265910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010) | 7 lines Not finding rows in the DB does not rise to the level of a warning. (closes issue #17062) Reported by: drookie Patches: 20100525__issue17062.diff.txt uploaded by tilghman (license 14) ........ ................ * configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged revisions 265894 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 | tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines Construct socket name, according to the Postgres docs, and document as such. (closes issue #17392) Reported by: dps Patches: 20100525__issue17392.diff.txt uploaded by tilghman (license 14) Tested by: dps ........ 2010-05-26 15:52 +0000 [r265890] Mark Michelson * /, channels/chan_sip.c: Recorded merge of revisions 265842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed, 26 May 2010) | 9 lines Re-enable "always" option for videosupport option in sip.conf. (closes issue #17016) Reported by: twilson Patches: 17016.patch uploaded by mmichelson (license 60) Tested by: devmod ........ 2010-05-26 00:33 +0000 [r265748] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 | tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines Use configure to determine the prefixes and include directories properly. This ensures cross-platform compatibility, even among Linux distributions, which don't always put headers in the same place. (closes issue #17391) Reported by: loloski ........ 2010-05-25 21:05 +0000 [r265699] Mark Michelson * /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 | mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12 lines Properly use peer's outboundproxy for outbound REGISTERs. The logic used in transmit_register to get the outboundproxy for a peer was flawed since this value would be overridden shortly afterwards when create_addr was called. In addition, this also fixes some logic used when parsing users.conf so that the peer name is placed in the internally-generated register string so that an outboundproxy set in the Asterisk GUI will be used for outbound REGISTERs. ........ 2010-05-25 17:15 +0000 [r265615] David Vossel * channels/chan_dahdi.c: fixes build issue with zaptel (closes issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff uploaded by dvossel (license 671) Tested by: aragon 2010-05-25 17:06 +0000 [r265612] Matthew Nicholson * apps/app_queue.c, /: Merged revisions 265611 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May 2010) | 15 lines Merged revisions 265610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal ........ ................ 2010-05-24 23:52 +0000 [r265521] Terry Wilson * include/asterisk/options.h, main/asterisk.c, Makefile, doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged revisions 265320,265467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines Add the FullyBooted AMI event It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ ........ r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line Merge the rest of the FullyBooted patch ........ 2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson * /, channels/h323/ast_h323.cxx: Merged revisions 265451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon, 24 May 2010) | 8 lines Print openh323 log to the Asterisk console. (closes issue #17109) Reported by: under Patches: logstream.diff uploaded by under (license 914) ........ * /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 | mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11 lines Allow type=user SIP endpoints to be loaded properly from realtime. (closes issue #16021) Reported by: Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand (license 897) (altered by me slightly to avoid ref leaks) Tested by: Guggemand ........ 2010-05-24 19:30 +0000 [r265364] David Vossel * main/channel.c, /: Merged revisions 265273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 | dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines fixes segfault when using generic plc ........ 2010-05-24 18:30 +0000 [r265318] Tilghman Lesher * main/asterisk.c, /: Merged revisions 265316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 | tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines On systems with a LOT of RAM, a signed integer sometimes printed negative. (closes issue #16837) Reported by: jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by tilghman (license 14) ........ 2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson * apps/app_queue.c: Fix memory hogging behavior of app_queue. From reviewboard: This review request is for the patch on issue 17081. A user reported that he saw increasing numbers of allocations stemming from app_queue.c when he would run the "queue show" CLI command. The user reported that he was using approximately 40 realtime queues and as he ran the CLI command more and more, the memory usage would shoot up. As it turns out, there was a memory leak and a separate usage of memory that, while not really a leak, was very irresponsible. Both memory problems can be attributed to the function init_queue(). When the "queue show" command is run, all realtime queues have the init_queue() function called on the in-memory queue. The idea is to place the queue in its default state and then overwrite options specified in the realtime backend as we read them. The first problem, the memory leak, had to do with the fact that the string field for the name of the first periodic announcement file was being re-created every time init_queue was called. This patch corrects the behavior by only calling ast_str_create if the memory has not already been allocated. The other problem is a bit more complicated. The majority of the strings in the call_queue structure were changed to use the ast_string_fields API for 1.6.0 and beyond. init_queue resets all string fields on the queue to their default values. Then, later in the realtime queue loading process, these string fields are set to their configured values. For those unfamiliar with string fields, frequent resizing of a string like this is not what the string fields API is designed for. The result of this constant resizing is that as the queue gets loaded, eventually space for the string runs out and so a new memory pool, at twice the size of the previously allocated one, is created for the string fields. The reporter of issue 17081 wrote a script that ran the "queue show" CLI command 2100 times. By the end, each of his 40 queues was taking about a megabyte of memory apiece just for their string fields. My fix for this problem is to revert the call_queue structure from using string fields. In my patch here, I have moved the queue back to using fixed-sized buffers. I ran the script provided by the reporter of 17081 and determined that I no longer saw the steadily-increasing memory usage that I had seen before applying the patch. (closes issue #17081) Reported by: wliegel Patches: 17081v2.patch uploaded by mmichelson (license 60) Tested by: wliegel, mmichelson Review: https://reviewboard.asterisk.org/r/651/ * apps/app_queue.c, include/asterisk/file.h, /: Merged revisions 265090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May 2010) | 15 lines Merged revisions 265089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines Don't hang up on a queue caller if the file we attempt to play does not exist. This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill ........ ................ * /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 | mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7 lines Be sure to set the sin_family on the proxy when allocating. (closes issue #17157) Reported by: stuarth ........ * /, include/asterisk/channel.h: Merged revisions 265000 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500 (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May 2010) | 3 lines Fix grammatical error in comment. ........ ................ * main/channel.c, main/autoservice.c, /, include/asterisk/channel.h: Merged revisions 264997 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500 (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific frames until after the sleep has concluded. From reviewboard Background: A Digium customer discovered a somewhat odd bug. The setup is that parties A and B are bridged, and party A places party B on hold. While party B is listening to hold music, he mashes a bunch of DTMF. Party A takes party B off hold while this is happening, but party B continues to hear hold music. I could reproduce this about 1 in 5 times. The issue: When DTMF features are enabled and a user presses keys, the channel that the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read from the channel during the sleep, the frame is dropped. Thus the unhold indication is never made to the channel that was originally placed on hold. The fix: Originally, I discussed with Kevin possible ways of fixing the specific problem reported. However, we determined that the same type of problem could happen in other situations where ast_safe_sleep() is used. Using autoservice as a model, I modified ast_safe_sleep_conditional() to defer specific frame types so they can be re-queued once the sleep has finished. I made a common function for determining if a frame should be deferred so that there are not two identical switch blocks to maintain. Review: https://reviewboard.asterisk.org/r/674/ ........ ................ 2010-05-20 23:34 +0000 [r264829] Richard Mudgett * /, main/callerid.c: Merged revisions 264828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010) | 13 lines Merged revisions 264820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010) | 6 lines ast_callerid_parse() had a path that left name uninitialized. Several callers of ast_callerid_parse() do not initialize the name parameter before calling thus there is the potential to use an uninitialized pointer. ........ ................ 2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher * main/pbx.c, /: Merged revisions 264779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 | tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines Let ExtensionState resolve dynamic hints. (closes issue #16623) Reported by: tilghman Patches: 20100116__issue16623.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ * apps/app_stack.c, /: Merged revisions 264752 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 | tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines Error message fix. (closes issue #17356) Reported by: kenner Patches: app_stack.c.diff uploaded by kenner (license 1040) ........ 2010-05-19 22:10 +0000 [r264453] Mark Michelson * include/asterisk/_private.h, include/asterisk/options.h, main/asterisk.c, main/loader.c, main/channel.c, /, channels/chan_sip.c: Merged revisions 264452 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 | mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86 lines Fix transcode_via_sln option with SIP calls and improve PLC usage. From reviewboard: The problem here is a bit complex, so try to bear with me... It was noticed by a Digium customer that generic PLC (as configured in codecs.conf) did not appear to actually be having any sort of benefit when packet loss was introduced on an RTP stream. I reproduced this issue myself by streaming a file across an RTP stream and dropping approx. 5% of the RTP packets. I saw no real difference between when PLC was enabled or disabled when using wireshark to analyze the RTP streams. After analyzing what was going on, it became clear that one of the problems faced was that when running my tests, the translation paths were being set up in such a way that PLC could not possibly work as expected. To illustrate, if packets are lost on channel A's read stream, then we expect that PLC will be applied to channel B's write stream. The problem is that generic PLC can only be done when there is a translation path that moves from some codec to SLINEAR. When I would run my tests, I found that every single time, read and write translation paths would be set up on channel A instead of channel B. There appeared to be no real way to predict which channel the translation paths would be set up on. This is where Kevin swooped in to let me know about the transcode_via_sln option in asterisk.conf. It is supposed to work by placing a read translation path on both channels from the channel's rawreadformat to SLINEAR. It also will place a write translation path on both channels from SLINEAR to the channel's rawwriteformat. Using this option allows one to predictably set up translation paths on all channels. There are two problems with this, though. First and foremost, the transcode_via_sln option did not appear to be working properly when I was placing a SIP call between two endpoints which did not share any common formats. Second, even if this option were to work, for PLC to be applied, there had to be a write translation path that would go from some format to SLINEAR. It would not work properly if the starting format of translation was SLINEAR. The one-line change presented in this review request in chan_sip.c fixed the first issue for me. The problem was that in sip_request_call, the jointcapability of the outbound channel was being set to the format passed to sip_request_call. This is nativeformats of the inbound channel. Because of this, when ast_channel_make_compatible was called by app_dial, both channels already had compatibly read and write formats. Thus, no translation path was set up at the time. My change is to set the jointcapability of the sip_pvt created during sip_request_call to the intersection of the inbound channel's nativeformats and the configured peer capability that we determined during the earlier call to create_addr. Doing this got the translation paths set up as expected when using transcode_via_sln. The changes presented in channel.c fixed the second issue for me. First and foremost, when Asterisk is started, we'll read codecs.conf to see the value of the genericplc option. If this option is set, and ast_write is called for a frame with no data, then we will attempt to fill in the missing samples for the frame. The implementation uses a channel datastore for maintaining the PLC state and for creating a buffer to store PLC samples in. Even when we receive a frame with data, we'll call plc_rx so that the PLC state will have knowledge of the previous voice frame, which it can use as a basis for when it comes time to actually do a PLC fill-in. So, reviewers, now I ask for your help. First off, there's the one line change in chan_sip that I have put in. Is it right? By my logic it seems correct, but I'm sure someone can tell me why it is not going to work. This is probably the change I'm least concerned about, though. What concerns me much more is the set of changes in channel.c. First off, am I even doing it right? When I run tests, I can clearly see that when PLC is activated, I see a significant increase in RTP traffic where I would expect it to be. However, in my humble opinion, the audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to me than when no PLC is used at all. I need someone to review the logic I have used to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm sure someone can point out somewhere where I've done something incorrectly. As I was writing this review request up, I decided to give the code a test run under valgrind, and I find that for some reason, calls to plc_rx are causing some invalid reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around a bit to see why that is the case. If it's obvious to someone reviewing, speak up! Finally, I have one other proposal that is not reflected in my code review. Since without transcode_via_sln set, one cannot predict or control where a translation path will be up, it seems to me that the current practice of using PLC only when transcoding to SLINEAR is not useful. I recommend that once it has been determined that the method used in this code review is correct and works as expected, then the code in translate.c that invokes PLC should be removed. Review: https://reviewboard.asterisk.org/r/622/ ........ 2010-05-19 20:31 +0000 [r264405] David Vossel * main/udptl.c, /: Merged revisions 264400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 | dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines fixes infinite loop during udptl.c's decode_open_type When decode_length returns the length there is a check to see if that length is negative, if so the decode loop breaks as this means the limit has been reached. The problem here is that length is an unsigned int, so length can never be negative. This resulted in an infinite loop. (issue #17352) ........ 2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson * main/udptl.c, /: Merged revisions 264379 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 | mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4 lines Cast an unsigned int to a signed int when comparing it with 0. (AST-377) ........ * apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500 (Wed, 19 May 2010) | 12 lines Merged revisions 264334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May 2010) | 5 lines Set quieted flag when receiving a dtmf tone during playback in speechbackground. (closes issue #16966) Reported by: asackheim ........ ................ 2010-05-19 19:25 +0000 [r264332] David Vossel * /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 | dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines fixes crash in check_rtp_timeout During deadlock avoidance the sip dialog pvt is locked and unlocked. When this occurs we have no guarantee the pvt's owner is still valid. We were trying to access the pvt's owner after this without checking to see if it still existed first. (closes issue #17271) Reported by: under Patches: check_rtp_timeout.diff uploaded by under (license 914) Tested by: dvossel ........ 2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher * include/asterisk/options.h, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 264249 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010) | 24 lines Merged revisions 264248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) | 17 lines Internal timing is now on by default, if you're using DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is that this version ensures that a timer is always available, whereas in previous versions, it was possible for DAHDI to be loaded, but have no drivers to actually generate timing. If internal_timing was turned on in this circumstance, a complete lack of audio would result. This is the reason why internal_timing was not on by default. However, now that DAHDI ensures the availability of a timer, there is no reason for this setting to be off (and in fact, it solves a great many initial user problems). (closes issue #15932) Reported by: dimas Patches: 20100519__issue15932.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ * main/dsp.c, /: Merged revisions 264204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 | tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines Keep track of digit duration, when we're decoding inband to pass DTMF frames. (closes issue #17235) Reported by: frawd Patches: new_dtmf_dsp_len.patch uploaded by frawd (license 610) 20100518__issue17235.diff.txt uploaded by tilghman (license 14) Tested by: frawd ........ 2010-05-19 14:47 +0000 [r264115] David Vossel * main/rtp.c, /: Merged revisions 264114 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 | dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines fixes crash during dtmf During the processing of Cisco dtmf the dtmf samples were not being calculated correctly. In an attempt to determine what sample rate was being used, a NULL frame was processed which caused a crash. This patch resolves this. (closes issue #17248) Reported by: falves11 Patches: issue_17248.diff uploaded by dvossel (license 671) ........ 2010-05-19 08:15 +0000 [r264032] Alec L Davis * /, configs/indications.conf.sample: Merged revisions 264031 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19 May 2010) | 8 lines fix incorrectly typed indications for [nz] stutter and dialrecall (closes issue #17359) Reported by: alecdavis Patches: bug17359.diff.txt uploaded by alecdavis (license 585) ........ 2010-05-19 06:41 +0000 [r263951] Tilghman Lesher * main/dsp.c, /: Merged revisions 263950 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010) | 15 lines Merged revisions 263949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) | 8 lines Because progress is called multiple times, across several frames, we must persist states when detecting multitone sequences. (closes issue #16749) Reported by: dant Patches: dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by: dant ........ ................ 2010-05-18 22:49 +0000 [r263906] David Vossel * main/strings.c, /: Merged revisions 263904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 | dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines fixes segfault on logging (closes issue #17331) Reported by: under Patches: utils.diff uploaded by under (license 914) segfault_on_logging.diff uploaded by dvossel (license 671) Tested by: under, dvossel ........ 2010-05-18 19:41 +0000 [r263809] Jeff Peeler * apps/app_directory.c, /: Merged revisions 263807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500 (Tue, 18 May 2010) | 17 lines Merged revisions 263769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines Modify directory name reading to be interrupted with operator or pound escape. In the case of accidentally entering the wrong first three letters for the reading, users could be very frustrated if the name listing is very long. This allows interrupting the reading by pressing 0 or #. 0 will attempt to execute a configured operator (o) extension and # will exit and proceed in the dialplan. ABE-2200 ........ ................ 2010-05-17 22:10 +0000 [r263642] Mark Michelson * /, main/devicestate.c: Merged revisions 263640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May 2010) | 16 lines Merged revisions 263639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May 2010) | 10 lines Fix logic error when checking for a devstate provider. When using strsep, if one of the list of specified separators is not found, it is the first parameter to strsep which is now NULL, not the pointer returned by strsep. This issue isn't especially severe in that the worst it is likely to do is waste some cycles when a device with no '/' and no ':' is passed to ast_device_state. ........ ................ 2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010) | 9 lines With IMAP backend, messages in INBOX were counted twice for MWI. (closes issue #17135) Reported by: edhorton Patches: 20100513__issue17135.diff.txt uploaded by tilghman (license 14) 17135_2.diff uploaded by ebroad (license 878) Tested by: edhorton, ebroad ........ * main/app.c: Don't close 'n', just close 'above_n'. (closes issue #17345) Reported by: wdoekes 2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen * main/manager.c, /: Merged revisions 263457 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010) | 19 lines Recorded merge of revisions 263456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) | 11 lines Manager cookies are not compatible with RFC2109. The Version field in the cookies we're setting contain quotes around the version number which is not compatible with RFC2109 and breaks some implementations. (closes issue #17231) Reported by: ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by ecarruda (license 559) Tested by: ecarruda, russell ........ ................ * sounds/Makefile, /: Merged revisions 263375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010) | 16 lines Merged revisions 263374 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010) | 8 lines Update link to new version of core sounds. The latest version of the core sounds files 1.4.19 now includes the missing queue-minute sound file which is called by app_queue but which has been missing. (closes issue #17123) Reported by: n8ideas ........ ................ 2010-05-17 13:03 +0000 [r263293] David Vossel * CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer policy dialstring option 2010-05-15 23:41 +0000 [r263202] Paul Belanger * /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 | tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines Make the Makefile logic more explicit and move the Snow Leopard logic down to where it's not executed on non-Darwin systems. (closes issue #17028) Reported by: pabelanger Patches: issue17028_20100315.patch uploaded by seanbright (license 71) 20100315__issue17028.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, pabelanger ........ 2010-05-13 22:13 +0000 [r263070] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010) | 1 line Fix inverted logic in cli command: ss7 set debug on/off ........ 2010-05-13 15:36 +0000 [r262898] Russell Bryant * channels/chan_console.c, /: Merged revisions 262897 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010) | 4 lines Fix an off by one error that causes a crash. Thanks to Raymond Burke for pointing it out. ........ 2010-05-12 20:01 +0000 [r262801] Paul Belanger * main/loader.c, main/cli.c, /: Merged revisions 262800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed, 12 May 2010) | 8 lines Notify CLI when modules is loaded / unloaded (closes issue #17308) Reported by: pabelanger Patches: cli.modules.patch uploaded by pabelanger (license 224) Tested by: pabelanger, russell ........ 2010-05-12 19:53 +0000 [r262797-262799] Leif Madsen * res/ael/pval.c, /: Merged revisions 262798 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 | lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines Revert previous WARNING message removal. Marquis42 suggested a better method of doing what I wanted because I ended up removing the WARNING message for all instances when really I just wanted to remove it for the 'return' keyword, not everything. (issue #17145) ........ * res/ael/pval.c, /: Merged revisions 262796 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 | lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines Remove unnecessary WARNING message in ael/pval.c (closes issue #17145) Reported by: okrief ........ 2010-05-12 18:03 +0000 [r262746] David Vossel * /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010) | 17 lines Merged revisions 262662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines fixes app_meetme dsp error We attempted to detect silence after translating a frame from signed linear. This caused a flooding of errors. To resolve this the code to detect silence was moved before the translation. (closes issue #17133) Reported by: jsdyer ........ ................ 2010-05-12 16:29 +0000 [r262516-262659] Tilghman Lesher * /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 | tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines Ensure the arguments are initialized. Also miscellaneous CG cleanup. (closes issue #16576) Reported by: uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman (license 14) Tested by: uxbod ........ * /, include/asterisk/causes.h: Merged revisions 262513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11 May 2010) | 7 lines Move cause 200 to cause 26, as specified in Q.850. Also cleanup the formatting and add a few more that seem like good candidates. (closes issue #16157) Reported by: wimpy ........ 2010-05-11 19:58 +0000 [r262425] Jason Parker * /, res/Makefile: Merged revisions 262422 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) | 18 lines Merged revisions 262421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines Use a less silly method for modifying a flex-generated file. The sed syntax that was used wasn't actually valid, causing some versions to choke. This is the method that is used in 1.6.x+ for similar changes. (closes issue #16696) Reported by: bklang Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested by: qwell ........ ................ 2010-05-11 19:41 +0000 [r262415-262420] Paul Belanger * pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 | pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8 lines Improve logging by displaying line number (closes issue #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger ........ * /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 | pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8 lines Improve logging information for misconfigured contexts (closes issue #17238) Reported by: pprindeville Patches: chan_sip-bug17238.patch uploaded by pprindeville (license 347) Tested by: pprindeville ........ 2010-05-11 17:25 +0000 [r262340] Tilghman Lesher * apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262330 | tilghman | 2010-05-11 12:23:51 -0500 (Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010) | 2 lines Fix issue #17302 a slightly different way (mad props to Qwell) ........ ................ 2010-05-10 19:06 +0000 [r262237-262241] David Vossel * /, apps/app_directed_pickup.c: Merged revisions 262240 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10 May 2010) | 9 lines fixes PickupChan application (closes issue #16863) Reported by: schern Patches: app_directed_pickup.c.patch uploaded by schern (license 995) for_trunk.diff uploaded by cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama, rickead2000, dvossel ........ * channels/chan_console.c, /: Merged revisions 262236 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010) | 11 lines fixes crash in chan_console There is a race condition between console_hangup() and start_stream(). It is possible for console_hangup() to be called and then the stream thread to begin after the hangup. To avoid this a check in start_stream() to make sure the pvt-owner still exists while the pvt lock is held is made. If the owner is gone that means the channel hung up and start_stream should be aborted. ........ 2010-05-10 16:39 +0000 [r262155] Tilghman Lesher * /, Makefile.rules: Merged revisions 262152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010) | 17 lines Merged revisions 262151 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010) | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes issue #17297) Reported by: jcovert Patches: 20100506__issue17297.diff.txt uploaded by tilghman (license 14) (closes issue #17302) Reported by: jcovert ........ ................ 2010-05-09 02:17 +0000 [r261916-262105] Tilghman Lesher * autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4, autoconf/ast_c_define_check.m4, /, configure, include/asterisk/autoconfig.h.in: Merged revisions 262102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08 May 2010) | 5 lines Cleanup a bit more by getting rid of useless version defines. Also make library detection use passed CFLAGS. (closes issue #17309) Reported by: stuarth ........ * /, configure, configure.ac: Merged revisions 262048 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010) | 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only ........ * /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 | tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines Double free crash (closes issue #17245) Reported by: thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by tilghman (license 14) Tested by: murraytm ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 261913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 | tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14 lines Use the detected pthread building flags in every place, instead of hardcoding -lpthread. We nicely detect the right flags on each system for building Asterisk with pthreads, then ignore it for every other build option that requires us to build with pthreads. This caused some items to return a false negative. Also cleanup some minor naming issues that caused "library library" redundancy in the output. (closes issue #17303) Reported by: stuarth Patches: 20100507__issue17303.diff.txt uploaded by tilghman (license 14) Tested by: stuarth ........ 2010-05-07 16:08 +0000 [r261868] Leif Madsen * UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 | lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines Update UPGRADE-1.6.txt stating insecure=very has been removed. (closes issue #17282) Reported by: stuarth Tested by: stuarth ........ 2010-05-06 20:13 +0000 [r261739] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500 (Thu, 06 May 2010) | 15 lines Merged revisions 261735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines Only allow the operator key to be accepted after leaving a voicemail. Or rather disallow the operator key from being accepted when not offered, such as after finishing a recording from within the mailbox options menu. ABE-2121 SWP-1267 ........ ................ 2010-05-06 17:08 +0000 [r261612] Jason Parker * sounds/Makefile, /: Merged revisions 261609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) | 11 lines Merged revisions 261608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) | 4 lines Use the versioned MOH tarballs, now that we have them. This makes for more reproducibility. Prompted by a discussion in #asterisk-dev ........ ................ 2010-05-06 15:43 +0000 [r261563] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 | tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines Permit more lines within a SIP body to be parsed. The example given within the related issue showed 120 lines, which was mostly a result of the body being XML. (closes issue #17179) Reported by: khw ........ 2010-06-01 Leif Madsen * Asterisk 1.6.2.8 Released. 2010-05-26 Leif Madsen * Asterisk 1.6.2.8-rc2 Released. 2010-05-26 10:56 -0500 [r265891] Matt Nicholson * Merged r265610 from 1.4: Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal 2010-05-06 Leif Madsen * Asterisk 1.6.2.8-rc1 Released 2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant * tests/test_heap.c: Add test case that ensures the heap handles arbitrary removals properly. (issue #17277) Reported by: cappucinoking Patches: test_heap.diff uploaded by cappucinoking (license 1036) Tested by: cappucinoking, russell * /, main/heap.c: Merged revisions 261496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 | russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines Fix handling of removing nodes from the middle of a heap. This bug surfaced in 1.6.2 and does not affect code in any other released version of Asterisk. It manifested itself as SIP qualify not happening when it should, causing peers to go unreachable. This was debugged down to scheduler entries sometimes not getting executed when they were supposed to, which was in turn caused by an error in the heap code. The problem only sometimes occurs, and it is due to the logic for removing an entry in the heap from an arbitrary location (not just popping off the top). The scheduler performs this operation frequently when entries are removed before they run (when ast_sched_del() is used). In a normal pop off of the top of the heap, a node is taken off the bottom, placed at the top, and then bubbled down until the max heap property is restored (see max_heapify()). This same logic was used for removing an arbitrary node from the middle of the heap. Unfortunately, that logic is full of fail. This patch fixes that by fully restoring the max heap property when a node is thrown into the middle of the heap. Instead of just pushing it down as appropriate, it first pushes it up as high as it will go, and _then_ pushes it down. Lastly, fix a minor problem in ast_heap_verify(), which is only used for debugging. If a parent and child node have the same value, that is not an error. The only error is if a parent's value is less than its children. A huge thanks goes out to cappucinoking for debugging this down to the scheduler, and then producing an ast_heap test case that demonstrated the breakage. That made it very easy for me to focus on the heap logic and produce a fix. Open source projects are awesome. (closes issue #16936) Reported by: ib2 Tested by: cappucinoking, crjw (closes issue #17277) Reported by: cappucinoking Patches: heap-fix.rev2.diff uploaded by russell (license 2) Tested by: cappucinoking, russell ........ 2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen * channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) | 4 lines When failing to configure, don't destroy 'cfg' twice Fixes a crash when some config section had an incorrect channel config. ........ 2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May 2010) | 19 lines Merged revisions 261274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May 2010) | 12 lines Registration fix for SIP realtime. Make sure realtime fields are not empty. (closes issue #17266) Reported by: Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis, sberney Review: https://reviewboard.asterisk.org/r/643/ ........ ................ * apps/app_queue.c, /: Merged revisions 261232 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 | pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10 lines 'queue reset stats' erroneously clears wrapuptime configuration. Resets each member's lastcall to 0 now. (closes issue #17262, #16519) Reported by: rain Patches: wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested by: rain ........ 2010-05-04 23:55 +0000 [r261098] Tilghman Lesher * main/channel.c, /: Merged revisions 261095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010) | 18 lines Merged revisions 261093-261094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010) | 7 lines Protect against overflow, when calculating how long to wait for a frame. (closes issue #17128) Reported by: under Patches: d.diff uploaded by under (license 914) ........ r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2 lines Add a tiny corner case to the previous commit ........ ................ 2010-05-04 19:01 +0000 [r260927] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500 (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) | 12 lines Voicemail transfer to operator should occur immediately, not after main menu. There were two scenarios in the advanced options that while using the operator=yes and review=yes options, the transfer occurred only after exiting the main menu (after sending a reply or leaving a message for an extension). Now after the audio is processed for the reply or message the transfer occurs immediately as expected. ABE-2107 ABE-2108 ........ ................ 2010-05-04 16:58 +0000 [r260884] Matthew Nicholson * configs/sip.conf.sample, include/asterisk/frame.h, main/channel.c, /, channels/chan_sip.c: Merged revisions 254450 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25 Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that arrive before a T.38-capable application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ ........ 2010-05-04 15:51 +0000 [r260746-260805] Jason Parker * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260802 | qwell | 2010-05-04 10:49:57 -0500 (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May 2010) | 1 line Fix fallout from removing from configure script. Pointed out by philipp64 on #asterisk-dev ........ ................ * /: Fix merge props 2010-05-03 17:42 +0000 [r260743] Paul Belanger * Makefile, /: Merged revisions 260661-260662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend libdir when executing mkpkgconfig allowing non-root installs to work. (closes issue #17268) Reported by: pabelanger Patches: issue17268.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41 -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/ part. Thanks Qwell. ........ 2010-05-03 14:59 +0000 [r260571] Leif Madsen * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500 (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010) | 1 line Minor typo pointed out by pabelanger on IRC. ........ ................ 2010-04-30 22:48 +0000 [r260441] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500 (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines Ensure channel state is not incorrectly set in the case of a very early answer. The needringing bit was being read in dahdi_read after answering thereby setting the state to ringing from up. This clears needringing upon answering so that is no longer possible. (closes issue #17067) Reported by: tzafrir Patches: needringing.diff uploaded by tzafrir (license 46) ........ ................ 2010-04-30 20:22 +0000 [r260373] Mark Michelson * res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500 (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan->stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 ........ ................ 2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher * /, main/app.c: Merged revisions 260292 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 | tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13 lines Don't allow file descriptors to go above 64k, when we're closing them in a fork(2). This saves time, when, even though the system allows the process limit to be that high, the practical limit is much lower. (closes issue #17223) Reported by: dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by tilghman (license 14) Tested by: dbackeberg ........ * configs/extensions.conf.sample, /: Merged revisions 260280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30 Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan context. (closes issue #17263) Reported by: pprindeville Patches: freenum-dialplan.patch#3 uploaded by pprindeville (license 347) ........ 2010-04-29 23:13 +0000 [r260234] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500 (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines DTMF CallerID detection problems. The code handling DTMF CallerID drops digits on long CallerID numbers and may timeout waiting for the first ring with shorter numbers. The DTMF emulation mode was not turned off when processing DTMF CallerID. When the emulation code gets behind in processing the DTMF digits it can skip a digit. For shorter numbers, the timeout may have been too short. I increased it from 2 seconds to 4 seconds. Four seconds is a typical time between rings for many countries. (closes issue #16460) Reported by: sum Patches: issue16460.patch uploaded by rmudgett (license 664) issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA AST-334 JIRA SWP-901 ........ ................ 2010-04-29 18:18 +0000 [r260156] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 260148 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29 Apr 2010) | 2 lines Pattern match fail. ........ 2010-04-29 15:35 +0000 [r260051] David Vossel * main/audiohook.c, /, include/asterisk/audiohook.h: Merged revisions 260050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010) | 21 lines Merged revisions 260049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) | 14 lines Fixes crash in audiohook_write_list The middle_frame in the audiohook_write_list function was being freed if a audiohook manipulator returned a failure. This is incorrect logic. This patch resolves this and adds detailed descriptions of how this function should work and why manipulator failures must be ignored. (closes issue #17052) Reported by: dvossel Tested by: dvossel (closes issue #16196) Reported by: atis Review: https://reviewboard.asterisk.org/r/623/ ........ ................ 2010-04-28 22:36 +0000 [r259959] Mark Michelson * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 | mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11 lines Don't override peer context with domain context. (closes issue #17040) Reported by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347) Tested by: pprindeville Review: https://reviewboard.asterisk.org/r/565/ ........ 2010-04-28 21:26 +0000 [r259899] David Vossel * main/channel.c, channels/chan_local.c, /: Merged revisions 259870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500 (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) | 33 lines resolves deadlocks in chan_local Issue_1. In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock avoidance is done when the channel to hangup is the outbound chan_local channel, but when it is not the outbound channel we have an issue... We attempt to do deadlock avoidance only on the tech pvt, when both the tech pvt and the pvt->owner are locked coming into that loop. By never giving up the pvt->owner channel deadlock avoidance is not entirely possible. This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is used in ast_activate_generator() to queue a frame on the channel and make the channel's read function get called. This function is used in ast_activate_generator() while the channel is locked, which mean's the channel will have a lock both from the generator code and the frame_queue code by the time it gets to chan_local.c's local_queue_frame code... local_queue_frame contains some of the same crazy deadlock avoidance that local_hangup requires, and this recursive lock prevents that deadlock avoidance from happening correctly. This patch removes ast_prod() from the channel lock so only one lock is held during the local_queue_frame function. (closes issue #17185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ ........ ................ 2010-04-28 21:09 +0000 [r259854] Leif Madsen * config.guess: Merged revisions 259853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010) | 14 lines Merged revisions 259852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010) | 6 lines Update config.guess. Updating config.guess because after installing Ubuntu Server 9.10 and running all the update scripts, running ./configure would not continue because it was unable to determine what kind of system I had. After updating config.guess things started working again. ........ ................ 2010-04-28 20:34 +0000 [r259781-259851] Jason Parker * /, configure, configure.ac: Merged revisions 259848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259848 | qwell | 2010-04-28 15:32:14 -0500 (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so systems without install can use install-sh from our source dir. ........ ................ * makeopts.in, /: Merged revisions 259837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) | 9 lines Merged revisions 259833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) | 1 line Missed this when removing $ID ........ ................ * Makefile, /, configure, configure.ac: Merged revisions 259760 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259760 | qwell | 2010-04-28 14:19:54 -0500 (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) | 7 lines Remove usage of `id` since it isn't useful and was causing breakge. Solaris `id` doesn't support the -u argument. Instead of figuring out how to fix this to work on Solaris, I decided to check why it was necessary and where else it was used. It was only used in one place, and it hasn't been needed for a very long time (I question whether it was ever needed). ........ ................ 2010-04-28 17:19 +0000 [r259681] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500 (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010) | 4 lines Do not play goodbye prompt after timeout of message review. ABE-2124 ........ ................ 2010-04-27 22:46 +0000 [r259616] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500 (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success" Changed the warning to "Failed to decode CallerID on channel 'name'". The message before it is likely more specific about why the CallerID decode failed. SWP-501 AST-283 ........ ................ 2010-04-27 21:50 +0000 [r259528] Leif Madsen * sounds/Makefile: Merged revisions 259527 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010) | 23 lines Merged revisions 259526 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010) | 15 lines Update sounds files. * Add additional sounds prompts for say_enumeration * Update the English conference sounds prompts so they are better quality and all sound more consistent * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files to include all present sound files Both core (en, fr, es) and extra (en, fr) sounds files have been updated. (closes issue #16200) Reported by: murf (closes issue #17137) Reported by: lmadsen ........ ................ 2010-04-27 21:25 +0000 [r259356-259486] Jason Parker * main/editline/configure.in, /, main/editline/configure, main/editline/Makefile.in: Merged revisions 259439 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) | 5 lines Add gar to the check for AR for those silly OSes (Solaris) that don't have ar. autoconf2.13 couldn't handle AC_PROG_GREP, so I removed it. This is fine, since we don't need to use anything that the configure script doesn't. ........ * /: Unblock revision 259439. * /, configure, configure.ac: Merged revisions 259353 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259353 | qwell | 2010-04-27 14:31:55 -0500 (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) | 5 lines Support the silly OSes that don't have ar and strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to AC_CHECK_TOOLS. ........ ................ 2010-04-27 19:03 +0000 [r259310] Richard Mudgett * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 259307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010) | 21 lines Merged revisions 259270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ ................ 2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson * main/channel.c, /: Merged revisions 259105 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr 2010) | 9 lines Merged revisions 259104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr 2010) | 3 lines Let compilation succeed warning-free when DONT_OPTIMIZE is turned off. ........ ................ * main/channel.c, /: Merged revisions 259023 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr 2010) | 19 lines Merged revisions 259018 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr 2010) | 13 lines Prevent Newchannel manager events for dummy channels. No Newchannel manager event will be fired for channels that are allocated to not match a registered technology type. Thus bogus channels allocated solely for variable substitution or CDR operations do not result in a Newchannel event. (closes issue #16957) Reported by: atis Review: https://reviewboard.asterisk.org/r/601 ........ ................ 2010-04-26 16:00 +0000 [r258935] Leif Madsen * /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 | lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines Small error in the T.140 RTP port verbose log. (closes issue #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610) Tested by: russell ........ 2010-04-25 18:14 +0000 [r258779] Tilghman Lesher * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010) | 13 lines Merged revisions 258775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) | 6 lines When StopMonitor is called, ensure that it will not be restarted by a channel event. (closes issue #16590) Reported by: kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888) ........ ................ 2010-04-22 22:15 +0000 [r258676] Matthew Nicholson * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions 258671,258675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May 2009) | 8 lines Set the proper disposition on originated calls. (closes issue #14167) Reported by: jpt Patches: call-file-missing-cdr2.diff uploaded by mnicholson (license 96) Tested by: dlotina, rmartinez, mnicholson ........ r258670 | mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11 lines Fix broken CDR behavior. This change allows a CDR record previously marked with disposition ANSWERED to be set as BUSY or NO ANSWER. Additionally this change partially reverts r235635 and does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call(). To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in ast_bridge_call(). (closes issue #16797) Reported by: VarnishedOtter Tested by: mnicholson ........ (closes issue #16222) Reported by: telles Tested by: mnicholson ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500 (Thu, 22 Apr 2010) | 2 lines Fix previous commit. ................ 2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant * /, main/event.c: Merged revisions 258632 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only merge the bug fixes, not the unit test. ........ r258632 | russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines Add ast_event subscription unit test and fix some ast_event API bugs. This patch introduces another test in test_event.c that exercises most of the subscription related ast_event API calls. I made some minor additions to the existing event allocation test to increase API coverage by the test code. Finally, I made a list in a comment of API calls not yet touched by the test module as a to-do list for future test development. During the development of this test code, I discovered a number of bugs in the event API. 1) subscriptions to AST_EVENT_ALL were not handled appropriately in a couple of different places. The API allows a subscription to all event types, but with IE parameters, just as if it was a subscription to a specific event type. However, the parameters were being ignored. This affected ast_event_check_subscriber() and event distribution to subscribers. 2) Some of the logic in ast_event_check_subscriber() for checking subscriptions against query parameters was wrong. Review: https://reviewboard.asterisk.org/r/617/ ........ * /, doc/tex/channelvariables.tex: Merged revisions 258515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22 Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........ 2010-04-21 22:11 +0000 [r258436] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500 (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) | 8 lines Fix looping forever when no input received in certain voicemail menu scenarios. Specifically, prompting for an extension (when leaving or forwarding a message) or when prompting for a digit (when saving a message or changing folders). ABE-2122 SWP-1268 ........ ................ 2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen * doc/tex/asterisk.tex: Remove missed line in previous merge. (issue #17220) * configure: Forgot to merge the updated configure script. (issue #17220) * doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in, doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile, configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex, build_tools/prep_tarball: Merged revisions 258351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010) | 20 lines Add ability to generate ASCII documentation from the TeX files. These changes add the ability to run 'make asterisk.txt' just like the existing 'make asterisk.pdf' commands to generate a text document from the TeX files we have in the doc/tex/ directory. I've also updated a few of the .tex files because they weren't properly escaping certain characters so they would show up as Unicode characters (like [U+021C]). Made changes to the configure scripts so it would detect the catdvi program which is required to convert the .dvi file generated by latex. I've also added a few lines to the build_tools/prep_tarball script so that the text documentation gets generated and added to future tarballs of Asterisk releases. (closes issue #17220) Reported by: lmadsen Patches: asterisk.txt.patch uploaded by lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger (license 224) Tested by: lmadsen, pabelanger ........ 2010-04-21 18:19 +0000 [r258314] David Vossel * /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 | dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines fixes issue with double "sip:" in header field This is a clear mistake in logic. Future discussions about how to avoid having to handle uri's like this should take place in the future, but this fix needs to go in for now. (closes issue #15847) Reported by: ebroad Patches: doublesip.patch uploaded by ebroad (license 878) ........ 2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen * /, configs/cli_aliases.conf.sample: Merged revisions 258149 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20 Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on asterisk-users. ........ * configs/extensions.conf.sample, /: Merged revisions 258147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20 Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers (http://www.freenum.org). Minor tweaks and documentation added by me. (closes issue #17058) Reported by: pprindeville Patches: freenum.patch#5 uploaded by pprindeville (license 347) Tested by: lmadsen ........ 2010-04-20 18:04 +0000 [r258108] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500 (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) | 11 lines Play correct prompt when voicemail store failure occurs after attempted forward. If a user's mailbox was full and a message was attempted to be forwarded to said box, warnings on the console would indicate failure. However, the played prompt was that of success (vm-msgsaved). Now storage failure is taken into account and the correct prompt (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262 ........ ................ 2010-04-20 18:02 +0000 [r258107] Leif Madsen * contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20 Apr 2010) | 7 lines Add missing 'useragent' field to sip-friends.sql file. (closes issue #17171) Reported by: thehar Patches: sip-friends.patch uploaded by thehar (license 831) Tested by: pabelanger, thehar ........ 2010-04-19 21:58 +0000 [r257948-257950] Jason Parker * main/indications.c, /: Merged revisions 257949 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 | qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line Change log message to match severity. ........ * main/indications.c, /: Merged revisions 257947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 | qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines Don't consider a missing indications.conf to be a critical error. There were many changes in revision 176627 which would avoid the error that a missing config would have caused. Other than this, there are no other config files (including asterisk.conf, surprisingly) that are required. ........ 2010-04-19 18:30 +0000 [r257850] Terry Wilson * /, main/features.c: Merged revisions 257810 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 | twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines Fix incomplete CDR merge from r195881 Because res/res_features.c was removed and main/cdr.c added, these changes didn't make it to trunk and the 1.6.x branches ........ 2010-04-18 17:28 +0000 [r257771] Tilghman Lesher * configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18 Apr 2010) | 2 lines Removing unused configuration parameters ........ 2010-04-16 21:47 +0000 [r257740] Dwayne M. Hubbard * apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500 (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines Make the mixmonitor thread process audio frames faster Mantis issue 17078 reports MixMonitor recordings have shorter durations than the call duration. This was because the mixmonitor thread was not processing frames from the audiohook fast enough. The mixmonitor thread would slowly fall behind the most recent audio frame and when the channel hangs up, the mixmonitor thread would exit without processing the same number of frames as the channel; leaving the mixmonitor recording shorter than actual call duration. This revision fixes this issue by moving the ast_audiohook_trigger_wait() and the subsequent audiohook.status check into the block where the ast_audiohook_read_frame() function returns NULL. (closes issue #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review: https://reviewboard.asterisk.org/r/611/ ........ ................ 2010-04-15 21:34 +0000 [r257510-257597] Tilghman Lesher * include/asterisk/app.h, /, main/app.c: Merged revisions 257560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500 (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) | 6 lines Allow application options with arguments to contain parentheses, through a variety of escaping techniques. Fixes SWP-1194 (ABE-2143). Review: https://reviewboard.asterisk.org/r/604/ ........ ................ * /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010) | 20 lines Merged revisions 257467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines Don't recreate peer, when responding to a repeated deregistration attempt. When a reply to a deregistration is lost in transmit, the client retries the deregistration. Previously, this would cause a realtime/autocreate peer to be loaded back into memory, after it had already been correctly purged. Instead, we just want to resend the reply without loading the peer. (closes issue #16908) Reported by: kkm Patches: 20100412__issue16908.diff.txt uploaded by tilghman (license 14) Tested by: kkm ........ ................ 2010-04-15 19:42 +0000 [r257344-257428] Leif Madsen * doc/backtrace.txt: Merged revisions 257427 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010) | 21 lines Merged revisions 257426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) | 13 lines Update backtrace.txt documentation. Update the backtrace.txt documentation so it conforms to the same layout as other documents we've been working on recently. Additionally, add a bunch of new information about gathering backtraces for crashes and deadlocks, along with ways of verifying your file before uploading it. Create a couple of one line commands for people to generate the files we need. (closes issue #17190) Reported by: lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen (license 10) Tested by: lmadsen, pabelanger ........ ................ * doc/backtrace.txt: Merged revisions 257343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010) | 9 lines Merged revisions 257342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) | 1 line Update address of the bug tracker. ........ ................ 2010-04-14 23:00 +0000 [r257265] Tilghman Lesher * configs/features.conf.sample, /, main/features.c: Merged revisions 257262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 | tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15 lines Yet another issue where the conversion of the application delimiter to comma caused an issue. Application arguments within the feature map could possibly contain a comma, which conflicts with the syntax of the features.conf configuration file. This patch allows the argument to be wrapped in parentheses or quoted, to allow the application arguments to be interpreted as a single configuration parameter. (closes issue #16646) Reported by: pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/547/ ........ 2010-04-13 19:20 +0000 [r257210] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 | tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10 lines Also unref the pvt when we delete the provisional keepalive job. (closes issue #16774) Reported by: kowalma Patches: 20100315__issue16774.diff.txt uploaded by tilghman (license 14) Tested by: falves11, jamicque Review: https://reviewboard.asterisk.org/r/591/ ........ 2010-04-13 18:43 +0000 [r257184] Matthew Nicholson * main/manager.c, /, configs/manager.conf.sample: Merged revisions 257146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr 2010) | 16 lines Merged revisions 257070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines Add an option to restore past broken behavor of the Events manager action Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested. (closes issue #17023) Reported by: nblasgen Review: https://reviewboard.asterisk.org/r/602/ ........ ................ 2010-04-13 16:38 +0000 [r257068] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010) | 8 lines Ensure that we can have commas within cdr values. (closes issue #17001) Reported by: snuffy Patches: 20100412__issue17001.diff.txt uploaded by tilghman (license 14) Tested by: snuffy ........ 2010-04-12 17:30 +0000 [r256822-256902] Leif Madsen * doc/HOWTO_collect_debug_information.txt (added): Merged revisions 256901 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010) | 23 lines Merged revisions 256900 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) | 15 lines Add How-To document on collecting debugging info for issues.asterisk.org Paul Belanger has been helping a lot with bug tracking recently and created this document that we can now point to when additional debugging information is required. This document will help those filing issues to know how to get the information required when filing their issues. This will make things easier on the developers. Initial text and changes by pabelanger. Tweaks and editing by myself. (closes issue #17159) Reported by: pabelanger Patches: HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10) Tested by: tzafrir, pabelanger, lmadsen ........ ................ * apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010) | 3 lines Remove silly debug message that is not useful. (issue #17159) ........ * /, main/logger.c: Merged revisions 256821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 | lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines CLI command logger set level auto complete. A simple patch to enable auto tab complete. (closes issue #17152) Reported by: pabelanger Patches: 0017152.patch uploaded by pabelanger (license 224) ........ 2010-04-08 22:03 +0000 [r256483] Tilghman Lesher * main/app.c: Backport /proc/%d/fd method of closing file descriptors to 1.6.2. 2010-04-06 19:40 +0000 [r256373] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/lock.h: Merged revisions 256370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010) | 2 lines Mac OS X does not support comparing a mutex to its initializer. Create a test for this. ........ 2010-04-06 18:53 +0000 [r256268-256368] Richard Mudgett * channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty after the first call. The bug is exposed if MFC/R2 support is built into asterisk (i.e., openr2.h is present in the include path). Code that unconditionally clears the CallerID name and number is included. Also fixed a malformed if test in mkintf() added by issue 15883. Converted the if statement to a switch statement for clarity. Regression of the issue 15883 fix. (closes issue #16968) Reported by: grecco Patches: issue16968.patch uploaded by rmudgett (license 664) (closes issue #16747) Reported by: viniciusfontes * channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500 (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock. SWP-1231 ABE-2163 ........ ................ 2010-05-03 Leif Madsen * Asterisk 1.6.2.7 Released 2010-04-29 Leif Madsen * Asterisk 1.6.2.7-rc3 Released 2010-04-29 10:31 +0000 [r260053] David Vossel * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in audiohook_write_list. (closes issue 0017052) Reported by: dvossel Tested by: dvossel. (closes issue 0016196) Reported by: atis. Review: https://reviewboard.asterisk.org/r/623/ 2010-04-28 10:31 +0000 [r259899] David Vossel * channels/chan_local.c, main/channel.c: Resolves deadlocks in chan_local. (closes issue 0017185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ 2010-04-13 Leif Madsen * Asterisk 1.6.2.7-rc2 Released 2010-04-13 [r257210] Tilghman Lesher Also unref the pvt when we delete the provisional keepalive job. (closes issue #16774) Reported by: kowalma Patches: 20100315__issue16774.diff.txt uploaded by tilghman (license 14) Tested by: falves11, jamicque Review: https://reviewboard.asterisk.org/r/591/ 2010-04-05 Leif Madsen * Asterisk 1.6.2.7-rc1 Released 2010-04-05 15:15 +0000 [r256162] Leif Madsen * doc/tex/localchannel.tex, /: Merged revisions 256161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010) | 1 line Fix for localchannel.tex to allow PDFs to be generated again. ........ 2010-04-02 23:56 +0000 [r256013-256020] Russell Bryant * /, apps/app_meetme.c: Merged revisions 256019 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256019 | russell | 2010-04-02 18:55:57 -0500 (Fri, 02 Apr 2010) | 10 lines Export MEETMEBOOKID and fix pin-less conferences with realtime conferences (closes issue #16866) Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA Review: https://reviewboard.asterisk.org/r/582/ ........ * channels/chan_local.c, /: Merged revisions 256015 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256015 | russell | 2010-04-02 18:46:45 -0500 (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) | 9 lines Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel() (closes issue #16840) Reported by: bzing2 Patches: patch.txt uploaded by bzing2 (license 902) issue_16840.rev1.diff uploaded by russell (license 2) Tested by: bzing2, russell ........ ................ * main/channel.c, /: Merged revisions 256010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010) | 9 lines Merged revisions 256009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010) | 2 lines Remove extremely verbose debug message. ........ ................ 2010-04-02 20:20 +0000 [r255955] Tilghman Lesher * main/asterisk.c, /: Merged revisions 255952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 | tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines Pass the PID of the Asterisk process, not the PID of the canary. (closes issue #17065) Reported by: globalnetinc Patches: astcanary.patch uploaded by makoto (license 38) Tested by: frawd, globalnetinc ........ 2010-04-01 18:21 +0000 [r255676-255816] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010) | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt uploaded by tilghman (license 14) ........ * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500 (Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) | 15 lines Ensure line terminators in email are consistent. Fixes an issue with certain Mail Transport Agents, where attachments are not interpreted correctly. (closes issue #16557) Reported by: jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14) Tested by: ebroad, zktech Reviewboard: https://reviewboard.asterisk.org/r/544/ ........ ................ 2010-03-31 17:49 +0000 [r255505] Leif Madsen * configs/sip.conf.sample, apps/app_dial.c: Merged revisions 255504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31 Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T' can be used. (closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad ........ 2010-03-30 20:58 +0000 [r255326-255413] Russell Bryant * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r255410 | russell | 2010-03-30 15:56:26 -0500 (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does not start. ........ ................ * /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010) | 9 lines Merged revisions 255322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010) | 2 lines Don't make Asterisk not start if pbx_dundi fails to initialize. ........ ................ 2010-03-26 19:28 +0000 [r255023-255067] Leif Madsen * configs/sip.conf.sample, /: Merged revisions 255066 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010) | 6 lines Replace some documentation from 1.6.x back into trunk. This documentation associated wth tlsbindaddr is still useful so lets synchronize it between trunk and 1.6.x branches. (issue #17054) ........ * configs/sip.conf.sample, /: Merged revisions 255021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010) | 8 lines Update confusing documentation for tlsbindaddr. Update some confusing documentation for the tlsbindaddr option in sip.conf.sample. Point at a link instead which has better documentation. (closes issue #17054) Reported by: klaus3000 ........ 2010-03-25 20:43 +0000 [r254770-254805] Jason Parker * utils/Makefile, /: Merged revisions 254802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) | 9 lines Merged revisions 254800 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) | 1 line Don't remove local copies of utils in uninstall. ........ ................ * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS issue with out-of-tree modules. Take 2, without ABI breakage this time. Review: https://reviewboard.asterisk.org/r/588/ 2010-03-25 20:09 +0000 [r254721] Russell Bryant * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010) | 2 lines chan_usbradio depends on alsa. ........ 2010-03-25 17:47 +0000 [r254556] Mark Michelson * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500 (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar 2010) | 5 lines Add doxygen for acl.h Review: https://reviewboard.asterisk.org/r/528 ........ ................ 2010-03-25 17:21 +0000 [r254548] Sean Bright * channels/chan_sip.c: Initialize stream to avoid a compilation error. 2010-03-25 17:12 +0000 [r254542] Mark Michelson * channels/chan_sip.c: Fix potential crashes from trying to reference nonexistent RTP streams. 2010-03-25 16:26 +0000 [r254499] Terry Wilson * /, main/file.c: Merged revisions 254453 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010) | 9 lines Merged revisions 254451 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) | 2 lines Handle new SRCCHANGE control message here too ........ ................ 2010-03-25 16:22 +0000 [r254482] Mark Michelson * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500 (Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection. Here is a copy and paste of the details from my request on reviewboard that dealt with these changes: Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too: seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF. Fix 2. The second change in place is to fix an issue like the following: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end) *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list. Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem ........ ................ 2010-03-25 15:21 +0000 [r254447] Leif Madsen * /, res/res_agi.c: Merged revisions 254446 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 | lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines handle_speechset has 4 arguments. Update code to reflect that handle_speechset has 4 arguments. (closes issue #17093) Reported by: gpatri Patches: res_agi.patch uploaded by gpatri (license 1014) Tested by: pabelanger, mmichelson ........ 2010-03-24 17:19 +0000 [r254288] Jeff Peeler * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010) | 78 lines Merged revisions 254235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) | 72 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248860. As such the dialplan test has been extended: ; non absolute path, not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test) exten => 5040, n, dial(sip/5001) ; absolute path, not combined exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten => 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1, monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ; combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m) exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n, dial(sip/5001) ; combined: changemonitor from no path to non absolute path exten => 5044, 1, monitor(wav,monitor_test6,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before exten => 5044, n, dial(sip/5001) ; non absolute path, combined exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n, dial(sip/5001) ; absolute path, combined exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n, dial(sip/5001) ; no path, combined exten => 5047, 1, monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ; combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n, dial(sip/5001) ; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n, changemonitor(tmp/jeff/monitor_test14) exten => 5049, n, dial(sip/5001) ; combined: changemonitor from no path to absolute exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n, dial(sip/5001) ; combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n, changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ; not combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19) exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n, dial(sip/5001) ; not combined: changemonitor from no path to non absolute exten => 5053, 1, monitor(wav,monitor_test21) exten => 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n, dial(sip/5001) ; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n, dial(sip/5001) ; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n, changemonitor(tmp/jeff/monitor_test25) exten => 5055, n, dial(sip/5001) ; not combined: changemonitor from no path to absolute exten => 5056, 1, monitor(wav,monitor_test26) exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056, n, dial(sip/5001) ; not combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n, changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001) ........ ................ 2010-03-23 22:05 +0000 [r254131] Tzafrir Cohen * tests/Makefile, /: Merged revisions 254001 via svnmerge from http://svn.digium.com/svn/asterisk/trunk ........ r254001 | tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines Change the name of the category 'TEST' to match the name of the subdir ........ 2010-03-23 21:20 +0000 [r254068] Jeff Peeler * main/channel.c, /: Merged revisions 254050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 | jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines Exit native bridging early for greater timing accuracy with warnings This changes native bridging to break one millisecond early so that the more accurate timeval calculations done in the generic bridge can be performed using the bridge config. Currently the time between exiting native bridging slightly late can sometimes cause a large enough discrepancy for warnings to be missed. For the record, 1.4 does not attempt to native bridge at all when warnings are enabled. (closes issue #15815) Reported by: adomjan Review: https://reviewboard.asterisk.org/r/577/ ........ 2010-03-22 19:55 +0000 [r253801] Matthew Nicholson * /, main/features.c: Merged revisions 253800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar 2010) | 11 lines Merged revisions 253799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar 2010) | 4 lines Unconditionally copy the caller's account code to the called party. (related to issue #16331) ........ ................ 2010-03-22 19:06 +0000 [r253714-253760] Tilghman Lesher * /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22 Mar 2010) | 2 lines Update query should be an UPDATE, not a SELECT. ........ * /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22 Mar 2010) | 4 lines Return the list for later manipulation. This fixes an issue with the update procedure. Debugging with mmichelson. ........ * configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged revisions 253712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 | tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines Accomodate equal signs in DSNs and add documentation, based upon mmichelson's feedback. ........ 2010-03-20 17:33 +0000 [r253595-253620] Russell Bryant * cdr/cdr_pgsql.c, main/stdtime/localtime.c, main/tcptls.c, /, main/features.c: Merged revisions 253540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253540 | russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve more compiler warnings on FreeBSD. ........ * apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD. ........ * /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 | russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve a compiler warning on FreeBSD. ........ * channels/chan_dahdi.c, /: Merged revisions 253536 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010) | 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix build issues I had with this module on FreeBSD. ........ 2010-03-19 08:05 +0000 [r253492] Alec L Davis * main/astobj2.c, /: Merged revisions 253490 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 | alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19 lines prevent segfault if bad magic number is encountered. internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic number', but internal_ao2_ref continues on, causing segfault. Although AO2_MAGIC number is checked by INTERNAL_OBJ before internal_ao2_ref is called, A02_MAGIC is being destroyed (or a wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ. internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad magic number. (issue #17037) Reported by: alecdavis Patches: bug17037.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ 2010-03-18 17:54 +0000 [r253257-253346] Leif Madsen * /, apps/app_userevent.c: Merged revisions 253345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253345 | lmadsen | 2010-03-18 12:52:35 -0500 (Thu, 18 Mar 2010) | 7 lines Change usage of pipe to comma in UserEvent docs. Change the example usage of pipe as a separator to comma in the UserEvent documentation. (closes issue #16961) Reported by: jlpedrosa ........ * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010) | 9 lines Update to new Local channel documentation. Add same changes as commit to 1.4, but convert to TeX. (issue #16963) Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz (license 834) ........ 2010-03-17 16:25 +0000 [r253158] Terry Wilson * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_mgcp.c, channels/chan_sip.c, include/asterisk/rtp.h: Revert API change in release branches This re-renames ast_rtp_update_source to ast_rtp_new_source 2010-03-17 00:41 +0000 [r253029-253033] Leif Madsen * main/xmldoc.c, /: Merged revisions 253032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253032 | lmadsen | 2010-03-16 19:40:51 -0500 (Tue, 16 Mar 2010) | 1 line Fix a typo. ........ * configs/say.conf.sample, /: Merged revisions 253028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500 (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010) | 6 lines Add french snipset to say.conf. Add the french snipset to say.conf. (Closes issue #15799) ........ ................ 2010-03-16 23:54 +0000 [r252978] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 | tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines Mask out previous arguments on each nested invocation of Gosub. (closes issue #16758) Reported by: wdoekes Patches: 20100316__issue16758.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/561/ ........ 2010-03-16 19:38 +0000 [r252850] Sean Bright * res/res_clialiases.c, /: Merged revisions 252848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252848 | seanbright | 2010-03-16 15:36:24 -0400 (Tue, 16 Mar 2010) | 10 lines Include an extra newline after "Aliased CLI command" to get back the prompt. The other issue mentioned in this bug will be more difficult to resolve since we have no idea (right now) of knowing if the command that is aliased has been installed yet. (issue #16978) Reported by: jw-asterisk Tested by: seanbright ........ 2010-03-16 19:02 +0000 [r252770] Russell Bryant * utils/Makefile, /: Merged revisions 252767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010) | 13 lines Merged revisions 252766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010) | 6 lines Don't treat warnings as errors for muted. muted supports OS X, but uses functions marked as deprecated in 10.6. However, the functions are still supported, so just ignore the warnings for now and allow the build to proceed. ........ ................ 2010-03-16 18:49 +0000 [r252763] Leif Madsen * configs/extensions.ael.sample, /: Merged revisions 252762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500 (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines Additional extensions.ael global variable fixes. Fixing up a couple more overlapping global variable namespaces shared with extensions.conf.sample. Also noticed a few of the lines that were commented out didn't have the closing semi-colon so I added that as well. (issue #17035) ........ ................ 2010-03-15 21:59 +0000 [r252626] Sean Bright * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 | seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4 lines Resolve a crash in SLATrunk when the specified trunk doesn't exist. Reported by philipp64 in #asterisk-dev. ........ 2010-03-15 21:54 +0000 [r252622] Tilghman Lesher * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions 252619 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010) | 9 lines Merged revisions 252617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010) | 2 lines Uh, yeah. Umask. I'm stupid. ........ ................ 2010-03-15 20:53 +0000 [r252535] Leif Madsen * configs/extensions.ael.sample: Merged revisions 252534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500 (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines Update extensions.ael file to not overlap extensions.conf. Updated the extensions.ael file so the global variables don't overlap those that we have in extensions.conf (sample files). This way unexpected things won't happed hopefully if both pbx_ael and res_config are loaded. (closes issue #17035) Reported by: pprindeville ........ ................ 2010-03-15 05:04 +0000 [r252365-252444] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 | tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines THIS IS NOT PYTHON. Indentation doesn't matter, only braces do. (closes issue #17025) Reported by: smurfix Patches: sip.patch uploaded by smurfix (license 547) ........ * main/asterisk.c, Makefile, contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged revisions 252362 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010) | 11 lines Merged revisions 252361 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010) | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard: https://reviewboard.asterisk.org/r/551/ ........ ................ 2010-03-14 17:48 +0000 [r252317] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar 2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They added a sqlite3_log() function which was conflicting with our function names. (closes issue #17017) Reported by: alephlg ........ 2010-03-13 00:32 +0000 [r252137-252178] Terry Wilson * main/rtp.c: Remove unusued field * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c, channels/chan_mgcp.c, main/channel.c, /, channels/chan_sip.c, channels/chan_skinny.c, include/asterisk/rtp.h, channels/chan_h323.c: Merged revisions 252089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ 2010-03-12 22:05 +0000 [r252090] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 252088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252088 | moy | 2010-03-12 16:57:40 -0500 (Fri, 12 Mar 2010) | 1 line add missing mfcr2_skip_category setting ........ 2010-03-12 19:50 +0000 [r251994] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010) | 8 lines Don't override a user option with the global option. (closes issue #16849) Reported by: ip-rob Patches: 20100311__issue16849.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob ........ 2010-03-12 19:49 +0000 [r251991] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 251946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251946 | rmudgett | 2010-03-12 13:05:40 -0600 (Fri, 12 Mar 2010) | 1 line Doxegen this chan_dahdi lock. ........ 2010-03-11 21:08 +0000 [r251879-251887] Tilghman Lesher * apps/app_exec.c, /: Merged revisions 251884 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 | tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines Because ExecIf needs to reprocess arguments, it's best if we don't remove quotes during parsing. (closes issue #16905) Reported by: ip-rob Patches: 20100303__issue16905.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob ........ * apps/app_system.c, /: Merged revisions 251877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 | tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines If the argument to the system application is quoted, ensure we remove the quotes before trying to execute. (closes issue #16842) Reported by: ip-rob Patches: 20100310__issue16842.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob ........ 2010-03-11 Leif Madsen * Asterisk 1.6.2.6 released 2010-03-05 Leif Madsen * Asterisk 1.6.2.6-rc2 released 2010-03-05 Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman | 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in ODBC query. (closes issue #16953) Reported by: elguero Patches: app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37) ........ 2010-03-04 Leif Madsen * Asterisk 1.6.2.6-rc1 released 2010-03-03 21:24 +0000 [r250610] Leif Madsen * doc/tex/localchannel.tex, /: Merged revisions 250609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010) | 11 lines Update existing Local channel documentation. A complete re-write of the Local channel documentation has been performed, with the existing information from localchannel.txt and localchannel.tex merged in. (closes issue #16637) Reported by: kobaz Patches: localchannel.tex uploaded by lmadsen (license 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson ........ 2010-03-03 19:13 +0000 [r250484] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines Make sure to clear red alarm after polarity reversal. From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ ................ 2010-03-03 18:05 +0000 [r250253-250396] David Vossel * channels/chan_iax2.c, /: Merged revisions 250395 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets When Asterisk receives an IAX2 TXREQ packet, try_transfer() will call store_by_transfercallno() to link the chan_iax2_pvt struct into iax_transfercallno_pvts. If a duplicate TXREQ packet is received for the same call, the pvt struct will be linked into iax_transfercallno_pvts multiple times. This patch fixes this. Thanks rain for debugging this and providing a patch! (closes issue #16904) Reported by: rain Patches: iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested by: rain, dvossel ........ ................ * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 | dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines fixes signed to unsigned int comparision issue for FaxMaxDatagram value. ........ 2010-03-02 21:10 +0000 [r249953-250052] Leif Madsen * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010) | 8 lines Update IMAP documentation. Update the IMAP documentation to make it clear that storing voicemails in the same folder as a large number of emails could potentially cause significant slow downs when writing or retrieving voicemails. (issue #16704) Reported by: TimeHider Tested by: lmadsen, TimeHider ........ * configs/cdr.conf.sample: Merged revisions 250045 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500 (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines Update documentation to clarify purpose of unanswered option. (closes issue #16267) Reported by: elsto Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested by: davidw, elsto ........ ................ * doc/tex/configuration.tex, /: Merged revisions 250037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 Mar 2010) | 4 lines Update documentation to not imply we support overriding options. (closes issue #16855) Reported by: davidw ........ * apps/app_directory.c, /: Merged revisions 249950 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010) | 4 lines Fix literal values wrapped in documentation. (closes issue #16145) Reported by: tilghman ........ 2010-03-02 19:50 +0000 [r249952] Alec L Davis * UPGRADE-1.6.txt, main/editline/makelist.in, apps/app_echo.c, UPGRADE.txt: revert ability to exit echo app caused a regression, as only supported VOICE, not VIDEO etc. (issue #16880) 2010-03-02 19:26 +0000 [r249916-249933] Leif Madsen * /, main/features.c: Merged revisions 249925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 | lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines Add missing description of the PARKINGLOT variable in XML documentation. (closes issue #16743) Reported by: snuffy Patches: parkingdoc.diff uploaded by snuffy (license 35) ........ * /, pbx/pbx_dundi.c: Merged revisions 249912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 | lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines Convert some DUNDI functions to XML documentation. (closes issue #16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by snuffy (license 35) ........ 2010-03-02 19:12 +0000 [r249895] David Vossel * channels/chan_console.c, channels/chan_gtalk.c, channels/chan_oss.c, channels/misdn_config.c, include/asterisk/abstract_jb.h, configs/alsa.conf.sample, channels/chan_jingle.c, channels/chan_usbradio.c, channels/chan_dahdi.c, channels/chan_skinny.c, configs/mgcp.conf.sample, main/abstract_jb.c, channels/chan_h323.c, channels/chan_alsa.c, configs/sip.conf.sample, channels/chan_mgcp.c, channels/chan_unistim.c, configs/console.conf.sample, configs/chan_dahdi.conf.sample, channels/chan_local.c, configs/oss.conf.sample, channels/chan_sip.c, /, configs/usbradio.conf.sample, configs/misdn.conf.sample: Merged revisions 249893 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ 2010-03-02 19:09 +0000 [r249894] Leif Madsen * /, apps/app_confbridge.c: Merged revisions 249892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010) | 1 line Fix several XML documentation validate errors. ........ 2010-03-02 09:05 +0000 [r249844] Alec L Davis * apps/app_echo.c: fixes ability to exit echo app when called from a ISDN channel, null frames prevent '#' exit. Now only echo back VOICE and DTMF frames (issue #16880) Reported by: alecdavis Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis 2010-03-01 19:40 +0000 [r249675] Sean Bright * apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar 2010) | 11 lines Fix crash in app_voicemail related to message counting. We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *' causing a segfault. (closes issue #16921) Reported by: whardier Patches: 20100301_issue16921.patch uploaded by seanbright (license 71) Tested by: whardier ........ ................ 2010-03-01 18:47 +0000 [r249625] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010) | 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP compile once again. ........ 2010-03-01 17:25 +0000 [r249580] Jeff Peeler * channels/chan_local.c, /: Merged revisions 249538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) | 11 lines Modify queued frames from local channels to not set the other side to up In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor ........ ................ 2010-02-28 20:51 +0000 [r249407-249493] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010) | 5 lines Fix unit test that Alec Davis broke. (closes issue #16927) Reported by: alecdavis ........ * apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions 249405 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 | tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines Properly document voicemail API documents. Also fix a crash reported via the -dev list. ........ 2010-02-27 23:04 +0000 [r249321] Alec L Davis * channels/chan_dahdi.c: overlap receiving: automatically send CALL PROCEEDING when dialplan starts Following Q.931 5.2.4 When the user has determined that sufficient call information has been received the user shall stop T302 and send CALL PROCEEDING to the network. Previously timeouts were possible if the dialplan took a long time to issue any response back to the network. Verified that our local TELCO also does the same. (issue #16789) Reported by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis 2010-02-27 14:10 +0000 [r249238] Kevin P. Fleming * channels/chan_iax2.c, /: Merged revisions 249235 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500 (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 Feb 2010) | 1 line add a reference to the now-published IAX2 RFC ........ ................ 2010-02-26 18:49 +0000 [r249190] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010) | 18 lines Cleanups to fix bugs in the VM count API functions. - Urgent voicemails were not attached, because the attachment code looked in the wrong folder. - Urgent voicemails were sometimes counted twice when displaying the count of new messages. - Backends were inconsistent as to which voicemails each API counted. (closes issue #15654) Reported by: tomo1657 Patches: 20100225__issue15654.diff.txt uploaded by tilghman (license 14) Tested by: tilghman (closes issue #16448) Reported by: hevad Review: https://reviewboard.asterisk.org/r/525/ ........ 2010-02-26 17:06 +0000 [r249104] Mark Michelson * /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb 2010) | 14 lines Merged revisions 249100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. (closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) ........ ................ 2010-02-25 23:12 +0000 [r248955] Jeff Peeler * res/res_monitor.c, /: Merged revisions 248952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010) | 24 lines Merged revisions 248860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248757. As such the dialplan test has been extended: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning exten => 5044, n, dial(sip/5001) ........ ................ 2010-02-25 22:42 +0000 [r248949] Mark Michelson * /, main/acl.c: Merged revisions 248946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 | mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5 lines Fix incorrect ACL behavior when CIDR notation of "/0" is used. AST-2010-003 ........ 2010-02-25 21:25 +0000 [r248864] Tilghman Lesher * main/asterisk.c, /: Merged revisions 248861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) | 15 lines Some platforms clear /var/run at boot, which makes connecting a remote console... difficult. Previously, we only created the default /var/run/asterisk directory at install time. While we could create it in the init script, that would not work for those who start asterisk manually from the command line. So the safest thing to do is to create it as part of the Asterisk boot process. This also changes the ownership of the directory, because the pid and ctl files are created after we setuid/setgid. (closes issue #16802) Reported by: Brian Patches: 20100224__issue16802.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir ........ ................ 2010-02-25 18:52 +0000 [r248797] Jeff Peeler * res/res_monitor.c, /: Merged revisions 248793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines Ensure that monitor recordings are written to the correct location. Recordings should be placed in the monitor directory when a non-absolute path is used. Exact dialplan used for testing: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) ABE-2101 ........ ................ 2010-02-24 21:29 +0000 [r248643] Tilghman Lesher * /, main/logger.c: Merged revisions 248584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010) | 14 lines Merged revisions 248582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) | 7 lines Remove color code sequences from verbose messages that go to logfiles. (closes issue #16786) Reported by: dodo Patches: logger2.patch uploaded by dodo (license 989) Tested by: tilghman ........ ................ 2010-02-23 16:37 +0000 [r248398] David Vossel * /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines Merged revisions 248396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines fixes invite with replaces deadlock (closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel ........ ................ 2010-02-19 19:07 +0000 [r248011] Tilghman Lesher * channels/chan_console.c, main/loader.c, /: Merged revisions 228798 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228798 | tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix various problems detected with Valgrind. * chan_console accessed pvts after deallocation. * The module loader did not check usecount on shutdown, which led to chan_iax2 reading a timer that was already unloaded. (closes issue #16062) Reported by: alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ 2010-02-19 19:00 +0000 [r248005] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 248003 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1 line mfcr2 issue 0016844 - Fix portability bit fields and make mfcr2_immediate_accept work again, reported and patched by korihor ........ 2010-02-19 18:45 +0000 [r248004] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600 (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ ................ 2010-02-19 17:41 +0000 [r247916] David Vossel * /, channels/chan_sip.c: Merged revisions 247915 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 | dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines handle_request_invite revise comment, fix coding guideline issues I'm working with this code right now trying to analyze a deadlock. This change is just to clean up a few things before I make a more complex patch. ........ 2010-02-18 23:15 +0000 [r247792-247845] Tilghman Lesher * res/res_speech.c, /: Merged revisions 247841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 | tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines Revert an errant part of a previous cleanup, to fix a memory corruption issue. (closes issue #16368) Reported by: thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf (license 955) ........ * /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 | tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 lines If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. NULL means the value is not specified for the column, which normally means the driver uses whatever is the default value. However, on MySQL, placing a NULL in either a float or integer column results in a retrieval of the 0 value. Hence, users get an errant error on load. This patch suppresses that error and makes the value as if it was not there. Note that this cannot be done in the realtime driver, because the lack of difference between NULL and 0 can only be intepreted correctly by the driver itself. If we did it in the realtime driver, then it would be effectively impossible to set any realtime field to 0, because it would act as if the field were unspecified and possibly take on a different value. (closes issue #16683) Reported by: wdoekes ........ 2010-02-18 21:25 +0000 [r247737-247776] David Vossel * /, bridges/bridge_softmix.c: Merged revisions 247770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010) | 9 lines fixes confbridge crash when no timing module is loaded. (closes issue #16471) Reported by: kjotte Patches: M16471.diff uploaded by junky (license 177) Tested by: kjotte, junky ........ * apps/app_queue.c, /: Merged revisions 247736 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 | dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines fixes Queue with C option crash (closes issue #16475) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) ........ 2010-02-18 19:41 +0000 [r247653] Matthew Nicholson * /, main/features.c: Merged revisions 247652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb 2010) | 13 lines Merged revisions 247651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb 2010) | 6 lines Copy the calling party's account code to the called party if they don't already have one. (closes issue #16331) Reported by: bluefox Tested by: mnicholson ........ ................ 2010-02-18 16:58 +0000 [r247506-247512] Leif Madsen * README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500 (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010) | 1 line Add additional link to best practices document per jsmith. ........ ................ * README-SERIOUSLY.bestpractices.txt (added): Merged revisions 247503 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010) | 18 lines Merged revisions 247502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) | 10 lines Add best practices documentation. (issue #16808) Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/507/ ........ ................ 2010-02-18 04:21 +0000 [r247426] Russell Bryant * sounds/Makefile, Makefile, /: Merged revisions 247423 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247423 | russell | 2010-02-17 22:20:11 -0600 (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) | 10 lines Tweak argument handling for wget in the sounds Makefile. 1) Fix the check to see if we are using wget to not be full of fail. The configure script populates this variable with the absolute path to wget if it is found, so it didn't work. 2) Allow some extra arguments to be passed in for wget. This is just a simple change to allow our Bamboo build script to tell wget to be quiet and not fill up our logs with download status output. ........ ................ 2010-02-17 21:32 +0000 [r246989-247337] Mark Michelson * include/asterisk/strings.h, main/strings.c, /: Merged revisions 247335 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 | mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20 lines Fix two problems in ast_str functions found while writing a unit test. 1. The documentation for ast_str_set and ast_str_append state that the max_len parameter may be -1 in order to limit the size of the ast_str to its current allocated size. The problem was that the max_len parameter in all cases was a size_t, which is unsigned. Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the max_len parameter to be ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an off-by-one error in the case where we attempted to write a string larger than the current allotted size to a string when -1 was passed as the max_len parameter. When trying to write more than the allotted size, the ast_str's __AST_STR_USED was set to 1 higher than it should have been. Thanks to Tilghman for quickly spotting the offending line of code. Oh, and the unit test that I referenced in the top line of this commit will be added to reviewboard shortly. Sit tight... ........ * apps/app_queue.c, /: Merged revisions 247169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb 2010) | 9 lines Merged revisions 247168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls. ........ ................ * main/strings.c, /: Merged revisions 247076 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247076 | mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12 lines Add va_end calls to __ast_str_helper. According to the man page for stdarg(3), "Each invocation of va_copy() must be matched by a corresponding invocation of va_end() in the same function." There were several cases in __ast_str_helper where va_copy was not matched with a corresponding call to va_end. ........ * include/asterisk/strings.h, /: Merged revisions 246985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue, 16 Feb 2010) | 3 lines Add some clarifying documentation to the ast_str_set and ast_str_append functions. ........ 2010-02-16 21:03 +0000 [r246900-246982] David Vossel * main/tcptls.c, /: Merged revisions 246980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 | dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines warning message if openssl support is missing while attempting tls connection (closes issue #16673) Reported by: michaesc Patches: tls_error_msg.diff uploaded by dvossel (license 671) ........ * main/channel.c: fixes merge error with Monitor calculation fix * main/channel.c, /: Merged revisions 246899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 | dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines fixes sample rate conversion issue with Monitor application When using ast_seekstream with the read/write streams of a monitor, the number of samples we are seeking must be of the same rate as the stream or the jump calculation will be incorrect. This patch adds logic to correctly convert the number of samples to jump to the sample rate the read/write stream is using. For example, if the call is G722 (16khz) and the read/write stream is recording a 8khz wav, seeking 320 samples of 16khz audio is not the same as seeking 320 samples of 8khz audio when performing the ast_seekstream on the stream. ABE-2044 ........ 2010-02-15 23:45 +0000 [r246713] Tilghman Lesher * Makefile, /: Merged revisions 246710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010) | 12 lines Merged revisions 246709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) | 5 lines Make the menuselect instructions correct by allowing 'make menuselect' to actually solve dependency problems. (Previously, it would fail out again with the same message about running 'make menuselect', which was NOT at all helpful.) ........ ................ 2010-02-12 23:33 +0000 [r246547] David Vossel * main/channel.c, /: Merged revisions 246546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010) | 21 lines Merged revisions 246545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) | 16 lines lock channel during datastore removal On channel destruction the channel's datastores are removed and destroyed. Since there are public API calls to find and remove datastores on a channel, a lock should be held whenever datastores are removed and destroyed. This resolves a crash caused by a race condition in app_chanspy.c. (closes issue #16678) Reported by: tim_ringenbach Patches: datastore_destroy_race.diff uploaded by tim ringenbach (license 540) Tested by: dvossel ........ ................ 2010-02-12 19:08 +0000 [r246464] Jason Parker * main/channel.c: Fix some silly formatting that made my head hurt. 2010-02-10 21:28 +0000 [r246199-246207] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 246204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010) | 2 lines Fussy compiler on another machine... ........ * /, funcs/func_strings.c: Merged revisions 246200 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010) | 2 lines Fix weird issue with unit tests on optimized build - turned out to be a signing issue. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, res/res_agi.c: Merged revisions 246030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246030 | tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12 lines Solaris doesn't like outputting a NULL to a %s in format strings. Detect all platforms that don't like that, either, and ensure that when documentation is missing, we pass a non-NULL pointer when outputting the corresponding documentation. (closes issue #16689) Reported by: bklang Patches: 20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/497/ ........ 2010-02-10 17:51 +0000 [r246117] David Vossel * apps/app_queue.c, /: Merged revisions 246116 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010) | 14 lines Merged revisions 246115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) | 8 lines fixes random deadlock in app_queue with use_weight during reload (closes issue #16677) Reported by: tim_ringenbach Patches: app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540) ........ ................ 2010-02-10 16:58 +0000 [r246073] Jeff Peeler * channels/chan_local.c, /: Merged revisions 246070 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010) | 22 lines Change channel state on local channels for busy,answer,ring. Previously local channels channel state never changed. This became problematic when the state of the other side of the local channel was lost, for example during a masquerade. Changing the state of the local channel allows for the scenario to be detected when the channel state is set to ringing, but the peer isn't ringing. The specific problem scenario is described in 164201. Although this was noted on one of the issues, here is the tested dialplan verified to work: exten => 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) exten => *9700,n,wait(3) ;3 works, 1 did not exten => *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did not exten => 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes issue #14992) Reported by: davidw ........ 2010-02-10 15:38 +0000 [r245948-246025] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 246022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010) | 2 lines Enable warnings on atypical conditions for the FILTER function (suggested by mmichelson on the -dev list). ........ * configs/extensions.conf.sample, /, funcs/func_strings.c: Merged revisions 245945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010) | 9 lines Merged revisions 245944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) | 2 lines Include examples of FILTER usage in extension patterns where a "." may be a risk. ........ ................ 2010-02-09 23:11 +0000 [r245794] David Vossel * channels/chan_iax2.c, /: Merged revisions 245793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600 (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 = 32768 which is the maximum allowed iax2 callnumber. Creating the iaxs and iaxsl array of size 32768 means the maximum callnumber is actually out of bounds. This causes a nasty crash. (closes issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded by dvossel (license 671) ........ ................ 2010-02-09 18:09 +0000 [r245732] Tilghman Lesher * /, apps/app_fax.c: Merged revisions 245729 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 | tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines Ensure frames are only freed once. (closes issue #16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt uploaded by tilghman (license 14) Tested by: kenny, bloodoff, misaksen ........ 2010-02-09 17:43 +0000 [r245728] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 245727 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245727 | mnicholson | 2010-02-09 11:40:04 -0600 (Tue, 09 Feb 2010) | 2 lines This commit removes an extra newline in T.38 generated SDP packets. This bug was caused by the fix introduced in r243860. (closes issue #16766) Reported by: raivisr Patches: t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96) Tested by: raivisr ........ 2010-02-09 16:26 +0000 [r245683] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 245680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 | kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8 lines Don't offer MMR or JBIG transcoding during T.38 negotiation. After further discussion with Steve Underwood, we should not (yet) be offering to receive MMR or JBIG transcoded streams from T.38 endpoints. A future spandsp release will support those features, and then they can be enabled during negotiation ........ 2010-02-08 23:47 +0000 [r245626] Russell Bryant * /, main/event.c: Merged revisions 245624 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245624 | russell | 2010-02-08 17:43:00 -0600 (Mon, 08 Feb 2010) | 5 lines Fix return value of get_ie_str() and get_ie_str_hash() for non-existent IE. I found this bug while developing a unit test for event allocation. Testing is awesome. ........ 2010-02-08 22:46 +0000 [r245581] Tilghman Lesher * channels/Makefile, /, main/Makefile: Merged revisions 245578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08 Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and channels/ Makefiles. They were previously passed correctly, but they simply weren't used. This caused issues with various platforms whose builds needed to pass special linker flags via the configure script. (closes issue #16596) Reported by: pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by pprindeville (license 347) Tested by: tilghman ........ 2010-02-08 20:43 +0000 [r245500] Jason Parker * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245497 | qwell | 2010-02-08 14:41:05 -0600 (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) | 4 lines Remove reference of documentation in source directory. People don't always build Asterisk from source (distro packages, anybody?). ........ ................ 2010-02-05 19:27 +0000 [r245097] Jeff Peeler * contrib/firmware (removed), /, LICENSE: Merged revisions 245090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600 (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb 2010) | 5 lines Remove contrib/firmware directory as it is empty Remove explicit license for IAXy firmware as it is no longer included in the tree ........ ................ 2010-02-05 17:10 +0000 [r244930] Sean Bright * main/asterisk.c, /: Merged revisions 244927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb 2010) | 9 lines Merged revisions 244926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb 2010) | 1 line Update main copyright date. ........ ................ 2010-02-03 19:28 +0000 [r244555] Mark Michelson * main/sched.c, /: Merged revisions 244547 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244547 | mmichelson | 2010-02-03 13:26:53 -0600 (Wed, 03 Feb 2010) | 3 lines Initialize counters in ast_sched_report so that resulting data is not bogus. ........ 2010-02-03 18:47 +0000 [r244508] Tilghman Lesher * channels/chan_dahdi.c, /, main/ast_expr2f.c: Merged revisions 244505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244505 | tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines The chanvar= setting should inherit the entire list of variables, not just the first one. (closes issue #16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded by raarts (license 937) Tested by: raarts ........ 2010-02-02 22:29 +0000 [r244445] David Vossel * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 244443 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 | dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported by: krn (closes issue #16724) Reported by: barthpbx (closes issue #16517) Reported by: bklang (closes issue #16485) Reported by: elsto ........ 2010-02-02 20:35 +0000 [r244395] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 244393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244393 | tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18 lines Properly respect GOSUB_RESULT as to what to do with the master channel. Previously, we would parse GOSUB_RESULT, but not actually do anything with it. (closes issue #16686) Reported by: bklang Patches: app_dial-respect-gosub_result.patch uploaded by bklang (license 919) (with modifications) ........ 2010-02-02 Leif Madsen * Release Asterisk 1.6.2.2 * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash occurs when the FaxMaxDatagram field is omitted from the SDP as well. 2010-01-14 Leif Madsen * Release Asterisk 1.6.2.1 2010-01-08 Leif Madsen * Release Asterisk 1.6.2.1-rc1 2010-01-07 21:17 +0000 [r238499] Tilghman Lesher * channels/chan_console.c, channels/chan_oss.c, main/poll.c, channels/chan_usbradio.c, include/asterisk/utils.h, /, channels/chan_sip.c, channels/chan_alsa.c: Merged revisions 209400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. (closes issue #16251) Reported by: asgaroth ........ 2010-01-07 20:17 +0000 [r238362-238416] David Vossel * channels/chan_iax2.c, /: Merged revisions 238412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600 (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in chan_iax A signed short was used to represent a callnumber. This is makes it possible to attempt to access the iaxs array with a negative index. (closes issue #16565) Reported by: jensvb ........ ................ * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 | dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines Change in sip show channels display format allowing more digits for CID (closes issue #16459) Reported by: Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins (license 953) ........ * apps/app_queue.c, /: Merged revisions 238361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 | dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines cli 'queue show' formatting fix. queue name was truncated over 12 characters (closes issue #16078) Reported by: RoadKill Patches: quequename_limit.patch uploaded by ppyy (license 906) Tested by: dvossel ........ 2010-01-07 09:49 +0000 [r238349] Tzafrir Cohen * configs/sip.conf.sample, /: Merged revisions 238313 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) | 2 lines Document the usefulness of explicit udp:// in the register string ........ 2010-01-06 21:48 +0000 [r238234] Tilghman Lesher * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010) | 11 lines Merged revisions 238230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010) | 4 lines Revise documentation on disposition values to the actual values used. (closes issue #16289) Reported by: wdoekes ........ ................ 2010-01-06 20:40 +0000 [r238137-238185] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 238181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 | jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines Fix misreverting from 177158. (closes issue #15725) Reported by: shanermn Patches: v1-15725.patch uploaded by dimas (license 88) Tested by: shanermn ........ * /, main/features.c: Merged revisions 238134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 | jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines Fix channel name comparison for bridge application. The channel name comparison was not comparing the whole string and therefore if one channel name was a substring of the other, the bridge would fail. (closes issue #16528) Reported by: telecos82 Patches: res_features_r236843.diff uploaded by telecos82 (license 687) ........ 2010-01-06 15:22 +0000 [r238013] Russell Bryant * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010) | 14 lines Merged revisions 238009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) | 7 lines Resolve a crash due to an ast_frame not being fully initialized. (closes issue #16531) Reported by: john8675309 (closes SWP-615) ........ ................ 2010-01-06 06:54 +0000 [r237969] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 237968 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237968 | tilghman | 2010-01-06 00:53:23 -0600 (Wed, 06 Jan 2010) | 2 lines Whoa, duplicate setting (dead code). ........ 2010-01-05 23:10 +0000 [r237924] Kinsey Moore * apps/app_test.c: Add a wait to ensure TestServer thinks it has finished sending the final digit. This was previously committed to 1.4, 1.6.0, 1.6.1, and trunk just after 1.6.2 was created (and missed). 1.6.2 also needs this patch to resolve the bug. (closes issue #16550) Reported by: opticron Patches: apptest.diff uploaded by opticron (license 267) 2010-01-05 23:09 +0000 [r237840-237921] David Vossel * apps/app_queue.c, /: Merged revisions 237920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 | dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines fixes holdtime playback issue in app_queue When reporting hold time, the number of seconds should be mod 60. Otherwise audio playback could be something like "2 minutes 123 seconds" rather than "2 minutes 3 seconds". Also, the "minute" sound file is missing, so for the moment until that file can be created the "minutes" file is used instead. (closes issue #16168) Reported by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by nickilo (license ) Tested by: nickilo, wonderg ........ * main/pbx.c, /: Merged revisions 237839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 | dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines fixes subscriptions being lost after 'module reload' During a module reload if multiple extension configs are present, such as both extensions.conf and extensions.ael, watchers for one config's hints will be lost during the merging of the other config. This happens because hint watchers are only preserved for the current config being merged. The old context list is destroyed after the merging takes place, meaning any watchers that were not perserved will be removed. Now all hints are preserved during merging regardless of what config file is being merged. These hints are only restored if they are present within the new context list. (closes issue #16093) Reported by: jlaroff ........ 2010-01-05 17:25 +0000 [r237743] Russell Bryant * /, main/utils.c: Merged revisions 237699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010) | 14 lines Merged revisions 237697 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010) | 7 lines Change a NOTICE log message to DEBUG where it belongs. (closes issue #16479) Reported by: alexrecarey (closes SWP-577) ........ ................ 2010-01-05 16:09 +0000 [r237657] Michiel van Baak * apps/app_mixmonitor.c, /: Merged revisions 237656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010) | 6 lines Make CLI command 'mixmonitor start|stop work again. (closes issue #16534) Reported by: jlaguilar Fix as suggested by jlaguilar in the bugreport ........ 2010-01-04 21:52 +0000 [r237409-237577] Tilghman Lesher * /, main/say.c: Merged revisions 237574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010) | 13 lines Merged revisions 237573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010) | 6 lines Bounds checking for input string (closes issue #16407) Reported by: qwell Patches: 20100104__issue16407.diff.txt uploaded by tilghman (license 14) ........ ................ * main/pbx.c, /: Merged revisions 237494 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010) | 15 lines Merged revisions 237493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010) | 8 lines Regression in issue #15421 - Pattern matching (closes issue #16482) Reported by: wdoekes Patches: astsvn-16482-betterfix.diff uploaded by wdoekes (license 717) 20091223__issue16482.diff.txt uploaded by tilghman (license 14) Tested by: wdoekes, tilghman ........ ................ * main/config.c, /: Merged revisions 237414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 | tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines Oops, didn't compile (thanks, kpfleming) ........ * main/config.c, /: Merged revisions 237410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 | tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines Further reduce the encoded blank values back to blank in the realtime API. (closes issue #16533) Reported by: sergee Patches: 200100104__issue16533.diff.txt uploaded by tilghman (license 14) Tested by: sergee ........ * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged revisions 237406 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) | 23 lines Merged revisions 237405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines Add a flag to disable the Background behavior, for AGI users. This is in a section of code that relates to two other issues, namely issue #14011 and issue #14940), one of which was the behavior of Background when called with a context argument that matched the current context. This fix broke FreePBX, however, in a post-Dial situation. Needless to say, this is an extremely difficult collision of several different issues. While the use of an exception flag is ugly, fixing all of the issues linked is rather difficult (although if someone would like to propose a better solution, we're happy to entertain that suggestion). (closes issue #16434) Reported by: rickead2000 Patches: 20091217__issue16434.diff.txt uploaded by tilghman (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: rickead2000 ........ ................ 2010-01-04 16:50 +0000 [r237328] David Vossel * apps/app_queue.c, /: Merged revisions 237327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237327 | dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines app_queue segfaults if realtime field uniqueid is NULL (closes issue #16385) Reported by: haakon Patches: app_queue.c.patch uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by dvossel (license 671) Tested by: haakon ........ 2010-01-04 16:27 +0000 [r237326] Jeff Peeler * /, res/res_agi.c: Merged revisions 237323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 | jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines Fix timeout for AGI command speech recognize. (closes issue #16297) Reported by: semond ........ 2010-01-04 16:21 +0000 [r237322] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 237319 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600 (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) | 3 lines It's also possible for the Local channel to directly execute an Application. Reviewboard: https://reviewboard.asterisk.org/r/452/ ........ ................ 2010-01-02 10:03 +0000 [r237139] Olle Johansson * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10 lines Merged revisions 237135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 lines Release memory of the contact acl before unloading module ........ ................ 2009-12-30 22:00 +0000 [r236985] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 236982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600 (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) | 9 lines Don't queue frames to channels that have no means to process them. (closes issue #15609) Reported by: aragon Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14) Tested by: aragon Review: https://reviewboard.asterisk.org/r/452/ ........ ................ 2009-12-30 21:13 +0000 [r236899-236905] Jeff Peeler * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 | jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines One more LOW_MEMORY compile fix. ........ * main/cli.c, /: Merged revisions 236893 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236893 | jpeeler | 2009-12-30 14:34:41 -0600 (Wed, 30 Dec 2009) | 11 lines Fix compiling with LOW_MEMORY. Modified handle_verbose to be LOW_MEMORY aware. (closes issue #16381) Reported by: michael_iedema Patches: ast_complete_source_filename.patch uploaded by michael iedema (license 942) modified by me ........ 2009-12-30 17:56 +0000 [r236804-236850] Tilghman Lesher * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009) | 4 lines When the field is blank, don't warn about the field being unable to be coerced, just skip the column. (closes http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html) Reported by Nic Colledge on the -dev list, fixed by me. ........ * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 | tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines Shut down the SIP session timers more gracefully, in order to prevent a possible crash. (closes issue #16452) Reported by: corruptor Patches: 20091221__issue16452.diff.txt uploaded by tilghman (license 14) Tested by: corruptor ........ 2009-12-28 22:13 +0000 [r236716] Jason Parker * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec 2009) | 8 lines Allow "REMAINDER" to function properly in expressions. (closes issue #16427) Reported by: wdoekes Patches: ast16-reminder-remainder.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ 2009-12-28 17:40 +0000 [r236670] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009) | 4 lines Use recommended option, not deprecated option. (closes issue #16515) Reported by: ManChicken ........ 2009-12-28 15:31 +0000 [r236513-236635] Sean Bright * include/asterisk/threadstorage.h, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 236613 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec 2009) | 14 lines Merged revisions 236585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces. There was conditional code (based on build platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add a configure-time check for it. ........ ................ * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec 2009) | 19 lines Merged revisions 236509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines Avoid a crash with large numbers of MeetMe conferences. Similar to changes made to Queue(), when we have large numbers of conferences in meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and crash, so instead just use a single fixed buffer. (closes issue #16509) Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ 2009-12-27 18:22 +0000 [r236437] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600 (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 Dec 2009) | 2 lines Turn on colors in the daemon, since there's many requests for it on Ubuntu. ........ ................ 2009-12-26 15:32 +0000 [r236361] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 236358 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec 2009) | 9 lines Merged revisions 236357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec 2009) | 1 line update to latest releases with zero uid/gid ........ ................ 2009-12-23 18:27 +0000 [r236189-236303] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 | tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines AGI may be invoked from outside the dialplan (closes issue #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ * /, res/res_agi.c: Merged revisions 236186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009) | 11 lines Merged revisions 236184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009) | 4 lines If EXEC only gets a single argument, don't crash when the second is used. (closes issue #16504) Reported by: bklang ........ ................ 2009-12-22 17:04 +0000 [r236064] David Vossel * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009) | 18 lines Merged revisions 236062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines fixes issue with p->method incorrectly set to ACK It is possible for a second ACK to come in for a retransmitted message. If an ack does not match an unacked message in our queue, restore the previous p->method as this ACK is completely ignored. (closes issue #16295) Reported by: omolenkamp Patches: issue16295_v2.diff uploaded by dvossel (license 671) ........ ................ 2009-12-21 19:58 +0000 [r235944] Jeff Peeler * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009) | 20 lines Merged revisions 235940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) | 13 lines Change Monitor to not assume file to write to does not contain pathing. 227944 changed the fname_base argument to always append the configured monitor path. This change was necessary to properly compare files for uniqueness. If a full path is given though, nothing needs to be appended and that is handled correctly now. (closes issue #16377) (closes issue #16376) Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch uploaded by dant (license 670) ........ ................ 2009-12-21 17:11 +0000 [r235826] Tilghman Lesher * /, main/features.c: Merged revisions 235822 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009) | 15 lines Merged revisions 235821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009) | 8 lines Send parking lot announcement to the channel which parked the call, not the park-ee. (closes issue #16234) Reported by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded by tilghman (license 14) 20091221__issue16234__1.4.diff.txt uploaded by tilghman (license 14) Tested by: yeshuawatso ........ ................ 2009-12-20 08:58 +0000 [r235775] Alec L Davis * main/dsp.c: restarts busydetector (if enabled) when DTMF is received after call is bridged. (closes issue #16389) Reported by: alecdavis Tested by: alecdavis Patch dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585) 2009-12-18 23:04 +0000 [r235665] Jeff Peeler * main/channel.c, /, include/asterisk/cdr.h: Merged revisions 235660 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009) | 55 lines Merged revisions 235635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is simple in that it reorders the disposition defines so that the fix for issue 12946 works properly (the default CDR disposition was changed to AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all CDR records are written. The side effects of CDR changes are scary, so I'm documenting the test cases performed to attempt to catch any regressions. The following tests were all performed using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A (Both SIP and features) A calls B A blind transfers to C Hangup C (Both SIP and features) A calls B A attended transfers to C Hangup C A calls B A attended transfers to C (SIP) C blind transfers to A (features) Hangup A All of the test scenario CDRs matched. The following tests were performed just with the patch to ensure proper operation (with unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue #16180) Reported by: aatef Patches: bug16180.patch uploaded by jpeeler (license 325) ........ ................ 2009-12-18 22:42 +0000 [r235576-235659] Tilghman Lesher * /, configure, configure.ac: Merged revisions 235656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600 (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18 Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion ........ ................ * /, configure, configure.ac: Merged revisions 235573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235573 | tilghman | 2009-12-18 15:19:43 -0600 (Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009) | 2 lines Point to the typical missing package, not the cryptic "termcap support". ........ ................ 2009-12-17 23:22 +0000 [r235522] Joshua Colp * /, channels/chan_sip.c: Merged revisions 235521 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r235521 | file | 2009-12-17 19:21:07 -0400 (Thu, 17 Dec 2009) | 3 lines Remove some old code for going to the 'fax' extension when a T.38 switchover occurs. This would have already happened when we detected the CNG tone so this was basically a noop. ........ 2009-12-17 Leif Madsen * Release Asterisk 1.6.2.0 2009-12-09 Leif Madsen * Release Asterisk 1.6.2.0-rc8 2009-12-08 18:33 +0000 [r233731] Tilghman Lesher * res/res_musiconhold.c, /: Merged revisions 233718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009) | 8 lines Find another ref leak and change how we manage module references. (closes issue #16388) Reported by: parisioa Patches: 20091208__issue16388.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman Review: https://reviewboard.asterisk.org/r/442/ ........ 2009-12-08 18:04 +0000 [r233694] Russell Bryant * formats/format_sln16.c, formats/format_wav_gsm.c, formats/format_siren7.c, formats/format_ilbc.c, formats/format_vox.c, formats/format_pcm.c, formats/format_h263.c, formats/format_g723.c, formats/format_h264.c, formats/format_siren14.c, formats/format_jpeg.c, formats/format_g726.c, formats/format_gsm.c, formats/format_g729.c, /, formats/format_sln.c, formats/format_wav.c, formats/format_ogg_vorbis.c: Merged revisions 233692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009) | 16 lines Set a module load priority for format modules. A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2. The fix for 1.4 and 1.6.0 will require a different approach since the module load priority functionality is not present in the module API. (issue #16412) Reported by: jiddings ........ 2009-12-08 07:41 +0000 [r233689] TransNexus OSP Development * apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6. 2009-12-07 23:54 +0000 [r233615] Atis Lezdins * contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines Fix compatibility with valgrind 3.3 and older. (noticed in issue #16388) Reported by: parisioa Patches: valgrind.supp uloaded by atis (license 242) Tested by: atis, parisioa ........ 2009-12-07 23:29 +0000 [r233473-233612] David Vossel * /, main/utils.c: Merged revisions 233611 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 | dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines fixes incorrect logic in ast_uri_encode issue #16299 ........ * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009) | 15 lines Merged revisions 233471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) | 9 lines fixes missing Contact header angle brackets (closes issue #16298) Reported by: mgernoth Patches: reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ 2009-12-07 16:16 +0000 [r233396] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 | mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8 lines Do not reject SDP packets describing only non audio streams. (closes issue #16387) Reported by: zalex1953 Patches: media-level-c-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, zalex1953 ........ 2009-12-04 21:55 +0000 [r233281] David Vossel * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600 (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) | 7 lines clarify requirecalltoken option in iax.sample.conf (closes issue #16223) Reported by: bklang Patches: clarify-iax-requirecalltoken.patch uploaded by bklang (license 919) ........ ................ 2009-12-04 21:07 +0000 [r233240] Matthias Nick * pbx/pbx_config.c, /: Merged revisions 233093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 | mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines Parse global variables or expressions in hint extensions Parse global variables or expressions in hint extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166) Reported by: rmudgett Tested by: mnick, rmudgett ........ 2009-12-04 17:36 +0000 [r233165] David Vossel * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600 (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) | 6 lines document and rename strip_control() in app_voicemail (closes issue #16291) Reported by: wdoekes ........ ................ 2009-12-04 17:23 +0000 [r233130] Russell Bryant * main/channel.c, /: Merged revisions 233100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009) | 14 lines Merged revisions 233092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009) | 7 lines Only do frame payload check for HOLD frames. This code was added for helping to debug the source of invalid HOLD frames. However, a side effect of this is that it will incorrectly report errors for frames that have an integer payload. Make the check for this block specific to the HOLD frame case. ........ ................ 2009-12-04 15:57 +0000 [r233049] Matthias Nick * main/dsp.c, /: Merged revisions 233046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) | 17 lines Merged revisions 233014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | 11 lines Warning message gets displayed only once Added additional field 'int display_inband_dtmf_warning', which when set to '1' displays the warning ('Inband DTMF is not supported on codec %s. Use RFC2833'), and when set to '0' doesn't display the warning. Otherwise you would get hundreds of warnings every second. (closes issue #15769) Reported by: falves11 Patches: patch_15769_14.txt uploaded by mnick (license 874) Tested by: mnick, falves11 ........ ................ 2009-12-03 21:03 +0000 [r232866] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600 (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) | 8 lines Deprecate "cz" in favor of "cs". Also, change the use of language codes so that language registers as a prefix, rather than an exact match. (closes issue #16272) Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-12-03 15:14 +0000 [r232813] David Ruggles * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 | diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12 lines Prevent double closing of FDs by EIVR This caused a problem when asterisk was under heavy load and running both AGI and EIVR applications. EIVR would close an FD at which point it would be considered freed and be used by a new AGI instance the second close would then close the FD now in use by AGI. (closes issue #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec Review: https://reviewboard.asterisk.org/r/436/ ........ 2009-12-03 00:20 +0000 [r232675-232678] Tilghman Lesher * res/res_musiconhold.c: Oops, really remove it this time * res/res_musiconhold.c, /: Recorded merge of revisions 232660-232661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232660 | tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 Dec 2009) | 19 lines Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. * Classes are now tracked past removal from the core container, and module removal is actively prevented until all references are freed. * A hanging reference stored in the channel has been removed. This could have caused a mismatch and the music state not properly cleared, if two or more reloads occurred between MOH being stopped and MOH being restarted. * In certain circumstances, duplicate classes were possible. * A race existed at reload time between a process being killed and the thread responsible for reading from the related pipe respawning that process. * Several reference counts have also been corrected. At least one could have caused deleted classes to stick around forever, consuming resources. This originally manifested as MOH external processes that were not killed at reload time. (closes issue #16279, closes issue #16207) Reported by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman ........ r232661 | tilghman | 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove debugging line ........ 2009-12-02 23:28 +0000 [r232658] David Vossel * CHANGES, /, UPGRADE.txt: Merged revisions 232657 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232657 | dvossel | 2009-12-02 17:27:45 -0600 (Wed, 02 Dec 2009) | 6 lines update CHANGES and UPGRADE.txt for early media behavior change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by: miki ........ 2009-12-02 22:05 +0000 [r232579-232585] Jeff Peeler * main/manager.c, /: Merged revisions 232582 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009) | 14 lines Merged revisions 232581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009) | 7 lines Send ack (response/message) after receiving manager action userevent (closes issue #16264) Reported by: dimas Patches: event-ack.patch uploaded by dimas (license 88) ........ ................ * main/manager.c, /: Merged revisions 232576 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 | jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines Make manager response to "Action: events" finish with empty line (closes issue #16275) Reported by: vnovy Patches: manager.c.diff uploaded by vnovy (license 922) ........ 2009-12-02 17:11 +0000 [r232359] Joshua Colp * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) | 12 lines Merged revisions 232355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG. (closes issue #16239) Reported by: CGMChris ........ ................ 2009-12-02 17:01 +0000 [r232352] David Vossel * /, main/acl.c: Merged revisions 232351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009) | 12 lines Merged revisions 232350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009) | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in strace. (closes issue #16290) Reported by: wdoekes ........ ................ 2009-12-02 16:43 +0000 [r232348] Joshua Colp * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 | file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add support for handling the 415 Unsupported media type response like we do for a 488 Not acceptable here response. (closes issue #16186) Reported by: atis Patches: sip_t38_response_415.patch uploaded by atis (license 242) ........ 2009-12-02 15:43 +0000 [r232270] David Vossel * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600 (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) | 9 lines fixes segfault in func_groupcount closes issue #16337) Reported by: Parantido Patches: issue_16337.diff uploaded by dvossel (license 671) Tested by: Parantido, dvossel ........ ................ 2009-12-02 14:55 +0000 [r232232] Joshua Colp * /, channels/chan_sip.c: Merged revisions 232230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 | file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where a scheduled item ID would get retained on registrations in a certain scenario causing code to execute during reload that should not. (issue AST-263) ........ 2009-12-02 00:52 +0000 [r232094] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600 (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines Do not modify the gain settings on data calls. (The digital flag actually represents a data call.) (closes issue #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ 2009-12-01 23:40 +0000 [r232011-232015] Russell Bryant * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 | russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines Fix a build error on FreeBSD. ........ * /, main/file.c: Merged revisions 232008 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 232007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) | 2 lines Fix a warning pointed out by buildbot. ........ ................ 2009-12-01 22:03 +0000 [r231930] Jeff Peeler * main/channel.c, /: Merged revisions 231927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009) | 19 lines Merged revisions 231911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) | 12 lines Fix crash with invalid frame data The crash was happening as a result of a frame containing an invalid data pointer, but was set with data length of zero. The few times the issue was reproduced it _seemed_ that the frame was queued properly, that is the data pointer was set to NULL. I never could reproduce the crash so as a last resort the crash has been fixed, but a check in __ast_read has been added to give as much information about the source of problematic frames in the future. (closes issue #16058) Reported by: atis ........ ................ 2009-12-01 21:21 +0000 [r231870] David Vossel * main/pbx.c, /: Merged revisions 231867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 231853 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009) | 3 lines WaitExten m option with no parameters generates frame with zero datalen but non-null data ptr ........ ................ 2009-12-01 Leif Madsen * Release Asterisk 1.6.2.0-rc7 2009-12-01 15:48 +0000 [r231743] Matthew Nicholson * /, main/file.c: Merged revisions 231741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 231740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce() and return an error if no know formats are found. ........ ................ 2009-11-30 21:59 +0000 [r231695-231696] Kevin P. Fleming * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 231692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines Another round of UDPTL stack fixes/improvements: 1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL session, so that log/error/debug messages generated by the UDPTL stack can be 'connected' to the endpoint that caused them to be generated. 2) Improve comments (and process) of calculating the far end's maximum IFP size when redundancy mode is in use for error correction. 3) When an IFP larger than the calculated 'far max IFP' size is presented for writing, truncate it rather than putting in the buffer and allowing the buffer to overflow; this will cause the ends to retrain to a lower bit rate that produces IFPs of an appropriate size if possible, and if not possible, the FAX transfer will fail completely. In these cases, it is due to the one endpoint supplying a T38FaxMaxDatagram value that is improperly calculated and is too low to be of use; we have configuration options available to override this behavior. 4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer needed. ........ * pbx/pbx_config.c: Backport a tiny fix from trunk that makes GCC 4.4.x happier. 2009-11-30 21:36 +0000 [r231689] Matthew Nicholson * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c, main/app.c: Merged revisions 231688 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov 2009) | 15 lines Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ ................ 2009-11-30 20:47 +0000 [r231605] Joshua Colp * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 | file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines When receiving SDP that matches the version of the last one do not treat it as a fatal error. (closes issue #16238) Reported by: seandarcy ........ 2009-11-30 18:57 +0000 [r231505-231558] David Vossel * apps/app_queue.c, /: Merged revisions 231556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 | dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines app_queue crashes randomly, often during call-transfers This patch adds a ref to the queue_ent object's parent call_queue in queue_exec() so the call_queue won't be destroyed while the the queue_ent still holds a pointer to it. (closes issue 0015686) Tested by: dvossel, aragon ........ * main/rtp.c, /: Merged revisions 231491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009) | 17 lines Merged revisions 231441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009) | 11 lines fixes crash caused by RTP comfort noise payload greater than 24 bytes AST-2009-010 (closes issue #16242) Reported by: amorsen Patches: issue16242.diff uploaded by oej (license 306) Tested by: amorsen, oej, dvossel ........ ................ 2009-11-25 22:34 +0000 [r231302] Tilghman Lesher * main/channel.c, /: Merged revisions 231299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009) | 9 lines Merged revisions 231298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) | 2 lines After a frame duplication failure, unlock the channel before returning. ........ ................ 2009-11-25 15:48 +0000 [r231191] Matthew Nicholson * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 | mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4 lines Load pbx_lua with global symbols to allow linking with other lua libraries. Found by Maxim Litnitskiy. ........ 2009-11-24 20:36 +0000 [r231136] Tilghman Lesher * apps/app_queue.c, /: Merged revisions 231134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 | tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines Found a few places where queue refcounts were counted incorrectly. Also add debug statements. (closes issue #15982, closes issue #15984) Reported by: atis Patches: 20091111__issue15982.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ 2009-11-24 18:54 +0000 [r231098] Jeff Peeler * /, main/features.c: Merged revisions 231095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 | jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines Fix erroneous hangup extension execution ast_spawn_extension behaves differently from 1.4 in that hangups and extensions that do not exist do not return an error, whereas in 1.6 it does. This is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN flag gets set properly. (closes issue #16106) Reported by: ajohnson Tested by: ajohnson ........ 2009-11-23 15:48 +0000 [r230884] Joshua Colp * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 230881 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines Change fax detection in chan_sip so it behaves as one would expect. Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. ........ 2009-11-23 15:38 +0000 [r230796-230880] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov 2009) | 9 lines Merged revisions 230839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov 2009) | 1 line Correct fix for issue #16268... the reporter's original patch was very close to correct. ........ ................ * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov 2009) | 12 lines Merged revisions 230772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov 2009) | 5 lines Ensure that SDP parsing does not ignore the last line of the SDP. (closes issue #16268) Reported by: sgimeno ........ ................ 2009-11-20 22:36 +0000 [r230727] David Vossel * channels/chan_iax2.c, /: Merged revisions 230726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009) | 7 lines fixes iax2 show cache locking error, thanks alecdavis! (closes issue #16094) Reported by: alecdavis Patches: bug16094.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, dvossel ........ 2009-11-20 21:07 +0000 [r230629] Matthew Nicholson * /, main/features.c: Merged revisions 230628 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov 2009) | 15 lines Merged revisions 230627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR if it exists. This is necessary for the recordagentcalls option in chan_agent to store the recorded file name in the bridge CDR. (closes issue #14590) Reported by: msetim Patches: queue_agent_userfield.patch uploaded by Laureano (license 265) Tested by: Laureano, mnicholson ........ ................ 2009-11-20 17:31 +0000 [r230510-230585] David Vossel * main/audiohook.c, /, include/asterisk/audiohook.h: Merged revisions 230583 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 | dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines audiohook signal trigger on every status change (issue #14618) Review: https://reviewboard.asterisk.org/r/434/ ........ * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600 (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) | 10 lines fixes MixMonitor thread not exiting when StopMixMonitor is used (closes issue #16152) Reported by: AlexMS Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, AlexMS Review: https://reviewboard.asterisk.org/r/424/ ........ ................ 2009-11-16 16:41 +0000 [r230250-230384] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 230381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230381 | kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line Fix another buglet in T.38 session teardown at the end of FAX sessions. ........ * /, apps/app_fax.c: Merged revisions 230343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230343 | kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2 lines Ensure that only one end of a T.38 session initiates teardown at completion. ........ * channels/chan_iax2.c, /: Merged revisions 230247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230247 | kpfleming | 2009-11-15 11:23:02 -0600 (Sun, 15 Nov 2009) | 12 lines Merged revisions 230246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov 2009) | 6 lines Correct mistaken option name in error message. The configuration option for allowing hosts to make non-token-based calls is 'calltokenoptional', not 'calltokenignore'. (reported on asterisk-users) ........ ................ 2009-11-13 22:01 +0000 [r229969-230148] Joshua Colp * /, channels/chan_sip.c: Merged revisions 230145 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) | 15 lines Merged revisions 230144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8 lines Respect the maddr parameter in the Via header. (closes issue #14446) Reported by: frawd Patches: via_maddr.patch uploaded by frawd (license 610) Tested by: frawd ........ ................ * channels/chan_local.c, /: Merged revisions 230039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230039 | file | 2009-11-13 13:44:53 -0600 (Fri, 13 Nov 2009) | 16 lines Merged revisions 230038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9 lines Fix a crash caused by two threads thinking they should both free the chan_local private structure when only one should. (closes issue #15314) Reported by: sroberts Patches: Issue15314_Move_Nulling_owner.patch uploaded by davidw (license 780) Tested by: davidw, lottc ........ ................ * configs/extensions.conf.sample, /, apps/app_chanisavail.c: Merged revisions 229966 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) | 13 lines Merged revisions 229965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6 lines Document a limitation in the AVAILSTATUS variable from ChanIsAvail and provide a workaround for it that does not change existing behavior. (closes issue #14426) Reported by: macli ........ ................ 2009-11-13 Leif Madsen * Release Asterisk 1.6.2.0-rc6 2009-11-13 15:57 +0000 [r229915] Joshua Colp * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 | file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix T.38 negotiation regression introduced with the SDP parser changes. ........ 2009-11-12 23:31 +0000 [r229752] Jason Parker * channels/chan_oss.c, /: Merged revisions 229750 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 | qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix mute toggling on OSS channels. ........ 2009-11-12 16:47 +0000 [r229671] David Vossel * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600 (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009) | 6 lines fixes merging error, datastore was being freed in the wrong function. (closes issue #16219) Reported by: aragon ........ ................ 2009-11-11 20:49 +0000 [r229570] David Ruggles * doc/externalivr.txt: Merged revisions 229568 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229568 | diruggles | 2009-11-11 15:47:06 -0500 (Wed, 11 Nov 2009) | 9 lines Remove non-functional feature from ExternalIVR documentation Remove non-functional socket implementation of ExternalIVR from documentation (closes issue #16225) Reported by: thedavidfactor Patches: externalivr.txt.20091111.1542.patch uploaded by thedavidfactor (license 903) ........ 2009-11-11 19:56 +0000 [r229492-229502] David Brooks * main/pbx.c, /: Merged revisions 229499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009) | 15 lines Merged revisions 229498 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009) | 8 lines Solaris doesn't like NULL going to ast_log Solaris will crash if NULL is passed to ast_log. This simple patch simply uses S_OR to get around this. (closes issue #15392) Reported by: yrashk ........ ................ * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009) | 7 lines Flags not initialized in app_softhangup.c, causing undefined behavior Trivial patch [kobaz] to initialize an ast_flags = {0} (closes issue #16129) Reported by: kobaz ........ 2009-11-10 22:17 +0000 [r229366] Tilghman Lesher * main/pbx.c, /: Merged revisions 229361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009) | 19 lines Merged revisions 229360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) | 12 lines If two pattern classes start with the same digit and have the same number of characters, they will compare equal. The example given in the issue report is that of [234] and [246], which have these characteristics, yet they are clearly not equivalent. The code still uses these two characteristics, yet when the two scores compare equal, an additional check will be done to compare all characters within the class to verify equality. (closes issue #15421) Reported by: jsmith Patches: 20091109__issue15421__2.diff.txt uploaded by tilghman (license 14) Tested by: jsmith, thedavidfactor ........ ................ 2009-11-10 22:04 +0000 [r229359] David Ruggles * doc/externalivr.txt: Merged revisions 229356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov 2009) | 16 lines Merged revisions 229355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov 2009) | 9 lines Fix ExternalIVR Documentation Remove documentation for event that doesn't function (closes issue #16220) Reported by: thedavidfactor Patches: externalivr.txt.20091110.1622.patch uploaded by thedavidfactor (license 903) ........ ................ 2009-11-10 21:33 +0000 [r229354] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 | tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines When GOSUB is invoked within an AGI, it may not exit correctly. (closes issue #16216) Reported by: atis Patches: 20091110__atis_work.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ 2009-11-10 20:09 +0000 [r229285] Joshua Colp * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) | 15 lines Merged revisions 229281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8 lines Remove broken support for direct transcoding between G.726 RFC3551 and G.726 AAL2. On some systems the translation core would actually consider g726aal2 -> g726 -> signed linear to be a quicker path then g726aal2 -> signed linear which exposed this problem. (closes issue #15504) Reported by: globalnetinc ........ ................ 2009-11-10 17:52 +0000 [r229232] David Vossel * channels/chan_iax2.c, /: Merged revisions 229168 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600 (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009) | 9 lines don't crash on log message in solaris AST-2009-006 (closes issue #16206) Reported by: bklang Tested by: bklang ........ ................ 2009-11-10 17:39 +0000 [r229231] David Ruggles * doc/externalivr.txt: Merged revisions 229228 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov 2009) | 18 lines Merged revisions 229191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov 2009) | 11 lines Document ExternalIVR event tag collision ExternalIVR uses the D tag for two different event types. This documents that behavior and how to differentiate between the two cases. Also includes a minor spelling fix and clarification (closes issue #16211) Reported by: thedavidfactor Patches: externalivr.txt.20091109.1507.patch uploaded by thedavidfactor (license 903) ........ ................ 2009-11-10 15:47 +0000 [r229101] Matthew Nicholson * UPGRADE-1.6.txt, main/editline/makelist.in, UPGRADE.txt: Reset props that were accidently deleted in 229088. 2009-11-10 15:28 +0000 [r229094] David Vossel * res/res_config_pgsql.c, /: Merged revisions 229093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229093 | dvossel | 2009-11-10 09:27:45 -0600 (Tue, 10 Nov 2009) | 11 lines fixes pgsql double free of threadstorage A thread storage variable was being freed incorrectly, which resulted in a double free if two queries were made in the same thread. (closes issue #16011) Reported by: cristiandimache Patches: issue16011.diff uploaded by dvossel (license 671) ........ 2009-11-10 15:16 +0000 [r229088] Matthew Nicholson * UPGRADE-1.6.txt, main/editline/makelist.in, channels/chan_sip.c, UPGRADE.txt: Reverted revision 202007. (closes issue #16175) Reported by: paul-tg Tested by: paul-tg 2009-11-10 11:25 +0000 [r229078] Gavin Henry * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10 Nov 2009) | 20 lines Schema file additions * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses to allow standalone dialplan, account and mailbox entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir, - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap, - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed redundant IPaddr (there's already IPAddress) - Gives more configuration Flags for SIP-Users available (tested) - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses without extensibleObject (which really should be the last resort); gives also additional possibilities for LDAP-filter (closes issue #15874) Reported by: Medozas Patches: asterisk.ldap-schema.patch uploaded by Medozas (license 41) Tested by: Medozas, suretec ........ 2009-11-09 22:59 +0000 [r229017] Terry Wilson * channels/chan_local.c, /: Merged revisions 229015 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009) | 8 lines Don't crash when bridge->tech_pvt == NULL This is a similar solution to what is in place for chan_agent (closes issue #16003) Reported by: atis Tested by: twilson ........ 2009-11-09 22:17 +0000 [r229012] David Vossel * channels/chan_sip.c: fixes segfault when transferring a queue caller In sip_hangup we attempted to lock p->owner after we set it to NULL. Thanks to fhackenberger for reporting the issue and submitting a patch. (closes issue #15848) Reported by: fhackenberger Patches: digium_bug_0015848 uploaded by fhackenberger (license 592) Tested by: fhackenberger, lmadsen, TomS, shin-shoryuken, dvossel 2009-11-09 Leif Madsen * Release Asterisk 1.6.2.0-rc5 2009-11-09 15:40 +0000 [r228900] Leif Madsen * main/channel.c: Merged revisions 228897 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009) | 14 lines Merged revisions 228896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009) | 6 lines Update WARNING message. Update a WARNING message to give a suggested fix when encountered. (closes issue #16198) Reported by: atis Tested by: atis ........ ................ 2009-11-09 14:48 +0000 [r228859] Matthew Nicholson * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600 (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov 2009) | 8 lines Perform limited bounds checking when destroying ast_mutex_t structures to make sure we don't try to use negative indices. (closes issue #15588) Reported by: zerohalo Patches: 20090820__issue15588.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ 2009-11-06 22:37 +0000 [r228694] David Vossel * main/channel.c, /: Merged revisions 228693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009) | 16 lines Merged revisions 228692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009) | 9 lines fixes audiohook write crash occuring in chan_spy whisper mode. After writing to the audiohook list in ast_write(), frames were being freed incorrectly. Under certain conditions this resulted in a double free crash. (closes issue #16133) Reported by: wetwired ........ ................ 2009-11-06 20:26 +0000 [r228649] Matthew Nicholson * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600 (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov 2009) | 8 lines Properly handle '=' while decoding base64 messages and null terminate strings returned from BASE64_DECODE. (closes issue #15271) Reported by: chappell Patches: base64_fix.patch uploaded by chappell (license 8) Tested by: kobaz ........ ................ 2009-11-06 18:43 +0000 [r228551] Joshua Colp * /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) | 11 lines Merged revisions 228547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4 lines Don't overwrite caller ID name on a trunk with the configured fullname when using users.conf (issue ABE-1989) ........ ................ 2009-11-06 Leif Madsen * Release Asterisk 1.6.2.0-rc4 2009-11-06 17:54 +0000 [r228504] Joshua Colp * doc/tex/localchannel.tex, /: Merged revisions 228499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2 lines Fix the localchannel.tex file. ........ 2009-11-06 17:24 +0000 [r228421-228447] David Vossel * codecs/codec_ilbc.c, /: Merged revisions 228441 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 | dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines Fixes merging issue from 1.4, frame data is held in data.ptr in trunk ........ * codecs/codec_ilbc.c, /: Merged revisions 228420 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009) | 19 lines Merged revisions 228418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009) | 13 lines fixes segfault in iLBC For reasons not yet known, it appears possible for an ast_frame to have a datalen greater than zero while the actual data is NULL during Packet Loss Concealment. Most codecs don't support PLC so this doesn't affect them. This patch catches the malformed frame and prevents the crash from occuring. Additional efforts to determine why it is possible for a frame to look like this are still being investigated. (issue #16979) ........ ................ 2009-11-06 16:44 +0000 [r228413] Joshua Colp * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) | 14 lines Merged revisions 228409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 lines Fix a bug caused by a partially invalid frame (from the jitterbuffer) passing through the Asterisk core. (closes issue #15560) Reported by: jvandal (closes issue #15709) Reported by: covici ........ ................ 2009-11-06 15:43 +0000 [r228269-228340] David Vossel * /, main/astfd.c: Merged revisions 228339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009) | 12 lines Merged revisions 228338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) | 5 lines fixes crash in astfd.c (closes issue #15981) Reported by: slavon ........ ................ * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06 Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c (closes issue #15394) Reported by: boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790) Tested by: dbrooks, boroda ........ 2009-11-05 22:13 +0000 [r228198] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 | tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines Yet another error message in the dialplan (thanks, rmudgett/russellb) ........ 2009-11-05 21:27 +0000 [r228195] Jeff Peeler * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 | jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines Fix the fix for chanspy option o In 224178, I assumed the uploaded patch was correct as it had received positive feedback. The flags were being checked in the incorrect location. Upon testing the fix this time it was also found that the flags from the dialplan weren't being copied to the chanspy_translation_helper. (closes issue #16167) Reported by: marhbere ........ 2009-11-05 21:27 +0000 [r228194] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 | tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines MEETME_INFO should not return a literal error message to the dialplan. (closes issue #15450) Reported by: JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested by: JimVanM ........ 2009-11-05 19:42 +0000 [r228148] David Brooks * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600 (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r ........ ................ 2009-11-05 19:20 +0000 [r228093] Jason Parker * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228080 | qwell | 2009-11-05 13:16:29 -0600 (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) | 8 lines Fix crash on VPB exception when no hardware is present. (closes issue #14970) Reported by: tzafrir Patches: vpb_exception.diff uploaded by tzafrir (license 46) Tested by: markwaters ........ ................ 2009-11-05 17:14 +0000 [r228017] Tilghman Lesher * apps/app_externalivr.c, /: Merged revisions 228015 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009) | 4 lines Don't crash if no arguments are passed. (closes issue #16119) Reported by: thedavidfactor ........ 2009-11-04 23:53 +0000 [r227947] Jeff Peeler * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009) | 21 lines Merged revisions 227944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) | 14 lines Fix incorrect filename comparsion after monitor file change The logic to detect if a requested file is indeed a different file from the current file was incorrect. The main issue being confusion of the use of filename_base which was previously set without pathing information and then compared to another full path. Robust file comparison logic has been added to properly check if two files are the same even if symlinks are used. (closes issue #15313) Reported by: caspy Patches: 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325) but mostly tilghman's work ........ ................ 2009-11-04 21:09 +0000 [r227760-227831] Matthew Nicholson * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17 lines Merged revisions 227827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements. (closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ ........ ................ * channels/chan_sip.c: Modify the SDP parsing code to parse session and media level items separately. With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future. (closes issue #14994) Reported by: frawd 2009-11-04 19:28 +0000 [r227733-227748] Joshua Colp * /, static-http/prototype.js: Merged revisions 227739 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where it may be possible for someone to execute a cross-site AJAX request exploit. (AST-2009-009) ........ ................ * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where sending a REGISTER with a differing username in the From URI and Authorization header would reveal whether it was valid or not. (AST-2009-008) ........ ................ 2009-11-03 20:01 +0000 [r227375] Jason Parker * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) | 9 lines Fix some build issues on Solaris. (closes issue #14517) (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell ........ 2009-11-03 19:49 +0000 [r227364-227371] Leif Madsen * apps/app_controlplayback.c, /: Merged revisions 227368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03 Nov 2009) | 8 lines Change warning message to debug message. app_controlplayback outputs a warning, when in fact it is normal. (closes issue #16071) Reported by: atis Patches: controlplayback_warning.patch uploaded by atis (license 242) ........ * configs/extensions.conf.sample, /: Merged revisions 227361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines Additional fixes to the extensions.conf.sample file. Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville ........ 2009-11-03 18:15 +0000 [r227280] Richard Mudgett * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) | 4 lines Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. This is the relevant portion of asterisk/trunk -r226648 ........ 2009-11-03 17:14 +0000 [r227239] David Vossel * /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 | dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines user.conf entries in SIP were not having their peer type set. (closes issue #16120) Reported by: jsmith ........ 2009-11-03 15:40 +0000 [r227170] Joshua Colp * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) | 12 lines Merged revisions 227166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines Fix a bug where an RPID header could be generated with a blank username in the URI. (closes issue #15909) Reported by: kobaz ........ ................ 2009-11-03 15:25 +0000 [r227165] Leif Madsen * configs/extensions.conf.sample, /: Merged revisions 227162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03 Nov 2009) | 7 lines Update extensions.conf.sample file to fix incorrect extensions. (closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville ........ 2009-11-03 13:51 +0000 [r227156] Olle Johansson * Makefile, /, channels/chan_sip.c: Merged revisions 227091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines Use proper response code when violating Contact ACL's. https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) ........ ................ 2009-11-02 21:06 +0000 [r226978] David Brooks * channels/chan_sip.c: SIP channel name uniqueness SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ 2009-11-02 18:12 +0000 [r226893] Joshua Colp * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines Merged revisions 226889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) ........ ................ 2009-11-02 17:17 +0000 [r226815] Tilghman Lesher * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600 (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) | 8 lines Don't allow two separate instances of safe_asterisk when restarting from the init script. (closes issue #14562) Reported by: davidw Patches: Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14) Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780) Tested by: davidw ........ ................ 2009-10-29 18:18 +0000 [r226540] Joshua Colp * doc/tex/localchannel.tex, channels/chan_local.c, /: Merged revisions 226532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | 13 lines Merged revisions 226531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ ................ 2009-10-28 21:32 +0000 [r226486] Tzafrir Cohen * build_tools/get_documentation, /: remove empty awk pattern (//) Solaris 10 nawk doesn't like the empty pattern such as '//' for 'always'. Just remove that. No pattern at all always matches. Merged revisions 226453 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-10-28 20:13 +0000 [r226379-226385] Leif Madsen * configs/sip.conf.sample: Merged revisions 226384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ ................ * doc/tex/channelvariables.tex: Merged revisions 226378 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500 (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines Update CALLINGSUBADDR channel variable documentation. (closes issue #15734) Reported by: alecdavis Patches: channelvariables.tex.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ 2009-10-28 18:06 +0000 [r226170-226308] Tilghman Lesher * /, include/asterisk/linkedlists.h: Merged revisions 226305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500 (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines Fix documentation (pointed out by TheDavidFactor on #-dev) ........ ................ * main/manager.c, /: Merged revisions 226159 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009) | 14 lines Merged revisions 226138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines Manager output is not always NULL-terminated, so force a NULL at the end of the filestream. (closes issue #15495) Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded by tilghman (license 14) Tested by: pdf ........ ................ 2009-10-27 17:12 +0000 [r226101] Terry Wilson * res/res_http_post.c, /: Merged revisions 226099 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 | twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines Don't prepend the URI prefix to the post directory ........ 2009-10-27 00:16 +0000 [r226055] Tzafrir Cohen * /, configure, configure.ac: detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even if host_os is linux-gnueabi * When checking if we are Linux, check OSARCH rather than host_os The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is tested for the value of 'linux-gnu' in one or two places in the tree. This patch also fixes the check libcap to check for $OSARCH rather than $host_os . See also: http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4 Merged revisions 226018 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-10-26 19:42 +0000 [r225914] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 | jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines ACL check not present for verifying SIP INVITEs The ACL check in check_peer_ok was missing and has now been restored. The missing check allowed for calls to be made on prohibited networks where an ACL was defined in sip.conf and the allowguest option was set to off. See the AST security advisory below for more information. Merge code associated with AST-2009-007. (closes issue #16091) Reported by: thom4fun ........ 2009-10-26 15:56 +0000 [r225871] Kevin P. Fleming * apps/app_fax.c: Backport audio handling loop fixes from trunk version of app_fax. This backport resolves some issues handling audio frames during FAX processing, and ensures that the FAX application doesn't accidentally get notified of a T.38 switchover at the end of a successful FAX. (closes issue #16127) 2009-10-23 14:46 +0000 [r225651] David Vossel * /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 | dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines Fixes an iterator memory leak and uninitialized memory ........ 2009-10-23 14:08 +0000 [r225585] Kevin P. Fleming * Makefile, /: Merged revisions 225582 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct 2009) | 17 lines Merged revisions 225581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on every build. For some reason the menuselect.makeopts file was listed as PHONY in the Makefile, resulting in 'make' needing to rebuild it for every build. This then resulted in the embedded module rules being rebuilt on every build, which can be slow and is unnecessary. This patch fixes the problem by properly allowing 'make' to know when the menuselect.makeopts file needs to be rebuilt (defining the proper dependencies). ........ ................ 2009-10-22 22:24 +0000 [r225516] Leif Madsen * README, /: Merged revisions 225515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225515 | lmadsen | 2009-10-22 17:24:03 -0500 (Thu, 22 Oct 2009) | 8 lines Update README documentation. Update the README documentation to correctly describe which CLI command you should use when attempting to get help from the CLI. (closes issue #16064) Reported by: thedavidfactor Patches: readme.patch uploaded by thedavidfactor (license 903) ........ 2009-10-22 21:55 +0000 [r225489] David Vossel * apps/app_externalivr.c, include/asterisk/tcptls.h, main/tcptls.c, /, channels/chan_sip.c: Merged revisions 225445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ ........ 2009-10-22 21:54 +0000 [r225488] Leif Madsen * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions 225485 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009) | 19 lines Merged revisions 225484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines Clean valgrind output by suppressing false errors. Update valgrind.txt documentation and add valgrind.supp file in order to allow those who are creating valgrind output to have less false errors in the logfile. (closes issue #16007) Reported by: atis Patches: valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp uploaded by atis (license 242) Tested by: atis, amorsen ........ ................ 2009-10-22 17:14 +0000 [r225363] Tilghman Lesher * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h: Merged revisions 225360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ 2009-10-21 22:00 +0000 [r225035-225308] David Vossel * channels/chan_iax2.c, /: Merged revisions 225307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ ................ * configs/sip.conf.sample, channels/chan_iax2.c, configs/iax.conf.sample, /, channels/chan_sip.c: Merged revisions 225033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines Merged revisions 225032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ ................ 2009-10-20 22:11 +0000 [r224859] Tilghman Lesher * main/audiohook.c, funcs/func_speex.c, /: Merged revisions 224856 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ ................ 2009-10-20 17:50 +0000 [r224777] Joshua Colp * /, main/features.c: Merged revisions 224774 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ ................ 2009-10-20 00:00 +0000 [r224674] Kevin P. Fleming * main/rtp.c, /: Merged revisions 224671 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct 2009) | 14 lines Merged revisions 224670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ ................ 2009-10-19 19:54 +0000 [r224571] Joshua Colp * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines Merged revisions 224565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ ................ 2009-10-19 19:41 +0000 [r224563] Kevin P. Fleming * formats/format_siren14.c, /: Merged revisions 224562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224562 | kpfleming | 2009-10-19 14:40:26 -0500 (Mon, 19 Oct 2009) | 1 line Remove useless debugging message. ........ 2009-10-19 00:13 +0000 [r224447-224451] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) | 3 lines Allow ODBC storage to be queried with multiple mailboxes, and remove multiple goto's. This corrects an issue reported on the -users list. ........ * configs/res_odbc.conf.sample, /: Merged revisions 224446 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18 Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias for "autocommit", but instead controls the default disposition of uncommitted transactions. ........ 2009-10-17 01:58 +0000 [r224334] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ ................ 2009-10-16 20:58 +0000 [r224264] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500 (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ ................ 2009-10-15 15:58 +0000 [r224181] Jeff Peeler * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 | jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines Readd removed ability to allow listening to one side of the call in app_chanspy (Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html ........ 2009-10-12 23:55 +0000 [r223835] Jeff Peeler * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines Merged revisions 223804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ ................ 2009-10-12 21:01 +0000 [r223757] David Vossel * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options SWP-151 ........ 2009-10-12 14:37 +0000 [r223655] Kevin P. Fleming * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque ........ 2009-10-11 17:32 +0000 [r223490] Russell Bryant * main/autoservice.c, /: Merged revisions 223487 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009) | 17 lines Merged revisions 223485-223486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines Don't use data outside of its scope. The purpose of this code was to have a hangup frame put on the list of deferred frames. However, the code that read the hangup frame was outside of the scope of where the hangup frame was declared. ........ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines Remove some unnecessary code. ........ ................ 2009-10-09 23:12 +0000 [r223406] Jeff Peeler * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation of PRIREDIRECTIONREASON set by chan_sip. This commit is the simplest way to solve a problem that has already been solved in trunk with the "COLP/CONP and Redirecting party information into Asterisk" commit. In trunk the redirection reason is translated into a generic redirect reason. I would have had to do the same fix except chan_sip never reads PRIREDIRECTREASON. So both chan_dahdi and chan_h323 have been modified to interpret the one different redirect reason of "no-answer" properly and set the ISDN reason code 2 of "no reply". (closes issue #15033) Reported by: steinwej 2009-10-09 21:01 +0000 [r223333] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines Initiate T.38 switchover when acting as called party, regardless of FAX direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque ........ 2009-10-09 18:53 +0000 [r223286] Matthew Nicholson * main/channel.c, /: Merged revisions 223273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct 2009) | 14 lines Merged revisions 223225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING when originating calls. (closes issue #15104) Reported by: nblasgen Patches: manager-timeout1.diff uploaded by mnicholson (license 96) Tested by: nblasgen, mnicholson ........ ................ 2009-10-09 18:29 +0000 [r223257] Mark Michelson * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c ........ ................ 2009-10-09 17:55 +0000 [r223208] David Vossel * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines Merged revisions 223205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines fixes sip registration using authuser in user.conf (closes issue #14954) Reported by: tornblad Tested by: mmichelson, tornblad, dvossel ........ ................ 2009-10-09 17:27 +0000 [r223173] Matthew Nicholson * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct 2009) | 8 lines Don't close the sqlite database when reloading. Only close the database when unloading. (closes issue #15953) Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by frawd (license 610) Tested by: frawd ........ 2009-10-09 17:09 +0000 [r223089-223133] David Vossel * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 | dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines 'auth=' did not parse md5 secret correctly (closes issue #15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad ........ * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 | dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines p->peerauth is always empty in transmit_register() When using callbackextension or specifing the peer name in a registration string, the peer's specific auth settings set by the "auth=" strings within the peer definition are not used by the registration. Thanks to ebroad for reporting the issue and providing the patch. (closes issue #15955) Reported by: ebroad Patches: regauthfix.patch uploaded by ebroad (license 878) ........ 2009-10-08 20:00 +0000 [r222883] Russell Bryant * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c, /, main/file.c: Merged revisions 222880 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ ................ 2009-10-08 19:41 +0000 [r222874] David Vossel * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions 222873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 | dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines fixes an ast_netsock_list memory leak. ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ ........ 2009-10-08 16:51 +0000 [r222695-222802] Richard Mudgett * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500 (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) | 12 lines Fix memory leak if chan_misdn config parameter is repeated. Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 ........ ................ * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500 (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 ........ ................ 2009-10-07 18:06 +0000 [r222549] Jason Parker * /, configs/queues.conf.sample: Merged revisions 222548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct 2009) | 5 lines Remove 'keepstats' queue option from sample config, as it's no longer used. https://reviewboard.asterisk.org/r/115/ (closes issue #15820) Reported by: kshumard ........ 2009-10-07 18:00 +0000 [r222547] Sean Bright * funcs/func_strings.c: Fix merge error. 2009-10-07 17:45 +0000 [r222544] David Vossel * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009) | 14 lines Merged revisions 222542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) | 8 lines crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) ........ ................ 2009-10-06 23:59 +0000 [r222354-222466] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500 (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) | 8 lines Add missing unlock(s) in dahdi_read (two cases in trunk, and 1.6.2) (closes issue #15683) Reported by: alecdavis ........ ................ * channels/chan_dahdi.c: Fix potential crash when entire span request is received. The variable index used in this scenario for accessing the dahdi_pvts was wrong and was most likely copied from the several other places it is used correctly. (closes issue #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch uploaded by tsearle (license 373) * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009) | 9 lines Fix 222298 (crash during destruction of second channel when variable set with setvar). I mistakenly reasoned that setvar would be used on all channels. Since it can be set per channel, give each dahdi channel a copy of the variable. (related to #15899) ........ 2009-10-06 19:41 +0000 [r222311] Tilghman Lesher * cdr/cdr_pgsql.c, res/res_config_pgsql.c, /: Merged revisions 222309 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 | tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10 lines Change schema query to involve the use of an optional schema parameter. This change is done in such a way as to allow the driver to continue to function with older databases which don't have these features. (closes issue #16000) Reported by: jamicque Patches: 20091002__issue16000.diff.txt uploaded by tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ 2009-10-06 19:27 +0000 [r222304] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009) | 9 lines Fix crash during destruction of second channel when variable set with setvar. The setvar line in chan_dahdi.conf is shared among all the channels, so make sure to only free the resources only when the last channel is destroyed. (closes issue #15899) Reported by: tzafrir ........ 2009-10-06 19:22 +0000 [r222289] Tilghman Lesher * res/ael/pval.c, /: Merged revisions 222273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 | tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines When we call a gosub routine, the variables should be scoped to avoid contaminating the caller. This affected the ~~EXTEN~~ hack, where a subroutine might have changed the value before it was used in the caller. Patch by myself, tested by ebroad on #asterisk ........ 2009-10-06 Leif Madsen * Released Asterisk 1.6.2.0-rc3 2009-10-06 01:39 +0000 [r222113-222187] Kevin P. Fleming * channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c, channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c, res/res_clialiases.c, /, channels/chan_sip.c, funcs/func_dialgroup.c, include/asterisk/astobj2.h, res/res_phoneprov.c: Merged revisions 222176 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ ................ * configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c, configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 222110 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ 2009-10-02 17:35 +0000 [r222032] David Vossel * channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a memcpy. ........ ................ 2009-10-02 17:01 +0000 [r221923-221974] Tilghman Lesher * main/astobj2.c, /: Merged revisions 221971 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 221970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines Ensure the result of the hash function is positive. Negative array offsets suck. ........ ................ * /, main/logger.c: Merged revisions 221920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 | tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines Initialize a variable that we check immediately upon startup. (closes issue #15973) Reported by: atis ........ 2009-10-02 01:35 +0000 [r221879] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: Merged revisions 221844 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines Merged revisions 221769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines Occasionally losing use of B channels in chan_misdn. I have not been able to reproduce the problem of losing channels. However, I have seen in the code a reentrancy problem that might give these symptoms. The reentrancy patch does several things: 1) Guards B channel and B channel structure allocation. 2) Makes the B channel structure find routines more precise in locating records. 3) Never leave a B channel allocated if we received cause 44. The last item may cause temporary outgoing call problems, but they should clear when the line becomes idle. (closes issue #15490) Reported by: slutec18 Patches: issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) Reported by: FabienToune Patches: issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: FabienToune, rmudgett, slutec18 ........ ................ 2009-10-02 00:07 +0000 [r221744-221780] Tilghman Lesher * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions 221777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines Fix a bunch of off-by-one errors ........ ................ * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 | tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers. ........ 2009-10-01 19:35 +0000 [r221698] David Vossel * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 | dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines outbound tls connections were not defaulting to port 5061 (closes issue #15854) Reported by: dvossel Patches: sip_port_config_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel ........ 2009-10-01 16:57 +0000 [r221660] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 221554,221589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu, 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE. ................ r221589 | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines Use unsigned ints for portinuri flags. ........ ................ 2009-10-01 16:25 +0000 [r221622] Kevin P. Fleming * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 221592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines Remove ability to control T.38 FAX error correction from udptl.conf. chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. ........ 2009-09-30 23:07 +0000 [r221477-221485] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 | mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2 lines Cleaned up merge from r221432 ........ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 221432 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines Merged revisions 221360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ ................ 2009-09-30 21:46 +0000 [r221371-221472] Matthias Nick * apps/app_queue.c, /: Merged revisions 221436 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 | mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines Prevents from division by zero ........ * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged revisions 221368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) | 23 lines Merged revisions 221153,221157,221303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines check bounds - prevents for buffer overflow ........ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines changed the prototype definition of csv_quote ........ ................ 2009-09-30 19:15 +0000 [r221304] Terry Wilson * configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: Merged revisions 221266 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ 2009-09-30 16:57 +0000 [r221204] Tilghman Lesher * main/channel.c, /: Merged revisions 221201 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009) | 14 lines Merged revisions 221200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines Avoid a potential NULL dereference. (closes issue #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt uploaded by tilghman (license 14) Tested by: kobaz ........ ................ 2009-09-30 14:57 +0000 [r221089] Sean Bright * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a() option. We require box numbers, not names as the documentation implies. (issue #14740) Reported by: pj Patches: __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10) Tested by: seanbright, lmadsen ........ 2009-09-30 04:41 +0000 [r221027-221047] Tilghman Lesher * /, funcs/func_lock.c: Recorded merge of revisions 221044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29 Sep 2009) | 8 lines Allow locks to be inherited through a masquerade without causing starvation. (closes issue #14859) Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: atis, tilghman ........ * include/asterisk/smdi.h, include/asterisk/optional_api.h (removed), apps/app_voicemail.c, include/asterisk/agi.h, include/asterisk/monitor.h: Remove optional_api from 1.6.2 branch, since it is not currently working. This is a blocking issue for the 1.6.2 release. (closes issue #15914) Reported by: mbeckwell Branch: http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162 Tested by: mbeckwell * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines Merged revisions 220873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines Reduce CPU usage related to building a peer merely for devicestates. This fixes a 100% CPU problem in the SIP driver, found by profiling the driver while the problem was occurring. (closes issue #14309) Reported by: pkempgen Patches: 20090924__issue14309.diff.txt uploaded by tilghman (license 14) Tested by: pkempgen, vrban ........ ................ 2009-09-29 20:24 +0000 [r220905-220934] Matthew Nicholson * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked. (closes issue #15965) Reported by: atis Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96) Tested by: atis * /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep 2009) | 5 lines Fix options 'm' and 's'. They were swapped in the code. Also document the fact that app_confbridge does not automatically answer the channel. (closes issue #15964) Reported by: shrift ........ 2009-09-29 17:06 +0000 [r220836] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines Make deletion of temporary greetings work properly with IMAP_STORAGE When imapgreetings was set to yes, the message was being deleted but wasn't actually being expunged. When imapgreetings was set to no, the file based message was not being deleted at all. All good now! (closes issue #14949) Reported by: noahisaac Patches: vm_tempgreeting_removal.patch uploaded by noahisaac (license 748), modified by me ........ 2009-09-28 19:13 +0000 [r220725] Sean Bright * /, Makefile.rules: Merged revisions 220721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep 2009) | 10 lines Merged revisions 220717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler so we override any default optimization levels for a particular install. ........ ................ 2009-09-28 19:11 +0000 [r220722] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 | jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines Fix building of registration entry in build_peer when using callbackextension Check for remotesecret option was unintentionally always true, which therefore caused the secret option to never be used. Thanks to dvossel for pointing out the exact fix. (closes issue #15943) Reported by: tpsast ........ 2009-09-27 20:45 +0000 [r220632] Michiel van Baak * funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009) | 3 lines add name argument for the CALLERID dialplan function to the xml documentation. Pointed out to me on IRC by snuff-home. Thanks ........ 2009-09-26 15:12 +0000 [r220589] Tilghman Lesher * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009) | 2 lines Allow AES to compile, when OpenSSL is not present. ........ 2009-09-24 20:38 +0000 [r220369] David Vossel * main/tcptls.c, /: Merged revisions 220365 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 | dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines fixes tcptls_session memory leak caused by ref count error (closes issue #15939) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/375/ ........ 2009-09-24 19:42 +0000 [r220292] Tilghman Lesher * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged revisions 220289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13 lines Merged revisions 220288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines Implicitly sending a progress signal breaks some applications. Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html ........ ................ 2009-09-24 18:22 +0000 [r220103-220221] Sean Bright * Makefile, /: Merged revisions 220217 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep 2009) | 9 lines Merged revisions 220213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep 2009) | 1 line Resolve parallel build warnings. Reported by Klaus Darilion on the asterisk-dev mailing list. ........ ................ * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400 (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep 2009) | 2 lines Remove the remaining bashisms in the Makefile/mkpkgconfig ........ ................ 2009-09-24 08:43 +0000 [r220031] Michiel van Baak * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200 (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009) | 7 lines mkpkgconfig does not need bash so make it use /bin/sh This fixes building on all systems that don't have bash at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on #asterisk-dev ........ ................ 2009-09-24 07:45 +0000 [r219989] Tilghman Lesher * apps/app_directory.c, /: Merged revisions 219987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009) | 8 lines Fix two possible crashes, one only in 1.6.1 and one in 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg Patches: 20090914__issue15739.diff.txt uploaded by tilghman (license 14) 20090922__issue15739.diff.txt uploaded by tilghman (license 14) Tested by: DLNoah, jeffg ........ 2009-09-22 21:48 +0000 [r219821] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines When IMAP variables were changed during a reload, Voicemail did not use the new values. This change introduces a configuration version variable, which ensures that connections with the old values are not reused but are allowed to expire normally. (closes issue #15934) Reported by: viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by tilghman (license 14) Tested by: viniciusfontes ........ ................ 2009-09-21 17:01 +0000 [r219722] David Vossel * channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500 (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 Sep 2009) | 3 lines Reverting merge 219520. This change was not necessary. ........ ................ 2009-09-20 18:21 +0000 [r219669] Tilghman Lesher * /, main/file.c: Merged revisions 219654 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009) | 15 lines Merged revisions 219653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines Really stop the stream, when ast_closestream() is called. (closes issue #15129) Reported by: bmh Patches: 20090918__issue15129.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/372/ ........ ................ 2009-09-19 03:14 +0000 [r219590] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219587 | russell | 2009-09-18 21:59:52 -0500 (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009) | 6 lines Make sure the iax_pvt exists before dereferencing it. This fixes the latest crash posted on issue 15609. (issue #15609) ........ ................ 2009-09-18 23:21 +0000 [r219452-219521] David Vossel * channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500 (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines iax2 frame double free The iax frame's retrans sched id was written over right before iax2_frame_free was called. In iax2_frame_free that retrans id is used to delete the sched item. By writing over the retrans field before the sched item could be deleted, it was possible for a retransmit to occur on a freed frame. ........ ................ * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) | 20 lines Merged revisions 219450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines via-header branches not updated correctly on INVITE INVITE requests must always contain a new unique branch id. When a new branch id is created for an INVITE, the dialog's invite_branch variable must be updated so CANCEL requests use the correct branch id. (closes issue #15262) Reported by: maniax Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608) invite_new_branch_trunk.diff uploaded by dvossel (license 671) Tested by: maniax, dvossel ........ ................ 2009-09-18 13:57 +0000 [r219415] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009) | 6 lines Missing value setting line for maxsecs/maxmessage (closes issue #15696) Reported by: fhackenberger Patches: maxsecs.patch uploaded by fhackenberger (license 592) ........ 2009-09-17 22:38 +0000 [r219376] David Vossel * /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 | dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines fixes deadlock when performing directed pickup w Invite/replaces (closes issue #15340) Reported by: lmsteffan Patches: deadlock.patch uploaded by lmsteffan (license 779) Tested by: lmsteffan ........ 2009-09-17 22:37 +0000 [r219370] Joshua Colp * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep 2009) | 12 lines Merged revisions 219320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep 2009) | 6 lines Send a 100 Trying response when we detect a spiral. This was problematic during spiral tests at SIPit... along with some other things as well. ........ ................ 2009-09-17 22:06 +0000 [r219307] David Vossel * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009) | 27 lines Merged revisions 219303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines INVITE w/Replaces deadlock fix This patch cleans up the locking logic in chan_sip.c's handle_invite_replaces() function as well as making use of ast_do_masquerade() rather than forcing the masquerade on an ast_read(). The code had several redundant unlocks that would result in 'freed more times than we've locked!' errors. I cleaned these up as well as moving all the unlock logic to the end of the function. This patch should also resolve the issue people were having with the replacecall channel never being unlocked with one legged calls. (closes issue #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff uploaded by dvossel (license 671) Tested by: irroot, dvossel Review: https://reviewboard.asterisk.org/r/371/ ........ ................ 2009-09-17 19:58 +0000 [r219267] Joshua Colp * /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 | file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines Ensure no spaces exist before "refresher=" when doing the comparison. ........ 2009-09-17 Leif Madsen * Released Asterisk 1.6.2.0-rc2 2009-09-17 15:38 +0000 [r219194] Matthew Nicholson * main/channel.c, /, include/asterisk/cdr.h, include/asterisk/channel.h: Merged revisions 219139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ ................ 2009-09-16 23:52 +0000 [r219063] Tilghman Lesher * main/config.c, configs/extensions.conf.sample, /: Merged revisions 219061 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009) | 15 lines Merged revisions 219023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ ................ 2009-09-16 19:40 +0000 [r218938] David Brooks * main/pbx.c, /: Merged revisions 218868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009) | 20 lines Merged revisions 218867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines Fixes CID pattern matching behavior to mirror that of extension pattern matching. Pattern matching for extensions uses a type of scoring system, giving values for specificity to each character in the pattern. Unfortunately, this is done character by character, in order. This does lead to some less specific patterns being first in line for matching, but it will usually get the job done. This patch merely brings CID matching to the same level as extension matching. This patch does not attempt to tackle the problem shared by extension matching. (closes issue #14708) Reported by: klaus3000 ........ ................ 2009-09-16 19:29 +0000 [r218937] Mark Michelson * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 | mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12 lines Reverse order of args to fread. This way, we don't always write a null byte into byte 1 of the buffer (closes issue #15905) Reported by: ebroad Patches: freadfix.patch uploaded by ebroad (license 878) Tested by: ebroad ........ 2009-09-16 19:25 +0000 [r218934] Joshua Colp * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 | file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On TCP and TLS connections do not attempt to stop retransmission of the packet internally. This was preventing responses from being properly processed because the packet was not being found causing handle_response to return prematurely. ........ 2009-09-16 13:38 +0000 [r218802] Russell Bryant * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged revisions 218799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009) | 16 lines Merged revisions 218798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines Remove the IAXy firmware from Asterisk. The firmware can now be found on downloads.digium.com, where the rest of our binary downloads live. This was the last part of our Asterisk tarballs that was considered non-free by Debian. :-) (closes issue #15838) Reported by: paravoid ........ ................ 2009-09-15 22:46 +0000 [r218733] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500 (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines If the user enters the same password as before, don't signal an error when the change does nothing. (closes issue #15492) Reported by: cbbs70a Patches: 20090713__issue15492.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-09-15 19:24 +0000 [r218688] David Vossel * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 | dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines upward bound checking for port string to int conversion ........ 2009-09-15 16:18 +0000 [r218590] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep 2009) | 15 lines Merged revisions 218578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep 2009) | 8 lines Send request contact header field with response to registrer queries instead of the address of record. (closes issue #14438) Reported by: ravindrad Patches: regquerypatch uploaded by ravindrad (license 684) Tested by: ravindrad ........ ................ 2009-09-15 16:06 +0000 [r218582] Tilghman Lesher * apps/app_followme.c, /: Merged revisions 218579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009) | 16 lines Merged revisions 218577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines Ensure FollowMe sets language in channels it creates. Also, not in the original bug report, but related fields are accountcode and musicclass, and the inheritance of datastores. (closes issue #15372) Reported by: Romik Patches: 20090828__issue15372.diff.txt uploaded by tilghman (license 14) Tested by: cervajs ........ ................ 2009-09-15 15:59 +0000 [r218576] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 218430 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor. After talking to rmudgett about some of his recent iflist locking changes, it was determined that the only place that would destroy a channel without being explicitly to do so was in handle_init_event. The loop to walk the interface list has been modified to wait to destroy the channel until the dahdi_pvt of the channel to be destroyed is no longer needed. (closes issue #15378) Reported by: samy ........ ................ 2009-09-15 15:42 +0000 [r218507-218575] Mark Michelson * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 | mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4 lines Use a better method of ensuring null-termination of the buffer while reading the SDP when using TCP. ........ * /, channels/chan_sip.c: Merged revisions 218499,218504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue, 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53 -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP socket is null-terminated. ........ 2009-09-15 15:05 +0000 [r218503] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 218500 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep 2009) | 9 lines Merged revisions 218497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep 2009) | 1 line Use proper hostname for downloading sound files. ........ ................ 2009-09-14 19:49 +0000 [r218364] Tilghman Lesher * sounds/Makefile, apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 218361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines Don't say "Please try again" if we don't give the user another chance to try again. (issue #15055, SWP-129) Reported by: jthurman ........ ................ 2009-09-14 18:18 +0000 [r218300] Joshua Colp * /, main/features.c: Merged revisions 218295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 | file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do not attempt to add a parking extension if an error occurred while reading the configuration. ........ 2009-09-14 15:20 +0000 [r218238] Matthew Nicholson * /, apps/app_directed_pickup.c: Merged revisions 218224 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500 (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines Ensure we don't pickup ourselves when doing pickup by exten. (closes issue #15100) Reported by: lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan (license 779) ........ ................ 2009-09-13 22:12 +0000 [r218219] Tzafrir Cohen * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0 that annoys gcc This memset doesn't write beyond the end of the buffer. (tmpbuf has size of 4). Merged revisions 218184 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-09-13 05:59 +0000 [r218151] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 218150 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1 line get rid of mfcr2 monitor thread condition, is problematic ........ 2009-09-11 06:00 +0000 [r217926-218055] Tilghman Lesher * main/pbx.c, /: Merged revisions 218050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 | tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines Check the origination priority for more matches, not the current priority. Found by Pavel Troller on the -dev list. ........ * apps/app_queue.c, /: Merged revisions 217990 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009) | 10 lines Merged revisions 217989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines Don't ring another channel, if there's not enough time for a queue member to answer. (Fixes AST-228) ........ ................ * channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /, channels/chan_sip.c: Merged revisions 217916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 | tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines Make calltoken support work with realtime users and peers. ........ 2009-09-10 21:21 +0000 [r217821] David Vossel * channels/chan_iax2.c, /: Merged revisions 217807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500 (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call Token security patch inadvertently broke the use of encryption due to the reorganization of code in the socket_process() function. When encryption is used, an incoming full frame must first be decrypted before the information elements can be parsed. The security release mistakenly moved IE parsing before decryption in order to process the new Call Token IE. To resolve this, decryption of full frames is once again done before looking into the frame. This involves searching for an existing callno, checking the pvt to see if encryption is turned on, and decrypting the packet before the internal fields of the full frame are accessed. (closes issue #15834) Reported by: karesmakro Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, karesmakro Review: https://reviewboard.asterisk.org/r/355/ ........ ................ 2009-09-10 19:56 +0000 [r217739] mnick : * res/res_musiconhold.c, /: Merged revisions 217730 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) | 17 lines Sets the correct musicclass after an announcement (closes issue #15279) Reported by: mbeckwell Patches: patch.txt uploaded by mnick (license ) Tested by: mnick (closes issue #15832) Reported by: mbeckwell Patches: patch.txt uploaded by mnick (license 874) Tested by: mnick ........ 2009-09-10 18:40 +0000 [r217665] Olle Johansson * /, channels/chan_sip.c: Merged revisions 216805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 | oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Since it's possible to have more than 999 calls, I'm changing the call counter roof to something higher. ........ 2009-09-10 18:19 +0000 [r217647] Tilghman Lesher * res/res_config_odbc.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 217638 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 | tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines Verify support for wide ODBC character types before using them. (closes issue #15870) Reported by: nic_bellamy ........ 2009-09-10 15:14 +0000 [r217632] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 217524 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1 line ast_log replaced for ast_verbose in MFCR2 event notifications ........ 2009-09-10 12:09 +0000 [r217594] Olle Johansson * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 | oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines Include ActionID in all events that are responsed to AMI Action SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy Patches: manager_SIPshowregistry_actionid.patch uploaded by nic bellamy (license 299) ........ 2009-09-09 20:37 +0000 [r217519] Tzafrir Cohen * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc 4.4 has more strict rules for aliasing. It doesn't like a struct sockaddr_in pointer pointing to a struct sockaddr. So we make it a union. Merged revisions 217445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk 2009-09-09 10:58 +0000 [r217369] Olle Johansson * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 | oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not having any TLS session to write to is a serious XMIT_ERROR. ........ 2009-09-08 22:20 +0000 [r217299] Sean Bright * /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 | seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4 lines Fix compilation of app_meetme. Reported by ebroad in #asterisk-bugs ........ 2009-09-08 20:33 +0000 [r217217] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009) | 14 lines Merged revisions 217156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines When MOH is playing on the channel, announcements sent through the conference are not heard. (closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman ........ ................ 2009-09-08 16:39 +0000 [r217077] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 217074 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 | kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9 lines Ensure that the default autoconf CFLAGS are not used. A recent change to the configure script that allows the user to specify CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That problem is now corrected. ........ 2009-09-08 15:36 +0000 [r217036] Tilghman Lesher * /, res/res_limit.c: Merged revisions 217033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 | tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines Remove what appears to be an unnecessary define. (closes issue #15851) Reported by: tzafrir ........ 2009-09-08 14:27 +0000 [r216994] David Vossel * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines caller id number empty parse_uri was not being given the correct scheme's, as a result, uri parsing did not parse the username correctly. One of the side effects of this is an empty caller id. (closes issue #15839) Reported by: ebroad Patches: blank_cidv2.patch uploaded by ebroad (license 878) parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: ebroad, dvossel ........ 2009-09-07 16:43 +0000 [r216647-216845] Olle Johansson * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 | oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Make sure we reset global_exclude_static at channel reload ........ * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 | oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If there is no session timer in the INVITE, set it to default value (not unset minimum = -1) Patch by oej closes issue #15621 Reported by: fnordian Tested by: atis ........ * CHANGES, UPGRADE.txt: Add docs * configs/sip.conf.sample, apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c, apps/app_disa.c: Merged revisions 216438 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ 2009-09-04 19:42 +0000 [r216598] David Vossel * /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 | dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines sip peer matching by address only with TCP/TLS This patch removes the contact header matching logic and adds logic to match all tcp/tls connections by ip only Review: https://reviewboard.asterisk.org/r/354/ ........ 2009-09-04 19:32 +0000 [r216597] Sean Bright * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep 2009) | 1 line Use ast_free() instead of free(). ........ 2009-09-04 17:53 +0000 [r216550-216553] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009) | 2 lines Fix trunk breakage. ........ * UPGRADE-1.6.txt, main/pbx.c, /: Merged revisions 216547 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04 Sep 2009) | 3 lines Enable turning off the application delimiter warning with the 'dontwarn' option. Suggested on the -dev list, and implemented in an alternate way by me. ........ 2009-09-04 15:11 +0000 [r216469-216509] Michiel van Baak * /, main/utils.c: Merged revisions 216506 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009) | 9 lines Merged revisions 216435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD ........ ................ * /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009) | 2 lines make sure canlog is set so we can compile with DEBUG_THREADS enabled on OpenBSD ........ 2009-09-04 13:57 +0000 [r216267-216436] Russell Bryant * /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 | russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines Do not treat every SIP peer as if they were configured with insecure=port. There was a problem in the function responsible for doing peer matching by IP address and port number such that during the second pass for checking for a peer configured with insecure=port, it would end up treating every peer as if it had been configured that way. These changes fix the logic in the peer IP and port comparison callback to handle insecure=port checking properly. This problem was introduced when SIP peers were converted to astobj2. Many thanks to dvossel for noticing this while working on another peer matching issue. ........ * doc/IAX2-security.txt (added), /: Merged revisions 216264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216264 | russell | 2009-09-04 05:48:44 -0500 (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216263 | russell | 2009-09-04 05:48:00 -0500 (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines Add a plain text version of the IAX2 security document. ........ ................ ................ 2009-09-04 06:14 +0000 [r216225] Michiel van Baak * main/astobj2.c, /: Merged revisions 216222 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 | mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines make sure 'start' is always initialized. Makes asterisk compile with --enable-dev-mode ........ 2009-09-03 19:44 +0000 [r216014-216099] Russell Bryant * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009) | 16 lines Merged revisions 216085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216085 | russell | 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt. ........ ................ ................ * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216009 | russell | 2009-09-03 13:45:54 -0500 (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216008 | russell | 2009-09-03 13:44:58 -0500 (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines Add IAX2 security document related to AST-2009-006. ........ ................ ................ 2009-09-03 18:42 +0000 [r216007] David Vossel * channels/chan_iax2.c, channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample, include/asterisk/acl.h, channels/iax2-parser.h, /, include/asterisk/astobj2.h, channels/iax2.h, main/acl.c: Merged revisions 215955 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ 2009-09-03 14:21 +0000 [r215887-215929] Olle Johansson * /, channels/chan_sip.c: Merged revisions 215891 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 | oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add known internal IP address when autodomain=yes (closes issue #14573) Reported by: pj Patches: sip-internip-autodomain1.diff uploaded by mnicholson (license 96) modified by oej Tested by: pj ........ * main/rtp.c, channels/chan_sip.c: Fix bad reports in "sip show channelstats". Not directly mergeable in svn trunk, needs more tests, therefore committed directly to 1.6.2. (closes issue #15819) Reported by: klaus3000 Patches: asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded by klaus3000 (license 65) Tested by: klaus3000, oej 2009-09-03 06:02 +0000 [r215841] Michiel van Baak * doc/manager_1_1.txt, /: Merged revisions 215838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215838 | mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines Document that SIPshowpeer and SKINNYshowline now include the configured parkinglot in their response. Prodded by snuff-work on #asterisk-dev IRC channel ........ 2009-09-03 03:44 +0000 [r215802] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 215801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 | tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines Default the callback extension to "s". This is a regression. (closes issue #15764) Reported by: elguero Change-type: bugfix ........ 2009-09-03 00:34 +0000 [r215795] Terry Wilson * /, channels/chan_sip.c: Merged revisions 215758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009) | 25 lines Merged revisions 215682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines Re-send non-100 provisional responses to prevent cancellation From section 13.3.1.1 of RFC 3261: If the UAS desires an extended period of time to answer the INVITE, it will need to ask for an "extension" in order to prevent proxies from canceling the transaction. A proxy has the option of canceling a transaction when there is a gap of 3 minutes between responses in a transaction. To prevent cancellation, the UAS MUST send a non-100 provisional response at every minute, to handle the possibility of lost provisional responses. (closes issue #11157) Reported by: rjain Tested by: twilson Review: https://reviewboard.asterisk.org/r/315/ ........ ................ 2009-09-02 21:46 +0000 [r215683] David Vossel * /, channels/chan_sip.c: Merged revisions 215681 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 | dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines port string to int conversion using sscanf There are several instances where a port is parsed from a uri or some other source and converted to an int value using atoi(), if for some reason the port string is empty, then a standard port is used. This logic is used over and over, so I created a function to handle it in a safer way using sscanf(). ........ 2009-09-02 21:37 +0000 [r215647-215680] Michiel van Baak * /, channels/chan_sip.c, channels/chan_skinny.c: Merged revisions 215665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 | mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines add Parkinglot info to sip show peer and skinny show line If we had this from the start, debugging the 'parking not using configured parkinglot' bug would have been easier. ........ * /, main/features.c: Merged revisions 215622 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215622 | mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines - lock channel before looking for a channel variable - Init the parkings list member of struct parkinglot. Thanks Sean for the explanation why this should be here. ........ 2009-09-02 18:52 +0000 [r215569-215570] Tilghman Lesher * /, main/Makefile, main/app.c: Merged revisions 215567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215567 | tilghman | 2009-09-02 13:37:25 -0500 (Wed, 02 Sep 2009) | 9 lines Close up to the soft open file limit (same on Linux, but varies drastically on OS X). Also, a Makefile fix for Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches: 20090901__issue14542.diff.txt uploaded by tilghman (license 14) Tested by: jtodd, tilghman Change-type: bugfix ........ * /, channels/chan_sip.c: Merged revisions 215222 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 | tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines Fix register such that lines with a transport string, but without an authuser, parse correctly. (AST-228) ........ 2009-09-02 17:35 +0000 [r215523] David Vossel * /, channels/chan_sip.c: Merged revisions 215522 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 | dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines SIP uri parsing cleanup Now, the scheme passed to parse_uri can either be a single scheme, or a list of schemes ',' delimited. This gets rid of the whole problem of having to create two buffers and calling parse_uri twice to check for separate schemes. Review: https://reviewboard.asterisk.org/r/343/ ........ 2009-09-02 16:35 +0000 [r215512] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 215479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009) | 3 lines like in chan_sip's sip_new skinny should copy the configured parkinglot from a line to the newly created channel. This makes callparking honor the configured parkinglot for skinny lines as well. ........ 2009-09-02 16:09 +0000 [r215467] David Vossel * /, channels/chan_sip.c: Merged revisions 215466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 | dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines SIP support for keep-alive event keep-alive events are used by Sipura/Linksys for NAT keepalive. There currently don't appear to be any problems with NAT, but everytime a keep-alive event is received, Asterisk responds with a "489 Bad event". This error may indicate to a user that NAT problems exist just because this even is not supported. Now, rather than respond with an error, the packet is consumed and a "200 ok" is sent just to indicate we received the packet. (issue #15084) Patches: chan_sip.keepalive.v1.diff uploaded by IgorG (license 20) ........ 2009-09-02 16:07 +0000 [r215465] Michiel van Baak * /, channels/chan_sip.c: Merged revisions 215462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 | mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12 lines Honor configured parkinglot when parking and retrieving parked calls Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer into the newly created channel. (closes issue #15538) Reported by: gracedman Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7) With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call) Tested by: gracedman, mvanbaak ........ 2009-09-02 01:49 +0000 [r215376] Dwayne M. Hubbard * /, apps/app_softhangup.c: Merged revisions 215338 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500 (Tue, 01 Sep 2009) | 18 lines Merged revisions 215270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) | 12 lines Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names In general channel names are in the form Foo/Bar-Z, but the channel name could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the channel name at the last hyphen. (closes issue #15810) Reported by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard (license 733) ........ ................ 2009-09-01 20:00 +0000 [r215165] Kevin P. Fleming * main/frame.c, /: Merged revisions 215161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215161 | kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3 lines Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS frames are properly decoded. ........ 2009-08-31 16:22 +0000 [r214822-214960] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 214945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500 (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009) | 7 lines Also unlock the "other" channel, when returning, due to glare. (closes issue #15787) Reported by: tim_ringenbach Patches: chan_local.diff uploaded by tim ringenbach (license 540) Tested by: tim_ringenbach ........ ................ * Makefile, /: Merged revisions 214898 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214898 | tilghman | 2009-08-30 17:10:35 -0500 (Sun, 30 Aug 2009) | 2 lines Force Darwin on ppc platforms to compile with a target level that supports aliasing. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c: Merged revisions 214819 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 | tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines If lua is detected with the lua5.1 prefix (or not), adjust the include path accordingly. Based upon feedback to a release announcement on the -users list. See http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html ........ 2009-08-29 Leif Madsen * Asterisk 1.6.2.0-rc1 released. 2009-08-28 20:17 +0000 [r214707] Tilghman Lesher * main/channel.c, /: Merged revisions 214702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009) | 15 lines Merged revisions 214701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) | 8 lines Modify comment to be a bit more accurate. We have kept this comment around long enough, that it's pretty clear that we're keeping the code, because changing the code would require a pretty fundamental architectural shift. We've also taken criticism in some quarters, because it was believed that it was referring to the code being nasty. No, the code isn't nasty, just the operation itself is rather odd. Fixed for eternity (probably not). ........ ................ 2009-08-28 20:05 +0000 [r214700] Kevin P. Fleming * makeopts.in, Makefile, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 214696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 | kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9 lines Ensure that CFLAGS and/or LDFLAGS provided to configure script are preserved. Cross-compilation environments want to provide 'defaults' for compiler and linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the environment or as command-line arguments to the configure script. This patch modifies the configure script and Makefile to preserve these settings and ensure they are used in the build process. ........ 2009-08-28 18:43 +0000 [r214653] Mark Michelson * /, include/asterisk/sched.h: Merged revisions 214650 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug 2009) | 3 lines Fix some incorrect documentation of sched_thread functions. ........ 2009-08-27 21:49 +0000 [r214202-214521] Tilghman Lesher * autoconf/libcurl.m4 (added), /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 214518 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009) | 14 lines Merged revisions 214517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009) | 7 lines Use autoconf to detect libcurl, as this enables cross-compilation checks, something we didn't allow before. (closes issue #15714) Reported by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded by tilghman (license 14) Tested by: pprindeville ........ ................ * main/manager.c, /: Merged revisions 214514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214514 | tilghman | 2009-08-27 16:26:37 -0500 (Thu, 27 Aug 2009) | 7 lines Ensure that we check for the special value CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by: a_villacis Patches: asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch uploaded by a villacis (license 660) (Plus a few of my own, to catch the remaining places within manager.c where it could have been a problem) ........ * autoconf/ast_ext_lib.m4, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 214466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009) | 9 lines Merged revisions 214436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009) | 2 lines One more build system change, to make the descriptions look better, if we have better information. ........ ................ * autoconf/ast_ext_lib.m4, /, configure, include/asterisk/autoconfig.h.in: Merged revisions 214360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500 (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009) | 3 lines Make autoheader descriptions render correctly in our autoconfig.h file. (Figured out while working with issue #14906) ........ ................ * /, channels/chan_sip.c: Merged revisions 214199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 | tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue #15362) Reported by: klaus3000 Patches: chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license 65) ........ 2009-08-26 16:39 +0000 [r214196] David Vossel * main/channel.c, /: Merged revisions 214195 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009) | 25 lines Merged revisions 214194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) | 19 lines ast_write() ignores ast_audiohook_write() results In ast_write(), if a channel has a list of audiohooks, those lists are written to and the resulting frame is what ast_write() should continue with. The problem was the returned audiohook frame was not being handled at all, and the original frame passed into it did not contain the mixed audio, so essentially audio was being lost. One result of this was chan_spy's whisper mode no longer worked. To complicate the issue, frames passed into ast_write may either be a single frame, or a list of frames. So, as the list of frames is processed in the audiohook_write, the returned frames had to be added to a new list. (closes issue #15660) Reported by: corruptor Tested by: dvossel ........ ................ 2009-08-25 22:43 +0000 [r213903-214155] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 214152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 | tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines Not all versions of gnu-linux use glibc, which contains iconv. Some (especially embedded systems) don't have iconv at all. (closes issue #15169) Reported by: pprindeville ........ * /, main/say.c: Merged revisions 214071 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009) | 17 lines Merged revisions 214068-214069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009) | 6 lines Fix pronunciation of German dates. (closes issue #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded by Benjamin Kluck (license 803) ........ r214069 | tilghman | 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should always compile before committing... ........ ................ * /, pbx/pbx_dundi.c: Merged revisions 213975 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213975 | tilghman | 2009-08-25 01:51:12 -0500 (Tue, 25 Aug 2009) | 6 lines DUNDILOOKUP function in 1.6 should use comma delimiters. (closes issue #15322) Reported by: chappell Patches: dundilookup-0015322.patch uploaded by chappell (license 8) ........ * main/pbx.c, /: Merged revisions 213971 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009) | 14 lines Merged revisions 213970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009) | 7 lines Improve error message by informing user exactly which function is missing a parethesis. (closes issue #15242) Reported by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by loloski (license 68) ........ ................ * Makefile, /: Merged revisions 213904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213904 | tilghman | 2009-08-24 21:54:07 -0500 (Mon, 24 Aug 2009) | 6 lines The DTD should be installed in the same path as the rest of the XML documentation. (closes issue #15344) Reported by: tzafrir Patches: makefile_appdocs_dtd.diff uploaded by tzafrir (license 46) ........ * Makefile, /: Merged revisions 213900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213900 | tilghman | 2009-08-24 21:41:17 -0500 (Mon, 24 Aug 2009) | 11 lines Merged revisions 213899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009) | 4 lines Use the default runlevels for Debian derivatives, instead of making up our own. (closes issue #14730) Reported by: pkempgen ........ ................ 2009-08-24 16:49 +0000 [r213836] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 213833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009) | 14 lines Fix storage of greetings when using IMAP_STORAGE The store macro was not getting called preventing storage of IMAP greetings at all. This has been corrected along with fixing checking if the imapgreetings option is turned on to store the greeting in IMAP. Lastly, the attachment filename was incorrectly using the full path instead of just the basename, which was causing problems with retrieval of the greeting. (closes issue #14950) Reported by: noahisaac (closes issue #15729) Reported by: lmadsen ........ 2009-08-24 04:54 +0000 [r213791] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 213790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213790 | moy | 2009-08-24 00:46:28 -0400 (Mon, 24 Aug 2009) | 1 line improve handling of openr2_chan_disconnect_call API failure, unlikely, but happened on openr2 library bug ........ 2009-08-21 22:54 +0000 [r213739] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 213738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 | tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines Clarifying comments in sip_register, and removing a dead section ........ 2009-08-21 22:23 +0000 [r213721] David Vossel * /, channels/chan_sip.c: Merged revisions 213716 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 | dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines Register request line contains wrong address when user domain and register host differ (closes issue #15539) Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ 2009-08-21 21:44 +0000 [r213698] Kevin P. Fleming * apps/app_voicemail.c, /: Merged revisions 213697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug 2009) | 12 lines Ensure that realtime mailboxes properly report status on subscription. This patch modifies app_voicemail's response to mailbox status subscriptions (via the internal event system) to ensure that a subscription triggers an explicit poll of the mailbox, so the subscriber can get an immediate cached event with that status. Previously, the cache was only populated with the status of non-realtime mailboxes. (closes issue #15717) Reported by: natmlt ........ 2009-08-21 21:12 +0000 [r213636] David Vossel * /, channels/chan_sip.c: Merged revisions 213635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 | dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines fixes sip register parsing when user@domain is used (issue #15008) (issue #15672) ........ 2009-08-21 16:55 +0000 [r213563] Tilghman Lesher * include/asterisk.h, /: Merged revisions 213560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009) | 14 lines Merged revisions 213559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009) | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files. (closes issue #15698) Reported by: slavon Patches: 20090817__issue15698.diff.txt uploaded by tilghman (license 14) Tested by: slavon, tilghman ........ ................ 2009-08-21 16:06 +0000 [r213497] Jason Parker * /, configs/queues.conf.sample: Merged revisions 213494 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213494 | qwell | 2009-08-21 11:04:21 -0500 (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) | 5 lines Clarify queues.conf comments to specify that variables should be set in the dialplan. (closes issue #15755) Reported by: trendboy ........ ................ 2009-08-21 04:25 +0000 [r213475-213481] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 213454 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213454 | moy | 2009-08-21 00:09:26 -0400 (Fri, 21 Aug 2009) | 1 line increment the mfcr2 monitor count when clearing the call request ........ * channels/chan_dahdi.c, /: Merged revisions 213216 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213216 | moy | 2009-08-19 23:26:59 -0400 (Wed, 19 Aug 2009) | 1 line fixed bug caused by calling ast_request without calling ast_call on an R2 channel, ie, CHANISAVAIL ........ 2009-08-21 03:53 +0000 [r213453] Terry Wilson * main/loader.c, /: Merged revisions 213450 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 | twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines Make LOAD_ORDER actually work ........ 2009-08-20 21:50 +0000 [r213413] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 213404 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009) | 12 lines Fix greeting retrieval from IMAP Properly check for the current voicemail state and if it doesn't exist, create it. (closes issue #14597) Reported by: wtca Patches: 14597_v2.patch uploaded by mmichelson (license 60) Tested by: jpeeler ........ 2009-08-20 20:37 +0000 [r213350] Matthew Nicholson * /, main/features.c: Merged revisions 213327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 | mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7 lines Fix a crash by checking the proper pointer for validity before deferencing it. (closes issue #15751) Reported by: atis Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license 242) ........ 2009-08-19 22:41 +0000 [r213182] Jason Parker * main/alaw.c, main/ulaw.c, /: Merged revisions 213179 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) | 5 lines Fix compile when certain G711 menuselect options are enabled. (closes issue #15697) Reported by: slavon ........ 2009-08-19 21:25 +0000 [r213128] David Vossel * apps/app_mixmonitor.c, /: Merged revisions 213113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500 (Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009) | 8 lines Fixes memory leak caused by incorrectly freeing mixmonitor (closes issue #15699) Reported by: edantie Patches: mixmonitor.patch uploaded by edantie (license 862) ........ ................ 2009-08-19 21:22 +0000 [r213095-213117] Tilghman Lesher * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 213098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 | tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines Better parsing for the "register" line Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman ........ * /, channels/chan_sip.c: Merged revisions 213093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 | tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines If we have realtime caching enabled, 'sip reload' must purge users/peers, even if the config files haven't changed. (closes issue #12869) Reported by: bcnit Patches: 20090819__issue12869__2.diff.txt uploaded by tilghman (license 14) Tested by: lasko ........ 2009-08-19 15:35 +0000 [r213047] Russell Bryant * /, main/features.c: Merged revisions 213046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213046 | russell | 2009-08-19 10:32:18 -0500 (Wed, 19 Aug 2009) | 4 lines Don't blow up on a NULL cdr. Reported in #asterisk-dev. ........ 2009-08-18 20:34 +0000 [r212931-212944] Kevin P. Fleming * /: Merged revisions 212939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212939 | kpfleming | 2009-08-18 15:33:34 -0500 (Tue, 18 Aug 2009) | 1 line Remove some accidentally-committed properties. ........ * sounds/Makefile, doc/tex/asterisk.tex, CREDITS, /, UPGRADE-1.4.txt, sounds/sounds.xml, build_tools/prep_tarball: Merged revisions 212922 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212922 | kpfleming | 2009-08-18 15:29:37 -0500 (Tue, 18 Aug 2009) | 6 lines Convert this branch to Opsound music-on-hold. For more details: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ ........ 2009-08-18 19:28 +0000 [r212866] Tilghman Lesher * /, configs/extconfig.conf.sample: Merged revisions 212857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18 Aug 2009) | 4 lines Make the default extconfig.conf match entries with the sample res_mysql.conf. This eliminates a future source of possible confusion with the configuration of 1.6.1 and higher. ........ 2009-08-18 16:56 +0000 [r212769] Richard Mudgett * channels/misdn/isdn_lib.c, /: Merged revisions 212758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500 (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009) | 1 line Removed some deadwood and added some doxygen comments. ........ ................ 2009-08-18 16:41 +0000 [r212767] Sean Bright * main/manager.c, /: Merged revisions 212764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r212764 | seanbright | 2009-08-18 12:38:36 -0400 (Tue, 18 Aug 2009) | 18 lines Merged revisions 212763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug 2009) | 11 lines Delay the creation of temporary files until we have a valid manager command to handle. Without this patch, asterisk creates a temporary file before determining if the specified command is valid. If invalid, we weren't properly cleaning up the file. (closes issue #15730) Reported by: zmehmood Patches: M15730.diff uploaded by junky (license 177) Tested by: zmehmood ........ ................ 2009-08-17 20:01 +0000 [r212631] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 212627 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009) | 4 lines Check the return value of opendir(3), or we may crash. (closes issue #15720) Reported by: tobias_e ........ 2009-08-17 18:56 +0000 [r212580-212584] Sean Bright * /, channels/chan_agent.c: Merged revisions 212581 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug 2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in chan_agent. (closes issue #15668) Reported by: davidw ........ * main/logger.c: Merged revisions 212574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212574 | seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8 lines Correct the return value check for ast_safe_system. The logic here was reversed as ast_safe_system returns -1 on error and not on success. Fix suggested by reporter. (closes issue #15667) Reported by: loic ........ 2009-08-17 16:52 +0000 [r212509] Jeff Peeler * channels/misdn_config.c, /: Merged revisions 212506 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500 (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If more ports were specified than configured in misdn.conf a reload would crash asterisk. The problem was the unconfigured port was using data from the previously configured port. When the data for an unconfigured port was freed a crash would result from the double free. (closes issue #12113) Reported by: agupta Patches: bug12113.patch uploaded by jpeeler (license 325) ........ ................ 2009-08-17 15:51 +0000 [r212434] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 212431 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500 (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix uninitialized variable causing random MWI indications. (closes issue #15727) Reported by: doda Patches: dahdi_changes.patch uploaded by doda (license 853) ........ r212430 | rmudgett | 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix uninitialized variable. ........ ................ 2009-08-14 17:37 +0000 [r212250] Tilghman Lesher * funcs/func_curl.c, /: Merged revisions 212249 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212249 | tilghman | 2009-08-14 12:36:40 -0500 (Fri, 14 Aug 2009) | 2 lines Add SSL_VERIFYPEER, as requested on the -users list ........ 2009-08-13 15:47 +0000 [r212116] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 212113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 | kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3 lines Ensure that T38FaxVersion is put into outgoing SDP in the proper case. ........ 2009-08-13 13:56 +0000 [r212070] Joshua Colp * /, channels/chan_sip.c: Merged revisions 212067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 | file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines Check an actual populated variable when seeing if we need to do video or not. ........ 2009-08-13 11:47 +0000 [r212030] Gavin Henry * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 212027 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212027 | ghenry | 2009-08-13 12:37:12 +0100 (Thu, 13 Aug 2009) | 6 lines Fixed typo (closes issue #15710) Reported by: suretec ........ 2009-08-12 23:16 +0000 [r211951-211959] Matthew Nicholson * apps/app_queue.c, /: Merged revisions 211957 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug 2009) | 17 lines Merged revisions 211953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug 2009) | 10 lines This patch adds additional checking when generating queue log TRANSFER events. The additional checks prevent generation of false TRANSFER events in certain situations. (closes issue #14536) Reported by: aragon Patches: queue-log-xfer-fix1.diff uploaded by mnicholson (license 96) Tested by: aragon, mnicholson ........ ................ * /, channels/chan_sip.c: Merged revisions 211876 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 | mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11 lines Make asterisk handle 423 Interval Too Short messages better. This change uses separate values for the acceptable minimum expiry provided by the 423 error and the expiry value stored in the configuration file. Previously, the value pulled from the configuration file would be overwritten. (closes issue #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch uploaded by Nick (license 657) Tested by: mnicholson ........ 2009-08-12 16:21 +0000 [r211785] Gavin Henry * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 211767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211767 | ghenry | 2009-08-12 17:00:46 +0100 (Wed, 12 Aug 2009) | 33 lines Added three new attributes and applied a patch to res_config_ldap.c attributetype ( AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent' DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) and patch fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725) Reported by: macogeek Patches: fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license 863) Tested by: suretec ........ 2009-08-10 19:51 +0000 [r211580-211585] Tilghman Lesher * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500 (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10 Aug 2009) | 1 line Conversion specifiers, not format specifiers ........ ................ * apps/app_queue.c, apps/app_talkdetect.c, agi/eagi-sphinx-test.c, res/res_config_curl.c, channels/chan_usbradio.c, channels/chan_misdn.c, res/snmp/agent.c, apps/app_sms.c, apps/app_verbose.c, apps/app_stack.c, apps/app_mixmonitor.c, main/asterisk.c, main/dsp.c, main/timing.c, doc/CODING-GUIDELINES, funcs/func_speex.c, main/frame.c, utils/muted.c, apps/app_meetme.c, apps/app_alarmreceiver.c, cdr/cdr_pgsql.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c, main/enum.c, main/indications.c, res/res_config_sqlite.c, channels/misdn_config.c, utils/frame.c, main/cli.c, pbx/pbx_loopback.c, channels/chan_phone.c, funcs/func_enum.c, res/res_smdi.c, channels/chan_skinny.c, funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c, res/res_config_ldap.c, apps/app_adsiprog.c, funcs/func_dialplan.c, main/pbx.c, main/dnsmgr.c, funcs/func_sprintf.c, funcs/func_timeout.c, channels/chan_sip.c, apps/app_privacy.c, res/res_limit.c, apps/app_waitforsilence.c, codecs/codec_speex.c, agi/eagi-test.c, apps/app_morsecode.c, funcs/func_cut.c, channels/chan_oss.c, main/netsock.c, apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c, pbx/pbx_dundi.c, utils/extconf.c, pbx/pbx_config.c, apps/app_chanspy.c, res/res_odbc.c, apps/app_voicemail.c, apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, /, apps/app_record.c, main/utils.c, cdr/cdr_adaptive_odbc.c, res/res_http_post.c, main/config.c, res/ael/pval.c, main/cdr.c, main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c, main/manager.c, apps/app_setcallerid.c, apps/app_osplookup.c, main/features.c, main/http.c, channels/xpmr/xpmr.c, apps/app_rpt.c, channels/chan_mgcp.c, res/res_config_pgsql.c, channels/chan_agent.c, funcs/func_math.c, apps/app_waituntil.c, apps/app_disa.c, main/acl.c, apps/app_originate.c, channels/iax2-provision.c: AST-2009-005 2009-08-10 14:15 +0000 [r211350] Joshua Colp * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 | file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix retrieval of the port used for the video stream when adding SDP to a SIP message. (closes issue #15121) Reported by: jsmith ........ 2009-08-09 15:43 +0000 [r211235-211278] Tilghman Lesher * /, main/astfd.c: Merged revisions 211275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009) | 9 lines Merged revisions 211274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009) | 2 lines Small oops. Clear the flags which have been checked. ........ ................ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 | tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines Check for NULL frame, before dereferencing pointer. (closes issue #15617) Reported by: rain ........ 2009-08-07 20:18 +0000 [r211122] Russell Bryant * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009) | 11 lines Recorded merge of revisions 211112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009) | 4 lines Resolve a deadlock involving app_chanspy and masquerades. (ABE-1936) ........ ................ 2009-08-07 18:20 +0000 [r211051] Tilghman Lesher * apps/app_queue.c, /: Merged revisions 211040 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009) | 21 lines Merged revisions 211038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername. This is a partial revert of revision 82590, which was an attempted cleanup, but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended as a method by which component interfaces could be queried from the queue. Membername isn't useful here, because that field cannot be used to obtain further information about the member. See the documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various AMI commands which take a member argument for further justification. (closes issue #15664) Reported by: rain Patches: app_queue-queue_member_list.diff uploaded by rain (license 327) ........ ................ 2009-08-07 13:10 +0000 [r210995] Kevin P. Fleming * main/udptl.c, /: Merged revisions 210992 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 | kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13 lines Workaround broken T.38 endpoints that offer tiny MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as the maximum IFP size that should be sent to them, rather than the maximum packet payload size. If such an endpoint also requests UDPRedundancy as the error correction mode, we'll end up calculating a tiny maximum IFP size, so small as to be unusable. This patch sets a lower bound on what we'll consider the remote's maximum IFP size to be, assuming that endpoints that do this really can accept larger packets than they've offered to accept. (closes issue #15649) Reported by: dazza76 ........ 2009-08-06 21:47 +0000 [r210911-210917] Tilghman Lesher * main/channel.c, /: Merged revisions 210914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009) | 14 lines Merged revisions 210913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009) | 7 lines Because channel information can be accessed outside of the channel thread, we must lock the channel prior to modifying it. (closes issue #15397) Reported by: caspy Patches: 20090714__issue15397.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged revisions 210908 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines Allow Gosub to recognize quote delimiters without consuming them. (closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ ........ 2009-08-06 17:49 +0000 [r210820] Joshua Colp * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 | file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines Accept additional T.38 reinvites after an initial one has been handled. Discussion of this subject has yielded that it is not actually acceptable to change T.38 parameters after the initial reinvite but declining is harsh and can cause the fax to fail when it may be possible to allow it to continue. This patch changes things so that additional T.38 reinvites are accepted but parameter changes ignored. This gives the fax a fighting chance. (closes issue #15610) Reported by: huangtx2009 ........ 2009-08-05 20:43 +0000 [r210686] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500 (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) | 14 lines Dialplan starts execution before the channel setup is complete. * Issue 15655: For the case where dialing is complete for an incoming call, dahdi_new() was asked to start the PBX and then the code set more channel variables. If the dialplan hungup before these channel variables got set, asterisk would likely crash. * Fixed potential for overlap incoming call to erroneously set channel variables as global dialplan variables if the ast_channel structure failed to get allocated. * Added missing set of CALLINGSUBADDR in the dialing is complete case. (closes issue #15655) Reported by: alecdavis ........ ................ 2009-08-05 18:56 +0000 [r210565-210566] Leif Madsen * /: Merged revisions 210564 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) | 11 lines Update imapstorage.txt documentation. Updated the imapstorage.txt documentation to reflect that issues with c-client versions older than 2007 seem to cause crashing issues that are not seen with more recent versions. Documentation has been updated to reflect this. (closes issue #14496) Reported by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, dbrooks ........ ................ * doc/tex/imapstorage.tex: Merged revisions 210564 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) | 11 lines Update imapstorage.txt documentation. Updated the imapstorage.txt documentation to reflect that issues with c-client versions older than 2007 seem to cause crashing issues that are not seen with more recent versions. Documentation has been updated to reflect this. (closes issue #14496) Reported by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, dbrooks ........ ................ 2009-08-04 14:55 +0000 [r210191-210241] Kevin P. Fleming * Makefile, /: Merged revisions 210238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug 2009) | 16 lines Merged revisions 210237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug 2009) | 10 lines Eliminate spurious compiler warnings from system headers on *BSD platforms. Ensure that system headers located in /usr/local/include are actually treated as system headers by the compiler, and not as local headers which are subject to warnings from the -Wundef compiler option and others. (closes issue #15606) Reported by: mvanbaak ........ ................ * configs/sip.conf.sample, configs/skinny.conf.sample, main/rtp.c, channels/chan_mgcp.c, doc/chan_sip-perf-testing.txt, contrib/scripts/realtime_pgsql.sql, /, channels/chan_sip.c, channels/chan_skinny.c, configs/mgcp.conf.sample, doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt, configs/res_ldap.conf.sample: Merged revisions 210190 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines Rename 'canreinvite' option to 'directmedia', with backwards compatibility. It is clear from multiple mailing list, forum, wiki and other sorts of posts that users don't really understand the effects that the 'canreinvite' config option actually has, and that in some cases they think that setting it to 'no' will actually cause various other features (T.38, MOH, etc.) to not work properly, when in fact this is not the case. This patch changes the proper name of the option to what it should have been from the beginning ('directmedia'), but preserves backwards compatibility for existing configurations. ........ 2009-08-01 11:33 +0000 [r209837-209906] Russell Bryant * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209887 | russell | 2009-08-01 06:29:25 -0500 (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009) | 5 lines Resolve a valgrind warning about a read from uninitialized memory. (issue #15396) Reported by: aragon ........ ................ * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209839 | russell | 2009-08-01 06:02:07 -0500 (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) | 13 lines Modify how Playtones() is used in Milliwatt() to resolve gain issue. When Milliwatt() was changed internally to use Playtones() so that the proper tone was used, it introduced a drop in gain in the output signal. So, use the playtones API directly and specify a volume argument such that the output matches the gain of the original Milliwatt() code. (closes issue #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff uploaded by russell (license 2) Tested by: rue_mohr ........ ................ * /, main/event.c: Merged revisions 209835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 | russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines Fix ast_event_queue_and_cache() to actually do the cache() part. (closes issue #15624) Reported by: ffossard Tested by: russell ........ 2009-08-01 01:34 +0000 [r209816] Kevin P. Fleming * pbx/pbx_config.c, channels/misdn/isdn_lib.c, utils/frame.c, main/pbx.c, /, main/Makefile, channels/misdn/ie.c: Merged revisions 209760-209761 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines Minor changes inspired by testing with latest GCC. The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. ........ ................ r209761 | kpfleming | 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert accidental Makefile change. ................ 2009-07-31 22:01 +0000 [r209715] Russell Bryant * /, main/event.c: Merged revisions 209711 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 | russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines Fix some places where ast_event_type was used instead of ast_event_ie_type. ........ 2009-07-30 18:51 +0000 [r209594] David Brooks * channels/chan_console.c, include/asterisk/abstract_jb.h, apps/app_forkcdr.c, channels/chan_dahdi.c, contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, codecs/lpc10/pitsyn.c: Merged revisions 209554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ 2009-07-30 14:40 +0000 [r209518] Mark Michelson * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 | mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8 lines Fix a crash that can result if text codecs are allowed but textsupport is disabled. (closes issue #15596) Reported by: fabled Patches: sip-red.patch uploaded by fabled (license 448) ........ 2009-07-28 Leif Madsen * Release Asterisk 1.6.2.0-beta4 2009-07-28 00:19 +0000 [r209328] Tilghman Lesher * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009) | 9 lines Merged revisions 209315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) | 2 lines Publish French extra sounds ........ ................ 2009-07-27 21:44 +0000 [r209265-209282] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 | kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7 lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages, clean up some looping logic, and correct an improper use of ast_free() for freeing an ast_frame. ........ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 | kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10 lines Make T.38 switchover in ReceiveFAX synchronous. In receive mode, if the channel that ReceiveFAX is running on supports T.38, we should *always* attempt to switch T.38, rather than listening for an incoming CNG tone and only triggering on that. The channel may be using a low-bitrate codec that distorts the CNG tone, the sending FAX endpoint may not send CNG at all, or there could be a variety of other reasons that we don't detect it, but in all those cases if T.38 is available we certainly want to use it. ........ 2009-07-27 20:58 +0000 [r209238] Mark Michelson * main/rtp.c, /: Merged revisions 209235 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 | mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5 lines Gracefully handle malformed RTP text packets. AST-2009-004 ........ 2009-07-27 20:33 +0000 [r209234] David Brooks * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /, include/asterisk/module.h, main/features.c, res/res_agi.c: Merged revisions 209098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ 2009-07-27 20:23 +0000 [r209135-209222] Mark Michelson * res/res_musiconhold.c, /: Merged revisions 209197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul 2009) | 9 lines Honor channel's music class when using realtime music on hold. (closes issue #15051) Reported by: alexh Patches: 15051.patch uploaded by mmichelson (license 60) Tested by: alexh ........ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions 209132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) | 24 lines Merged revisions 209131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ ................ 2009-07-27 16:07 +0000 [r209063] David Brooks * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing typos "recieved" with "received". From issue #15360, forgot to apply to trunk and other branches. 2009-07-27 15:40 +0000 [r209059] Kevin P. Fleming * Makefile, /: Merged revisions 209056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 | kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10 lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes. During the recent Makefile improvements I made, it seemed the 'make' was automatically carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so I removed the explict export of them. However, there are some circumstances where make does this, and some where it does not, so I've brought them back to ensure they are always exported. I also removed an extraneous double setting of _ASTLDFLAGS on *BSD platforms. ........ 2009-07-27 01:23 +0000 [r208927] Jeff Peeler * channels/chan_iax2.c, /, main/translate.c: Merged revisions 208924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) | 9 lines Merged revisions 208923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines Fix logic errors from 208746 ........ ................ 2009-07-26 14:07 +0000 [r208889] Michiel van Baak * contrib/scripts/install_prereq, /: Merged revisions 208886 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26 Jul 2009) | 2 lines add OpenBSD to the install_prereq script ........ 2009-07-25 12:31 +0000 [r208816-208853] Michiel van Baak * contrib/scripts/install_prereq, /: Merged revisions 208848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25 Jul 2009) | 2 lines libxml2-dev is needed as well by default. ........ * main/cli.c, /, configs/cli_aliases.conf.sample: Merged revisions 208813 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 | mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10 lines add default alias reload to run module reload. Requiring 'module reload' to reload everything, including core etc makes russell very unhappy. The default configuration already loads the 'friendly' aliases template. Added 'reload=module reload' to that template. Also removed the comment in main/cli.c that reload should come back. ........ 2009-07-25 06:26 +0000 [r208755] Jeff Peeler * channels/chan_iax2.c, /, channels/chan_skinny.c, main/translate.c: Merged revisions 208749 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) | 13 lines Merged revisions 208746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. ........ ................ 2009-07-24 21:13 +0000 [r208695-208710] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 208709 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208709 | russell | 2009-07-24 16:12:43 -0500 (Fri, 24 Jul 2009) | 2 lines Remove trailing whitespace. ........ * main/cli.c, /: Merged revisions 208706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208706 | russell | 2009-07-24 15:54:37 -0500 (Fri, 24 Jul 2009) | 6 lines Note that "reload" needs to be added back. I keep getting annoyed at having to type "module reload" to reload everything, so I'm adding a note that we need to add "reload" back. "module reload" doesn't really make sense as the command to reload everything, including the core. ........ * main/cli.c, /: Merged revisions 208693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208693 | russell | 2009-07-24 15:25:23 -0500 (Fri, 24 Jul 2009) | 2 lines Don't log a warning for something that does not affect operation. ........ 2009-07-24 19:42 +0000 [r208664] Mark Michelson * /: Fixing trunk-blocked property. 2009-07-24 18:56 +0000 [r208596] Russell Bryant * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) | 14 lines Merged revisions 208592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines Do not log an ERROR if autoservice_stop() returns -1. This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) ........ ................ 2009-07-24 18:32 +0000 [r208591] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16 lines Merged revisions 208587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines Only send a BYE when hanging up a channel that is up. For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac ........ ................ 2009-07-24 15:06 +0000 [r208551] Kevin P. Fleming * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 208548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ 2009-07-24 14:39 +0000 [r208545] Michiel van Baak * contrib/scripts/install_prereq, /: Merged revisions 208542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24 Jul 2009) | 13 lines use aptitude for debian based systems The function to check wether we need to install packages was using dpkg-query which was gives wrong output on Debian 5 Also, the apt-get has been replaced with aptitude because aptitude is now the preferred way to handle packages on Debian (closes issue #15570) Reported by: mvanbaak Patches: 2009072400_installprereq-aptitude.diff uploaded by mvanbaak (license 7) ........ 2009-07-23 22:31 +0000 [r208501] Kevin P. Fleming * include/asterisk/frame.h, main/rtp.c, main/channel.c, main/udptl.c, main/frame.c, /, channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h: Merged revisions 208464 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ 2009-07-23 19:36 +0000 [r208391] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul 2009) | 24 lines Merged revisions 208386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) | 17 lines Fix a problem where a 491 response could be sent out of dialog. This generalizes the fix for issue 13849. The initial fix corrected the problem that Asterisk would reply with a 491 if a reinvite were received from an endpoint and we had not yet received an ACK from that endpoint for the initial INVITE it had sent us. This expansion also allows Asterisk to appropriately handle an INVITE with authorization credentials if Asterisk had not received an ACK from the previous transaction in which Asterisk had responded to an unauthorized INVITE with a 407. (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ ................ 2009-07-23 19:25 +0000 [r208387] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500 (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) | 6 lines Only set the priindication setting when not performing a reload (closes issue #14696) Reported by: fdecher ........ ................ 2009-07-23 16:30 +0000 [r208266-208320] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul 2009) | 9 lines Merged revisions 208312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) | 3 lines Remove inaccurate XXX comment. ........ ................ * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul 2009) | 15 lines Merged revisions 208262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) | 8 lines Properly handle 183 responses which do not contain an SDP. (closes issue #15442) Reported by: ffloimair Patches: 15442.patch uploaded by mmichelson (license 60) Tested by: tkarl, ffloimair ........ ................ 2009-07-22 21:46 +0000 [r208116] Jason Parker * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 | qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines Restore an int declaration on PPC platforms. This x is one crafty little bugger... It was used for 2 different things (one of which was only done on PPC) in 1.4. One of the uses were removed in trunk, and with it went the declaration. (closes issue #14038) Reported by: ffloimair ........ 2009-07-22 16:52 +0000 [r207949-208053] Tilghman Lesher * /, res/res_realtime.c: Merged revisions 208052 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208052 | tilghman | 2009-07-22 11:49:42 -0500 (Wed, 22 Jul 2009) | 7 lines Clarify documentation on 'realtime update2' to show more than one condition. (closes issue #15357) Reported by: snuffy Patches: bug_fix_doc_update2.diff uploaded by snuffy (license 35) (slightly modified by me) ........ * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500 (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) | 8 lines Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional). This change makes URIENCODE and QUOTE behave similarly, since the documentation states that the argument is not optional, for both. (closes issue #15439) Reported by: pkempgen Patches: 20090706__issue15439.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-07-21 22:23 +0000 [r207930] Russell Bryant * doc/CODING-GUIDELINES, /: Merged revisions 207925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207925 | russell | 2009-07-21 17:22:18 -0500 (Tue, 21 Jul 2009) | 4 lines Note that we use tabs instead of spaces for indentation. I'm surprised this was never actually in here... ........ 2009-07-21 20:30 +0000 [r207785-207862] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500 (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines Wait for wink before dialing when using E&M wink signaling There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simply add SIG_EMWINK to the list. (closes issue #14434) Reported by: araasch ........ ................ * channels/chan_dahdi.c: Revert r207638, this approach could potentially block for an unacceptable amount of time. 2009-07-21 14:32 +0000 [r207727] Mark Michelson * main/manager.c, /: Merged revisions 207723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul 2009) | 11 lines Merged revisions 207714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) | 5 lines Document default timeout for AMI originations. AST-224 ........ ................ 2009-07-21 13:56 +0000 [r207685] Kevin P. Fleming * channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, funcs/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules, pbx/Makefile, res/Makefile: Merged revisions 207680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either* in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. ........ ................ 2009-07-21 04:51 +0000 [r207638] Jeff Peeler * channels/chan_dahdi.c: Wait for wink before dialing when using E&M wink signaling This patch adds a new dahdi_wait function to specifically wait for the wink event. If the wink is not eventually received the channel is hung up. (closes issue #14434) Reported by: araasch Patches: emwinkmod uploaded by araasch (license 693) 2009-07-20 22:14 +0000 [r207523] Mark Michelson * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) | 39 lines Merged revisions 207423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines Answer video SDP offers properly when videosupport is not enabled. Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch *only* helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 ........ ................ 2009-07-20 16:41 +0000 [r207364] Russell Bryant * main/channel.c, /: Merged revisions 207361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009) | 16 lines Merged revisions 207360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines Only do the chan->fdno check in ast_read() in a developer build. I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller ........ ................ 2009-07-18 04:19 +0000 [r207327] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 207317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17 Jul 2009) | 3 lines Flag field in wrong position. Reported by "Hoggins!" on asterisk-dev list. ........ 2009-07-18 03:50 +0000 [r207315] Richard Mudgett * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Merged revisions 145293,158010 from https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. These changes are already on trunk. ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. ................ r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines Merged revision 157977 from https://origsvn.digium.com/svn/asterisk/team/group/issue8824 ........ Fixes JIRA ABE-1726 The dial extension could be empty if you are using MISDN_KEYPAD to control ISDN provider features. ................ 2009-07-17 22:31 +0000 [r207226-207257] Tilghman Lesher * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17 Jul 2009) | 2 lines Add flag here, too (as requested by jsmith) ........ * /, doc/tex/odbcstorage.tex, UPGRADE.txt: Merged revisions 207224 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17 Jul 2009) | 2 lines Document the "flag" field in the voicemessages table. ........ 2009-07-17 19:40 +0000 [r207104-207159] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500 (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines Fix format specifier to print out an unsigned long long. Yep, it's even ifdefed out code. But it made it to the RR list... (closes issue #14726) Reported by: lmadsen ........ ................ * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) | 2 lines Update some missing allowed options for overlapdial ........ 2009-07-17 17:52 +0000 [r206869-207030] David Vossel * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 | dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines sip option flags handled incorrectly (closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima ........ * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines Merged revisions 206938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ ................ * /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009) | 6 lines TIMEOUT(absolute) returned negative value. (closes issue #15513) Reported by: ys ........ * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ ................ * /, main/callerid.c: Merged revisions 206868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) | 14 lines Merged revisions 206867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines avoid segfault caused by user error If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. ........ ................ 2009-07-16 16:53 +0000 [r206811] Tilghman Lesher * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500 (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517) Reported by: adomjan Patches: func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487) ........ ................ 2009-07-15 22:04 +0000 [r206770] David Vossel * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 | dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines Session timer were not activated if Supported header field in INVITE had both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license ........ 2009-07-15 21:50 +0000 [r206765] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: Merged revisions 206707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) | 33 lines Merged revisions 206706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... ................ ................ 2009-07-15 20:20 +0000 [r206703] David Vossel * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 | dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines callerid(num) is wrong when username is missing A domain only sip uri would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor ........ 2009-07-15 16:04 +0000 [r206639] Sean Bright * codecs/codec_dahdi.c, /: Merged revisions 206636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400 (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we are asking for it. ........ ................ 2009-07-14 20:26 +0000 [r206598] Tilghman Lesher * /, apps/app_meetme.c, contrib/scripts/meetme.sql: Merged revisions 206567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines Document all meetme realtime fields, and in the process, make some field lengths more consistent. (closes issue #15493) Reported by: lasko Patches: meetme.diff uploaded by lasko (license 833) ........ 2009-07-14 19:49 +0000 [r206565] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500 (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ ................ 2009-07-14 14:59 +0000 [r206389] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 206386 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206386 | russell | 2009-07-14 09:51:44 -0500 (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ ................ ................ 2009-07-14 01:59 +0000 [r206373] Richard Mudgett * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 206341 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009) | 11 lines Merged revisions 206284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ ................ 2009-07-13 23:27 +0000 [r206281] David Vossel * /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 | dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines dns lookup of peername rather than peer's host in transmit_register() (closes issue #15052) Reported by: fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by fsantulli (license 818) Tested by: fsantulli ........ 2009-07-13 16:24 +0000 [r206187] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009) | 2 lines Remove reference to non-existent help file ........ 2009-07-10 21:46 +0000 [r205986] David Vossel * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 | dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines SIP register not using peer's outbound proxy If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ ........ 2009-07-10 18:45 +0000 [r205942] Kevin P. Fleming * main/udptl.c, /: Merged revisions 205939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 | kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line Update comments about the level of T.38 support in Asterisk. ........ 2009-07-10 17:54 +0000 [r205882] Mark Michelson * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) | 30 lines Merged revisions 205877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ ................ 2009-07-10 16:47 +0000 [r205841] David Vossel * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines Merged revisions 205804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ ................ 2009-07-10 16:01 +0000 [r205781] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 205780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205780 | kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11 lines Eliminate extraneous LOG_DEBUG messages generated by app_fax. The transmit_audio() and transmit_t38() functions in app_fax have processing loops that are supposed to wait for frames to arrive on the channel and then handle them, but they also have short timeouts so that the loops can have watchdog timers and do other required processing. This commit changes the loops to not actually call ast_read() and attempt to process the returned frame unless a frame actually arrived, eliminating hundreds of LOG_DEBUG messages and slightly improving performance. ........ 2009-07-10 15:58 +0000 [r205779] Mark Michelson * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ 2009-07-10 15:36 +0000 [r205773] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines Fix some remaining T.38 negotiation problems in app_fax. Revision 205696 did not quite fix all the issues with the T.38 negotiation changes and app_fax; this patch corrects them, along with a couple of other minor issues. (closes issue #15480) Reported by: dimas Patches: test2-15480.patch uploaded by dimas (license 88) ........ 2009-07-09 23:56 +0000 [r205731] Richard Mudgett * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller. Add missing clearing of the dialing flag when the ISDN call is CONNECTED. (i.e. When libpri generates the event PRI_EVENT_ANSWER.) (closes issue #15420) Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585) Tested by: scottbmilne, alecdavis (closes issue #15416) Reported by: avinoash (closes issue #15389) Reported by: alecdavis This patch should also fix the following issue: (issue #15205) Reported by: vinsik ........ 2009-07-09 21:27 +0000 [r205699] Kevin P. Fleming * include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 205696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ 2009-07-09 16:19 +0000 [r205595-205603] David Vossel * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ ................ * channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /: Merged revisions 205479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ 2009-07-09 08:34 +0000 [r205535] Michiel van Baak * /, main/ssl.c: Merged revisions 205532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 | mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines pthread_self returns a pthread_t which is not an unsigned int on all pthread implementations. Casting it to an unsigned int fixes compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit ........ 2009-07-08 22:15 +0000 [r205411-205413] David Vossel * include/asterisk/pbx.h, include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c: Merged revisions 205412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ ................ * /, main/devicestate.c: Merged revisions 205410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205410 | dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines missing comma in devstatestring array ........ 2009-07-08 19:28 +0000 [r205353] Mark Michelson * apps/app_queue.c, /: Merged revisions 205350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines Merged revisions 205349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines Prevent phantom calls to queue members. If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) ........ ................ 2009-07-08 18:22 +0000 [r205302] Jason Parker * config.guess, config.sub, /: Merged revisions 205291 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205291 | qwell | 2009-07-08 13:19:46 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line Update config.guess and config.sub from the savannah.gnu.org git repo. ........ ................ 2009-07-08 18:18 +0000 [r205287] David Brooks * /, main/features.c: Merged revisions 205254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 | dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines Fixes Park() argument handling Park() was not respecting the arguments passed to it. Any extension/context/priority given to it was being ignored. This patch remedies this. (closes issue #15380) Reported by: DLNoah ........ 2009-07-08 17:00 +0000 [r205223] Tilghman Lesher * main/say.c: oops, fixing build 2009-07-08 16:55 +0000 [r205217] David Vossel * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ ................ 2009-07-08 16:30 +0000 [r205207] Tilghman Lesher * /, main/say.c: Merged revisions 205196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines Add redirection warnings for the invalid language codes previously removed. ........ ................ 2009-07-08 15:57 +0000 [r205148-205154] Russell Bryant * /, main/ssl.c: Merged revisions 205151 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 | russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines Use tabs instead of spaces for indentation. ........ * include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c, /, main/Makefile, res/res_crypto.c, main/ssl.c (added): Merged revisions 205120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ 2009-07-06 13:41 +0000 [r204951] Kevin P. Fleming * main/channel.c, /: Merged revisions 204948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 | kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 lines Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This change allows applications that request T.38 negotiation on a channel that does not support it to get the proper indication that it is not supported, rather than thinking that negotiation was started when it was not. ........ 2009-07-02 22:06 +0000 [r204838] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ ................ 2009-07-02 16:12 +0000 [r204711] David Vossel * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c: Merged revisions 204710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ ................ 2009-06-30 21:30 +0000 [r204611] Tilghman Lesher * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar. (closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-06-30 18:55 +0000 [r204478] Jason Parker * /, main/say.c: Merged revisions 204475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | 9 lines Merged revisions 204474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing. ........ ................ 2009-06-30 18:44 +0000 [r204473] Tilghman Lesher * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge of revisions 204470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines Recorded merge of revisions 204469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines "tw" is the language specification for Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier ........ ................ 2009-06-30 17:22 +0000 [r204442] Russell Bryant * configs/res_config_sqlite.conf (removed), configs/res_config_sqlite.conf.sample (added), /: Merged revisions 204440 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 | russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines Rename res_config_sqlite.conf to res_config_sqlite.conf.sample (missing .sample). ........ 2009-06-29 22:53 +0000 [r204250-204304] Mark Michelson * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines Merged revisions 204300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ ................ * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines Fix build oops. ........ ................ 2009-06-27 09:55 +0000 [r203961] Russell Bryant * CHANGES, /: Merged revisions 203960 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 | russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt. ........ 2009-06-27 01:24 +0000 [r203941] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines The ISDN CPE side should not exclusively pick B channels normally. Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni ........ ................ 2009-06-26 22:14 +0000 [r203857] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500 (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo channel after dahdi restart (closes issue #14477) Reported by: timking ........ ................ 2009-06-26 21:27 +0000 [r203782-203828] Russell Bryant * /, main/file.c: Merged revisions 203802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines Merged revisions 203785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines Don't fast forward past the end of a message. This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj ........ ................ * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 | russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines Ensure the TCP read buffer is fully initialized before handling each packet. (closes issue #14452) Reported by: umberto71 ........ 2009-06-26 20:18 +0000 [r203731] David Brooks * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines Fixing voicemail's error in checking max silence vs min message length Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented as seconds. Also, the inequality was reversed. The warning, if triggered, was "Max silence should be less than minmessage or you may get empty messages", which should have been logged if max silence was greater than minmessage, but the check was for less than. Also, conforming if statement to coding guidelines. closes issue #15331) Reported by: markd Review: https://reviewboard.asterisk.org/r/293/ ........ 2009-06-26 19:49 +0000 [r203715] Russell Bryant * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c: Merged revisions 203702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 | russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines Make invalid hints report Unavailable instead of Idle. (closes issue #14413) Reported by: pj ........ 2009-06-26 19:48 +0000 [r203712] David Vossel * channels/chan_iax2.c, /: Merged revisions 203710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009) | 7 lines moving debug message from level 0 to 1. (closes issue #15404) Reported by: leobrown Patches: iax_codec_debug.patch uploaded by leobrown (license 541) ........ 2009-06-26 19:42 +0000 [r203709] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) | 16 lines Check if polarityonanswerdelay has elapsed before setting a channel as answered after a polarity reversal. Previously on a polarity switch event chan_dahdi would set the channel immediately as answered. This would cause problems if a polarity reversal occurred when the line was picked up as the dial would not have yet occurred. Now if the polarity reversal occurs before delay has elapsed after coming off hook or an answer, it is ignored. Also, some refactoring was done in _handle_event. (closes issue #13917) Reported by: alecdavis Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ 2009-06-26 19:38 +0000 [r203705] Joshua Colp * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c, main/channel.c, main/frame.c, /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 203699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ 2009-06-25 21:46 +0000 [r203445] David Vossel * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25 Jun 2009) | 4 lines fixes a few redundant conditions (issue #15269) ........ 2009-06-25 21:21 +0000 [r203400] Terry Wilson * main/cli.c, /: Merged revisions 203381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009) | 11 lines Merged revisions 203380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines I didn't see that Mark already fixed the underlying issue! Yay for removing useless code. ........ ................ 2009-06-25 21:08 +0000 [r203379] Russell Bryant * /, main/features.c: Merged revisions 203376 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines Merged revisions 203375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines Fix a case where CDR answer time could be before the start time involving parking. (closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks ........ ................ 2009-06-25 19:27 +0000 [r203276] Jason Parker * channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) | 10 lines Unmute when we get a dtmfup (we muted on dtmfdown) event. This would occasionally cause one-way audio when using hardware DTMF detection. (closes issue #14761) Reported by: tzafrir Patches: v1-14761.patch uploaded by dimas (license 88) Tested by: tzafrir, dimas ........ 2009-06-25 16:08 +0000 [r203119] Russell Bryant * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) | 18 lines Merged revisions 203115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines Resolve a crash related to a T.38 reinvite race condition. This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) ........ ................ 2009-06-24 21:27 +0000 [r203077] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500 (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. ........ ................ 2009-06-24 18:30 +0000 [r202970] Mark Michelson * /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun 2009) | 9 lines Merged revisions 202966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding the same thing in-line. ........ ................ 2009-06-24 18:11 +0000 [r202928] Joshua Colp * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 | file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines Ensure the default settings are applied for T.38 when we set it up for a peer. ........ 2009-06-23 23:58 +0000 [r202842] Sean Bright * doc/tex/cdrdriver.tex, /, doc/tex/billing.tex: Merged revisions 202840-202841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202840 | seanbright | 2009-06-23 19:53:45 -0400 (Tue, 23 Jun 2009) | 1 line Remove some trailing whitespace before making content changes. ........ r202841 | seanbright | 2009-06-23 19:57:07 -0400 (Tue, 23 Jun 2009) | 1 line Change some section names in the CDR tex documentation. ........ 2009-06-23 22:47 +0000 [r202805] Russell Bryant * doc/tex/cdrdriver.tex, /: Merged revisions 202804 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202804 | russell | 2009-06-23 17:47:26 -0500 (Tue, 23 Jun 2009) | 2 lines Clean up section hierarchy for the CDR chapter. ........ 2009-06-23 22:12 +0000 [r202765] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) | 1 line I could have sworn I committed this patch ages ago, but... bug fix with setting NAI properly on linksets in certain situations. ........ 2009-06-23 16:33 +0000 [r202673] David Vossel * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009) | 18 lines Merged revisions 202671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport (closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ ........ ................ 2009-06-22 20:19 +0000 [r202495-202511] Russell Bryant * main/channel.c, /: Merged revisions 202497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009) | 11 lines Merged revisions 202496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines Report CallerID change during a masquerade. Reported by: markster ........ ................ * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) | 9 lines Merged revisions 202414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) | 2 lines Make Polycom subscription type override check more explicit. ........ ................ 2009-06-22 16:31 +0000 [r202473] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid potential crashes during reload. Pointed out by Russell while working on the CEL branch. ........ 2009-06-22 15:37 +0000 [r202411] David Vossel * main/loader.c, /, include/asterisk/module.h: Merged revisions 202410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 | dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines attempting to load running modules Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice. ........ 2009-06-22 15:17 +0000 [r202340-202346] Mark Michelson * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines Fix a situation in which Asterisk would not stop retransmitting 487s. If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line left from previous commit. ........ ................ * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) | 31 lines Merged revisions 202336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines Fix a possible infinite loop in SDP parsing during glare situation. There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon ........ ................ 2009-06-21 16:16 +0000 [r202261-202265] Russell Bryant * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 | russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines Fix possibility of crashiness during reload in custom fields handling. ........ * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 | russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines Standardize return values of load_config() so reload() doesn't report an error on success. ........ 2009-06-20 19:14 +0000 [r202186] Sean Bright * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 | seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5 lines Fix version detection for API changes in spandsp. (closes issue #15355) Reported by: deuffy ........ 2009-06-19 21:08 +0000 [r202007] Matthew Nicholson * channels/chan_sip.c: Added deadlock protection to try_suggested_sip_codec in chan_sip.c. Review: https://reviewboard.asterisk.org/r/287/ 2009-06-19 20:26 +0000 [r201995] David Vossel * channels/chan_iax2.c, /: Merged revisions 201994 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500 (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) | 8 lines timestamp was being converted to host order as a short rather than a long (closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) ........ ................ 2009-06-19 15:49 +0000 [r201785-201906] Tilghman Lesher * res/res_config_odbc.c, /: Merged revisions 201904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009) | 4 lines Fix 2 typos and add support for wide character types. Reported by Benny Amorsen via the asterisk-users mailing list. http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html ........ * /, main/features.c: Merged revisions 201829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009) | 13 lines Merged revisions 201828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines If the "h" extension fails, give it another chance in main/pbx.c. If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. ........ ................ * /, apps/Makefile: Merged revisions 201783 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 | tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines One of the changes in 1.6.1 was to allow app_directory to use functionality within app_voicemail for directory functions. It is therefore no longer necessary for app_directory to be linked against the ODBC libraries (and it never was necessary for app_directory to be linked against IMAP, though it was). ........ 2009-06-18 16:44 +0000 [r201679] David Vossel * channels/misdn/isdn_lib.c, utils/conf2ael.c, main/ast_expr2.c, utils/stereorize.c, main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c, utils/extconf.c, channels/xpmr/xpmr.c, pbx/pbx_config.c, res/res_config_ldap.c, apps/app_rpt.c, main/asterisk.c, codecs/gsm/src/gsm_destroy.c, /, channels/h323/ast_h323.cxx: Merged revisions 201678 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel ........ 2009-06-18 15:40 +0000 [r201614] Russell Bryant * res/res_musiconhold.c, /: Merged revisions 201610 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201610 | russell | 2009-06-18 10:27:10 -0500 (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines Fix memory corruption and leakage related reloads of non files mode MoH classes. For Music on Hold classes that are not files mode, meaning that we are executing an application that will feed us audio data, we use a thread to monitor the external application and read audio from it. This thread also makes use of the MoH class object. In the MoH class destructor, we used pthread_cancel() to ask the thread to exit. Unfortunately, the code did not wait to ensure that the thread actually went away. What needed to be done is a pthread_join() to ensure that the thread fully cleans up before we proceed. By adding this one line, we resolve two significant problems: 1) Since the thread was never joined, it never fully goes away. So, on every reload of non-files mode MoH, an unused thread was sticking around. 2) There was a race condition here where the application monitoring thread could still try to access the MoH class, even though the thread executing the MoH reload has already destroyed it. (issue #15109) Reported by: jvandal (issue #15123) Reported by: axisinternet (issue #15195) Reported by: amorsen (issue AST-208) ........ ................ 2009-06-18 15:23 +0000 [r201595] David Vossel * /, channels/chan_sip.c: Merged revisions 201570 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 | dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines parsing extension correctly from sip register lines If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'. (closes issue #15111) Reported by: ffs Patches: chan_sip.c_register-parser.patch uploaded by ffs (license 730) Tested by: ffs, dvossel ........ 2009-06-17 21:33 +0000 [r201533] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 201531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009) | 7 lines Initialize additional variables, to prevent a possible crash. (closes issue #15186) Reported by: ajohnson Patches: 20090528__issue15186.diff.txt uploaded by tilghman (license 14) Tested by: ajohnson ........ 2009-06-17 20:12 +0000 [r201461-201465] Mark Michelson * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines Fix problem with no audio due to ignoring the SDP. A recent change to our SDP version comparison made audio not function on some calls. This was because of a test wherein we were trying to see if an unsigned value was less than 0. This is a dumb comparison and arguably the compiler should have warned about it. Alas, though, it slipped past. Now it's fixed by changing the variable to be a signed type. Found by several developers. Tested by mnicholson and dbrooks. ........ * main/channel.c, /: Merged revisions 201458 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. ........ ................ 2009-06-17 20:01 +0000 [r201447-201454] David Vossel * doc/datastores.txt, /: Merged revisions 201453 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 | dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines ast_channel_datastore_alloc is no longer used. updating datastores.txt to reflect that. ........ * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines StopMixMonitor race condition (not giving up file immediately) StopMixMonitor only indicates to the MixMonitor thread to stop writing to the file. It does not guarantee that the recording's file handle is available to the dialplan immediately after execution. This results in a race condition. To resolve this, the filestream pointer is placed in a datastore on the channel. When StopMixMonitor is called, the datastore is retrieved from the channel and the filestream is closed immediately before returning to the dialplan. Documentation indicating the use of StopMixMonitor to free files has been updated as well. (closes issue #15259) Reported by: travisghansen Tested by: dvossel Review: https://reviewboard.asterisk.org/r/283/ ........ ................ 2009-06-17 19:49 +0000 [r201446] David Brooks * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) | 16 lines Merged revisions 201380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks ........ ................ 2009-06-17 15:25 +0000 [r201360] David Vossel * /, channels/chan_sip.c: Merged revisions 201344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 | dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines SIP registry ref count error During a sip reload, the list of sip_registry objects are supposed to be traversed, unlinked, and destroyed, but destruction never takes place due to a ref counting error. This causes a memory leak when registry items are removed from sip.conf and reloaded. While the registries are removed from the global list, they are not removed from the scheduler. Because of this, SIP register attempts continue to be sent out for the item even though it may no longer be in the .conf. (closes issue #15295) Reported by: amorsen Review: https://reviewboard.asterisk.org/r/282/ ........ 2009-06-17 12:06 +0000 [r201265] Kevin P. Fleming * /, include/asterisk/linkedlists.h: Merged revisions 201262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is *not*, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ ................ 2009-06-16 22:30 +0000 [r201224] David Vossel * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 | dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines fix issue with build_contact introduced by the "SIP trasnport type issues" commit ........ 2009-06-16 19:47 +0000 [r200990-201097] Kevin P. Fleming * include/asterisk/frame.h, apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c, main/autoservice.c, main/frame.c, /, apps/app_meetme.c, main/slinfactory.c, include/asterisk/linkedlists.h, main/file.c, include/asterisk/channel.h: Merged revisions 201056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ ................ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged revisions 201090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler attribute checking. Defaulting to 'static' for the function scope was bad... so remove it. ........ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged revisions 200985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 | kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7 lines Fix problems with new compiler attribute checking in configure script. The last changes to ast_gcc_attribute.m4 caused some problems checking for various attributes, because the scope of the symbol the attribute is applied to can be important; this patch allows the scope to be specified for the check. ........ 2009-06-16 16:28 +0000 [r200984] David Vossel * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines SIP transport type issues What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ ........ 2009-06-16 16:05 +0000 [r200948] Michiel van Baak * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009) | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail can only use one storage module at the moment. Because it's unclear that selecting one of the storage modules in menuselect will disable filesystem storage we now have a FILE_STORAGE option that conflicts with the other modules. (closes issue #15333) ........ 2009-06-16 12:55 +0000 [r200842] Eliel C. Sardanons * res/res_smdi.c, /: Merged revisions 200841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200841 | eliel | 2009-06-16 08:32:00 -0400 (Tue, 16 Jun 2009) | 6 lines Show the interface name on error, if it is not found. If the smdiport specified is not found, show the interface name instead of '(null)'. ........ 2009-06-16 02:41 +0000 [r200807] Moises Silva * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 200799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200799 | moy | 2009-06-15 21:24:30 -0500 (Mon, 15 Jun 2009) | 2 lines keep backwards compatible chan_dahdi with older openr2 versions by not using the new skip category feature unless supported ........ 2009-06-16 01:30 +0000 [r200690-200765] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines Ensure that configure-script testing for compiler attributes actually works. The configure script tests for compiler attributes didn't actually enable enough warnings or provide a proper test harness to determine whether the compiler supports the attribute in question or not; this caused gcc 4.1 to report that it supports 'weakref', but it doesn't actually support it in the way that is needed for our optional API mechanism. The new configure script test will properly distinguish between full support and partial support for this attribute, among others. ........ * CHANGES, /: Merged revisions 200726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 | kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 lines Document the new automatic 'ignoresdpversion' behavior. Asterisk will now automatically ignore incorrect incoming SDP version numbers when necessary to complete a T.38 re-INVITE operation. ........ * /, channels/chan_sip.c: Merged revisions 200689 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 11 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ 2009-06-15 15:23 +0000 [r200517] Mark Michelson * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) | 11 lines Merged revisions 200513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ ................ 2009-06-14 06:33 +0000 [r200512] Moises Silva * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, build_tools/menuselect-deps.in: Merged revisions 200477 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200477 | moy | 2009-06-14 01:13:48 -0500 (Sun, 14 Jun 2009) | 3 lines added openr2 to menuselect-deps.in, recent commit in menuselect made me realize this was never done but was working anyways also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample ........ 2009-06-12 19:08 +0000 [r200364] Mark Michelson * main/channel.c, /: Merged revisions 200361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines Merged revisions 200360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins ........ ................ 2009-06-12 02:20 +0000 [r200198-200255] Sean Bright * contrib/init.d/rc.debian.asterisk, /: Merged revisions 200254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200254 | seanbright | 2009-06-11 22:20:19 -0400 (Thu, 11 Jun 2009) | 5 lines Call chgrp instead of chown when setting run directory group ownership. (issue #13153) Reported by: pabelanger ........ * Makefile, /: Merged revisions 199781 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 | seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 lines Fix all of the parallel build warnings issued when running make -j#. ........ * /: Undo block of revision 199782 (will be merging it momentarily) 2009-06-11 21:35 +0000 [r200172] Terry Wilson * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null 2009-06-11 21:18 +0000 [r200154] Mark Michelson * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 | mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 lines Fix a crash due to a potentially NULL p->options. Thanks to mnicholson for pointing it out. ........ 2009-06-11 Leif Madsen * Release Asterisk 1.6.2.0-beta3 2009-06-11 12:19 +0000 [r200051] Leif Madsen * build_tools/make_version_h, /, build_tools/make_version_c: Merged revisions 200039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 | lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines Fix path for .flavor and .version (issue #14737) Reported by: davidw Patches: flavor.patch uploaded by davidw (license 780) Tested by: davidw ........ 2009-06-10 20:37 +0000 [r199998] David Brooks * main/pbx.c, /: Fixes the argument order in definition of new_find_extension(). In the definition of new_find_extension(), the arguments 'callerid' and 'label' were swapped. The prototype declaration and all calls to the function are ordered 'callerid' then 'label', but the function itself was ordered 'label' then 'callerid'. (closes issue #15303) Reported by: JimDickenson 2009-06-10 20:18 +0000 [r199966] Mark Michelson * /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 | mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6 lines Only try to use the invite_branch on outgoing INVITEs with auth credentials. I have added a comment to the code to help ease understanding of the logic here as well. ........ 2009-06-10 16:13 +0000 [r199860] Sean Bright * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ ................ 2009-06-09 20:48 +0000 [r199744-199819] David Vossel * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer. (closes issue #15283) Reported by: jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614) Tested by: jthurman, dvossel ........ * main/loader.c, /, res/res_timing_pthread.c, include/asterisk/module.h, res/res_timing_dahdi.c, res/res_timing_timerfd.c: Merged revisions 199743 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines module load priority This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ ........ 2009-06-08 19:39 +0000 [r199634] Sean Bright * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ ................ 2009-06-08 17:42 +0000 [r199591] Mark Michelson * /, channels/chan_sip.c: Recorded merge of revisions 199588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon, 08 Jun 2009) | 9 lines Fix a deadlock that could occur when setting rtp stats on SIP calls. (closes issue #15143) Reported by: cristiandimache Patches: 15143.patch uploaded by mmichelson (license 60) Tested by: cristiandimache ........ 2009-06-06 21:39 +0000 [r199369] Russell Bryant * Makefile, /: Merged revisions 199368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199368 | russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines Switch from "echo -n" to printf. On my mac, the -n was just getting printed out. ........ 2009-06-05 21:25 +0000 [r199299] David Vossel * include/asterisk/devicestate.h, /, main/devicestate.c: Merged revisions 199298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ ................ 2009-06-05 13:52 +0000 [r199230] Mark Michelson * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines Correct "dahdi show channels" output when specifying a group. Since a DAHDI channel may belong to multiple groups, we need to use a bitwise and instead of equivalence to determine whether to display the channel information. (closes issue #15248) Reported by: gentian Patches: 15248.patch uploaded by mmichelson (license 60) Tested by: gentian ........ 2009-06-04 19:15 +0000 [r199140] David Vossel * channels/chan_iax2.c, /: Merged revisions 199139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ ................ 2009-06-04 14:53 +0000 [r199054] Sean Bright * include/asterisk/_private.h, main/asterisk.c, main/loader.c, /: Merged revisions 199051 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ ................ 2009-06-03 15:24 +0000 [r198827-198886] David Vossel * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 198856 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ ........ * channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines fixes issue with channels not going down after transfer Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop. (closes issue #15216) Reported by: oxymoron Tested by: dvossel ........ 2009-06-02 13:51 +0000 [r198794] Joshua Colp * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 198791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ 2009-06-01 21:04 +0000 [r198730] Russell Bryant * channels/iax2-parser.c, /: Merged revisions 198729 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009) | 2 lines Tell the IAX2 parser about more control frame types. ........ 2009-06-01 18:44 +0000 [r198629] Tilghman Lesher * /, contrib/scripts/meetme.sql: Merged revisions 198626 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 Jun 2009) | 2 lines Add information for new meetme realtime fields ........ 2009-05-31 17:53 +0000 [r198471] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 198470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009) | 2 lines Fix documentation for FIELDQTY. ........ 2009-05-31 01:48 +0000 [r198440] Eliel C. Sardanons * /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) | 11 lines Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded. if dahdi_test_timer() fails, timing_funcs_handle remains NULL causing a crash when calling ast_unregister_timing_interface() with a NULL pointer. (closes issue #15234) Reported by: eliel Patches: timing_dahdi1.diff uploaded by eliel (license 64) ........ 2009-05-31 01:21 +0000 [r198436] Russell Bryant * res/res_smdi.c, /: Merged revisions 198312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) | 12 lines Merged revisions 198311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines Fix a crash that occurred when MWI SMDI messages expired. (closes issue #14561) Reported by: cmoss28 ........ ................ 2009-05-30 20:22 +0000 [r198297-198397] Sean Bright * res/res_jabber.c, /: Merged revisions 198375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 | seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 lines Properly terminate the receive buffer before sending to iksemel. aji_io_recv takes the maximum number of bytes to read (instead of the total buffer size), so we have to subtract 1 from our buffer size. Without this, when we receive packets that are larger than our buffer, iksemel will choke and things get wonky. (closes issue #15232) Reported by: lp0 Patches: 05302009_res_jabber.c.patch uploaded by seanbright (license 71) Tested by: seanbright, lp0 ........ * res/res_jabber.c, /: Merged revisions 198371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May 2009) | 19 lines Merged revisions 198370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines Properly terminate AMI JabberSend response messages. The response message (either Error or Success) needs an extra trailing \r\n after the fields to inform the client that the message is complete. (closes issue #14876) Reported by: srt Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71) asterisk_14876.patch uploaded by srt (license 378) trunk-14876-2.diff uploaded by phsultan (license 73) ........ ................ * apps/app_dial.c, /: Merged revisions 198285 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 198251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we treat a missing one. (closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer ........ ................ 2009-05-30 02:35 +0000 [r198250] Joshua Colp * /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 | file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines When removing all packets from a dialog we also need to free the data if present. ........ 2009-05-29 23:05 +0000 [r198148-198188] Russell Bryant * /, configs/modules.conf.sample: Merged revisions 198186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29 May 2009) | 2 lines Suggesting that only a single timing module be loaded is no longer necessary. ........ * /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009) | 2 lines Improve handling of trying to ACK too many timer expirations. ........ * /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009) | 38 lines Resolve issues with choppy sound when using res_timing_pthread. The situation that caused this problem was when continuous mode was being turned on and off while a rate was set for a timing interface. A very easy way to replicate this bug was to do a Playback() from behind a Local channel. In this scenario, a rate gets set on the channel for doing file playback. At the same time, continuous mode gets turned on and off about every 20 ms as frames get queued on to the PBX side channel from the other side of the Local channel. Essentially, this module treated continuous mode and a set rate as mutually exclusive states for the timer to be in. When I dug deep enough, I observed the following pattern: 1) Set timer to tick every 20 ms. 2) Wait almost 20 ms ... 3) Continuous mode gets turned on for a queued up frame 4) Continuous mode gets turned off 5) The timer goes back to its tick per 20 ms. state but starts counting at 0 ms. 6) Goto step 2. Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick, but not most of the time. This is what produced the choppy sound (or sometimes no sound at all). Now, the module treats continuous mode and a set rate as completely independent timer modes. They can be enabled and disabled independently of each other and things work as expected. (closes issue #14412) Reported by: dome Patches: issue14412.diff.txt uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt uploaded by russell (license 2) Tested by: DennisD, russell ........ 2009-05-29 19:26 +0000 [r198111] Eliel C. Sardanons * CREDITS, /: Merged revisions 198083 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198083 | eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines Apply anti-spam obfuscation to an email address. ........ 2009-05-29 19:14 +0000 [r198075] Matthew Nicholson * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged revisions 198072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ ................ 2009-05-29 18:40 +0000 [r198066] Joshua Colp * /, main/file.c: Merged revisions 198064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 | file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix a memory leak of the write buffer when writing a file. ........ 2009-05-29 18:18 +0000 [r198008] Sean Bright * Makefile, /: Merged revisions 198000 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 197998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May 2009) | 8 lines Fix 'make config' target for Slackware. There was a missing semi-colon after the echo statement in the Makefile that was causing problems for some users. Fix suggested by reporter. (closes issue #15225) Reported by: pdavis ........ ................ 2009-05-29 16:29 +0000 [r197994] Russell Bryant * /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009) | 2 lines Trim trailing whitespace so that I can work on this bug without it bothering me. :-) ........ 2009-05-28 23:54 +0000 [r197894] Leif Madsen * apps/app_mixmonitor.c, /: Merged revisions 197828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009) | 8 lines Update documentation in MixMonitor. Updated the MixMonitor documentation for the 'b' option so that it is more obvious that you must not optimize away the Local channel when using this option. (closes issue #14829) Reported by: licedey Tested by: mmichelson, licedey, lmadsen ........ 2009-05-28 18:50 +0000 [r197703] Joshua Colp * channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2 lines Fix a bug where the trunkmtu setting was not set to the default value of 1240 on load but was on reload. ........ 2009-05-28 16:15 +0000 [r197625] Eliel C. Sardanons * /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines Merged revisions 197562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines Use the address we already know when reloading a peer with nat=yes. If we already have an address for a peer, and we are reloading the sip configuration, try to use that address to contact the peer, instead of getting it from the Contact. (closes issue #15194) Reported by: ibc Patches: sip.patch uploaded by eliel (license 64) Tested by: manwe ........ ................ 2009-05-28 15:44 +0000 [r197548-197619] Mark Michelson * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: Merged revisions 197606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 ........ ................ * main/audiohook.c, apps/app_chanspy.c, /, include/asterisk/audiohook.h: Merged revisions 197543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ ................ 2009-05-28 14:56 +0000 [r197471-197542] Joshua Colp * /, main/utils.c: Merged revisions 197538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 | file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix a bug in stringfields where it did not actually free the pools of memory. (closes issue #15074) Reported by: pj ........ * /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines Merged revisions 197466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting. The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated (or it passes through unauthenticated) the proper nat flag is set. (closes issue #13823) Reported by: dimas ........ ................ 2009-05-28 11:40 +0000 [r197441] Gavin Henry * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, doc/ldap.txt, configs/res_ldap.conf.sample: issue #15155 and issue #15156 from trunk 2009-05-27 23:49 +0000 [r197375] Tilghman Lesher * /, main/xml.c: Merged revisions 197374 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197374 | tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines Revert commit 192032. This define is needed on Mac OS X. ........ 2009-05-27 22:23 +0000 [r197336] Kevin P. Fleming * include/asterisk/agi.h, /: Merged revisions 197335 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May 2009) | 3 lines Ensure that this header includes xmldoc.h, since it depends on it. ........ 2009-05-27 20:11 +0000 [r197263] Sean Bright * Makefile, /: Merged revisions 197260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 | seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6 lines Use bash explicitly when calling build_tools/mkpkgconfig from the Makefile. Since we use bashisms in build_tools/mkpkgconfig, we should call on bash explicitly when running from the Makefile, otherwise we get errors during a 'make install.' (closes issue #15209) Reported by: seandarcy ........ 2009-05-27 19:30 +0000 [r197247] Tilghman Lesher * /, funcs/func_cut.c: Recorded merge of revisions 197209 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500 (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines Use a different determinator on whether to print the delimiter, since leading fields may be blank. (closes issue #15208) Reported by: ramonpeek Patch by me, though inspired in part by a patch from ramonpeek ........ ................ 2009-05-27 17:28 +0000 [r197176] Jeff Peeler * main/channel.c, include/asterisk/channel.h: Fix broken attended transfers The bridge was terminating immediately after the attended transfer was completed. The problem was because upon reentering ast_channel_bridge nexteventts was checked to see if it was set and if so could possibly return AST_BRIDGE_COMPLETE. (closes issue #15183) Reported by: andrebarbosa Tested by: andrebarbosa, tootai, loloski 2009-05-27 16:12 +0000 [r196950-197092] Sean Bright * configs/smdi.conf.sample, configs/extensions.conf.sample, configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /, configs/vpb.conf.sample: Merged revisions 197089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May 2009) | 6 lines Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf in the sample configuration files. (closes issue #15207) Reported by: seandarcy ........ * /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines Display an error message when chan_alsa fails to load due to a missing or inaccessible configuration file. Before this change, when chan_alsa failed to load due to a missing or inaccessible configuration file, no message would be displayed. With this change, when chan_alsa fails to load due to a missing or inaccessible configuration file, a message will be displayed. (closes issue #14760) Reported by: Nick_Lewis Patches: chan_alsa.c-confload.patch uploaded by Nick (license 657) ........ * main/xmldoc.c, /: Merged revisions 196948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196948 | seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8 lines Reset the terminal to the correct fg/bg after XML documenation is rendered. (closes issue #15200) Reported by: ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright (license 71) Tested by: ajohnson ........ * main/manager.c, /: Merged revisions 196945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196945 | seanbright | 2009-05-26 18:38:05 -0400 (Tue, 26 May 2009) | 13 lines Add ActionID to CoreShowChannel event. There is inconsistency in how we handle manager responses that are lists of items and, unfortunately, third parties have come to rely on ActionID being on every event within those lists instead of just keeping track of the ActionID for the current response. This change makes CoreShowChannels include the ActionID with each CoreShowChannel event generated as a result of it being called. (closes issue #15001) Reported by: sum Patches: patchactionid2.patch uploaded by sum (license 766) ........ 2009-05-26 22:44 +0000 [r196870-196949] Russell Bryant * /, autoconf/ast_check_osptk.m4 (added), configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 196946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 | russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines Update configure script to check for OSP toolkit 3.5.0. (closes issue #14988) Reported by: tzafrir Patches: configure.ac.diff uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick (license 91) ........ * /, res/res_convert.c: Merged revisions 196843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009) | 16 lines Merged revisions 196826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines Resolve a file handle leak. The frames here should have always been freed. However, out of luck, there was never any memory leaked. However, after file streams became reference counted, this code would leak the file stream for the file being read. (closes issue #15181) Reported by: jkroon ........ ................ 2009-05-26 16:39 +0000 [r196793] Sean Bright * apps/app_queue.c, /: Merged revisions 196792 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196792 | seanbright | 2009-05-26 12:38:54 -0400 (Tue, 26 May 2009) | 2 lines Add a missing unref for queues in handle_statechange. ........ 2009-05-26 13:47 +0000 [r196661-196724] Joshua Colp * /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 | file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix a bug where the sip unregister CLI command did not completely unregister the peer. (closes issue #15118) Reported by: alecdavis Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585) ........ * contrib/scripts/safe_asterisk, /: Merged revisions 196658 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue, 26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7 lines Remove some bash specific stuff from safe_asterisk. (closes issue #10812) Reported by: paravoid Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license 46) ........ ................ 2009-05-23 05:29 +0000 [r196487] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 196456 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196456 | moy | 2009-05-22 23:27:47 -0500 (Fri, 22 May 2009) | 1 line set MFCR2_CATEGORY just when starting the pbx ........ 2009-05-22 21:59 +0000 [r196452] David Vossel * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 196416 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ 2009-05-22 19:48 +0000 [r196378] Eliel C. Sardanons * /, apps/app_minivm.c: Merged revisions 196377 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196377 | eliel | 2009-05-22 15:38:33 -0400 (Fri, 22 May 2009) | 11 lines Unregister every registered application by MiniVM. The MinivmMWI application was not being unregistered on unload and we were not able to load again the module or reload it. (closes issue #15174) Reported by: junky Patches: unregister_minivm_mwi.diff uploaded by junky (license 177) ........ 2009-05-22 13:59 +0000 [r196120] Joshua Colp * channels/chan_misdn.c, /: Merged revisions 196117 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri, 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist. (closes issue #12286) Reported by: lmamane ........ ................ 2009-05-21 19:15 +0000 [r196000] David Vossel * channels/chan_iax2.c, /: Merged revisions 195995 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement. (closes issue #15032) Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380) Tested by: guillecabeza (closes issue #14216) Reported by: Andrey Sofronov ........ ................ 2009-05-21 15:57 +0000 [r195883] Matthew Nicholson * main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195882 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ ................ 2009-05-20 23:31 +0000 [r195842] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 195839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195839 | tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines If a variable had a blank value upon the initial setting, then it would do nothing. Identified by Dmitry Andrianov via private email, fixed by me. ........ 2009-05-20 17:35 +0000 [r195639-195707] Joshua Colp * /, main/features.c: Merged revisions 195698 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 lines Fix some code that wrongly assumed a pointer would always be non-NULL when dealing with CDRs after a bridge. (closes issue #15079) Reported by: barryf ........ ................ * /, apps/app_meetme.c: Merged revisions 195636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 lines Fix a bug where the MeetMe option 'D' did not actually prompt for the pin. (closes issue #15050) Reported by: pmhaddad ........ ................ 2009-05-19 20:19 +0000 [r195531] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 195521 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195521 | tilghman | 2009-05-19 15:16:01 -0500 (Tue, 19 May 2009) | 14 lines Merged revisions 195520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009) | 7 lines Ensure thread keys are initialized before attempting to access them. (closes issue #14889) Reported by: jaroth Patches: app_voicemail.c.patch uploaded by msirota (license 758) Tested by: msirota, BlargMaN ........ ................ 2009-05-19 14:49 +0000 [r195452] Joshua Colp * /, channels/chan_sip.c: Merged revisions 195449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) | 14 lines Merged revisions 195448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 lines Fix a bug where direct RTP setup would partially occur even when disabled if the calling channel was answered. (issue #13545) Reported by: davidw (issue #14244) Reported by: mbnwa ........ ................ 2009-05-18 21:25 +0000 [r195405] Eliel C. Sardanons * main/manager.c, /: Merged revisions 195369 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195369 | eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines Fix the CLI command 'manager show command' documentation and functionality. The CLI command 'manager show command' supports passing multiple action names in the same line, but it was not allowing that because of a incorrect check in the argumentes counter. Also the documentation was updated to show that this usage of the command is possible. ........ 2009-05-18 20:55 +0000 [r195359-195373] Tilghman Lesher * apps/app_queue.c, include/asterisk/smdi.h, res/res_monitor.c, apps/app_voicemail.c, res/res_smdi.c, /, include/asterisk/monitor.h: Merged revisions 195370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195370 | tilghman | 2009-05-18 15:52:33 -0500 (Mon, 18 May 2009) | 15 lines Recorded merge of revisions 195366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ ................ * main/asterisk.c, /: Merged revisions 195320 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195320 | tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines Move the spawn of astcanary down, until after the call to daemon(3). This avoids possible conflicts with the internal implementation of daemon(3). (closes issue #15093) Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir ........ 2009-05-18 19:01 +0000 [r195319] Mark Michelson * apps/app_externalivr.c, /: Merged revisions 195316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May 2009) | 18 lines Fix externalivr's setvariable command so that it properly sets multiple variables. The command had a for loop that was guaranteed to only execute once since the continuation operation of the loop would set the input buffer NULL. I rewrote the loop so that its operation was more obvious, and it would set multiple variables correctly. I also reduced stack space required for the function, constified the input string, and modified the function so that it would not modify the input string while I was at it. (closes issue #15114) Reported by: chris-mac Patches: 15114.patch uploaded by mmichelson (license 60) Tested by: chris-mac ........ 2009-05-18 15:57 +0000 [r195212] Joshua Colp * main/frame.c, /: Merged revisions 195207 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195207 | file | 2009-05-18 12:53:26 -0300 (Mon, 18 May 2009) | 14 lines Merged revisions 195206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 lines Fix a typo which caused loss of audio when using G729 in some scenarios with a smoother present. (closes issue #15105) Reported by: bamby Patches: process-vad-correctly.diff uploaded by bamby (license 430) ........ ................ 2009-05-18 14:54 +0000 [r195164] Eliel C. Sardanons * apps/app_dial.c, main/pbx.c, /, apps/app_macro.c: Merged revisions 195162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195162 | eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines Warn about the use of the application WaitExten() within a Macro(). Update applications documentation to warn the user about the use of the WaitExten() application within a Macro(). Recommend the use of Read() instead. (closes issue #14444) Reported by: ewieling ........ 2009-05-18 14:00 +0000 [r195099] Joshua Colp * main/rtp.c, /: Merged revisions 195096 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) | 12 lines Merged revisions 195095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited. (closes issue #13569) Reported by: bkw918 ........ ................ 2009-05-18 13:50 +0000 [r195093-195094] Eliel C. Sardanons * /, main/xml.c: Merged revisions 195075 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195075 | eliel | 2009-05-18 09:30:34 -0400 (Mon, 18 May 2009) | 3 lines Do not avoid loading the XML documentation if not XInclude substitution is done. ........ * doc/appdocsxml.dtd, Makefile, /, main/xml.c: Merged revisions 194982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194982 | eliel | 2009-05-16 16:01:22 -0400 (Sat, 16 May 2009) | 20 lines Allow to include sections of other parts of the xml documentation. Avoid duplicating xml documentation by allowing to include other parts of the xml documentation using XInclude. Example: (Insert this line to include the synopsis of the CHANNEL function xml documentation). It is also possible to include documentation from other files in the 'documentation/' directory using the href="" attribute inside a xinclude element. (closes issue #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling ........ 2009-05-18 13:39 +0000 [r195092] Joshua Colp * /, channels/chan_sip.c: Merged revisions 195089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 | file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where specifying an empty outboundproxy would cause packets to get sent to ourself. (closes issue #15106) Reported by: timeshell ........ 2009-05-18 13:14 +0000 [r195024] Russell Bryant * main/manager.c, /: Merged revisions 195021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195021 | russell | 2009-05-18 07:59:11 -0500 (Mon, 18 May 2009) | 12 lines Recorded merge of revisions 195020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009) | 5 lines Don't try to unlock a bogus channel. (closes issue #15144) Reported by: cristiandimache ........ ................ 2009-05-16 18:43 +0000 [r194946] Eliel C. Sardanons * main/pbx.c, /: Merged revisions 194945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194945 | eliel | 2009-05-16 14:32:11 -0400 (Sat, 16 May 2009) | 8 lines Fix a missing unlock in case of error, and a missing free(). Always free the allocated memory for a string field, because we are always using it (not only when xmldocs are enabled). Also if there is an error allocating memory for the string field remember to unlock the list of registered applications, before returning. ........ 2009-05-15 22:48 +0000 [r194836-194877] David Vossel * channels/chan_iax2.c, /: Merged revisions 194874 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194874 | dvossel | 2009-05-15 17:44:44 -0500 (Fri, 15 May 2009) | 23 lines Merged revisions 194873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001) (closes issue #14867) Reported by: aragon Tested by: dvossel (closes issue #14717) Reported by: mobeck Patches: regauth_loop_update_patch.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ * channels/chan_iax2.c, channels/iax2-parser.c, channels/iax2-parser.h, /, channels/iax2.h: Merged revisions 194833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009) | 24 lines Merged revisions 194557,194685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away. (closes issue #14207) Reported by: clive18 Review: https://reviewboard.asterisk.org/r/246/ ........ r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines Update to previous IAX2 "Ghost" Channels patch. Fixed some comments made on reviewboard for the previous patch. (issue #14207) ........ ................ 2009-05-15 18:44 +0000 [r194717-194768] Russell Bryant * configs/logger.conf.sample, /: Merged revisions 194765 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194765 | russell | 2009-05-15 13:43:42 -0500 (Fri, 15 May 2009) | 10 lines Merged revisions 194764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) | 2 lines Fix some spelling fail. ........ ................ * /, codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Merged revisions 194722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194722 | russell | 2009-05-15 12:59:08 -0500 (Fri, 15 May 2009) | 4 lines Shuttle some bits around to address some gain issues with G.722. (closes AST-209) ........ * codecs/Makefile, codecs/g722/Makefile (removed), /: Merged revisions 194718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194718 | russell | 2009-05-15 12:37:12 -0500 (Fri, 15 May 2009) | 2 lines Further simplify codec_g722 build. ........ * codecs/Makefile, /: Merged revisions 194714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194714 | russell | 2009-05-15 12:24:39 -0500 (Fri, 15 May 2009) | 2 lines Actually force running make for g722. ........ 2009-05-15 13:47 +0000 [r194650] Michiel van Baak * CREDITS, /: Merged revisions 194649 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194649 | mvanbaak | 2009-05-15 15:43:24 +0200 (Fri, 15 May 2009) | 2 lines add eliel ........ 2009-05-15 13:42 +0000 [r194648] Eliel C. Sardanons * doc/appdocsxml.dtd, main/xmldoc.c, /: Merged revisions 194635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194635 | eliel | 2009-05-15 09:23:37 -0400 (Fri, 15 May 2009) | 16 lines Allow to specify an enumlist inside an enum. It was not possible to use an enumlist inside an enum: ... Now we will be able to insert as many levels as we want. (closes issue #15112) Reported by: lmadsen ........ 2009-05-14 22:31 +0000 [r194545] Kevin P. Fleming * /: Merged revisions 194520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194520 | kpfleming | 2009-05-14 17:26:02 -0500 (Thu, 14 May 2009) | 9 lines Merged revisions 194509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May 2009) | 1 line Update URL to Reviewboard ........ ................ 2009-05-14 22:23 +0000 [r194510] Mark Michelson * /, channels/chan_sip.c: Merged revisions 194496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May 2009) | 30 lines Merged revisions 194484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines Fix a race condition where a reinvite could trigger a 482 response. The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ 2009-05-14 17:07 +0000 [r194437] Joshua Colp * /, apps/app_meetme.c: Merged revisions 194434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194434 | file | 2009-05-14 14:05:33 -0300 (Thu, 14 May 2009) | 7 lines Fix a bug where the 'T' option to Meetme did not work. (closes issue #15031) Reported by: Stochastic (closes issue #13801) Reported by: justdave ........ 2009-05-14 16:23 +0000 [r194431] Tilghman Lesher * main/pbx.c, /: Merged revisions 194430 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194430 | tilghman | 2009-05-14 11:22:14 -0500 (Thu, 14 May 2009) | 7 lines If the timing ended on a zero, then we would loop forever. (closes issue #14983) Reported by: teox Patches: 20090513__issue14983.diff.txt uploaded by tilghman (license 14) Tested by: teox ........ 2009-05-13 13:42 +0000 [r194213] Joshua Colp * main/rtp.c, /: Merged revisions 194209 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) | 18 lines Merged revisions 194208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. (closes issue #14815) Reported by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue #14460) Reported by: moliveras Tested by: moliveras ........ ................ 2009-05-13 00:54 +0000 [r194141] Tilghman Lesher * main/pbx.c, /: Merged revisions 194138 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194138 | tilghman | 2009-05-12 19:52:49 -0500 (Tue, 12 May 2009) | 14 lines Merged revisions 194137 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009) | 7 lines Fix logic for how to proceed with a single digit extension. (closes issue #15091) Reported by: andrew Patches: 20090512__issue15091.diff.txt uploaded by tilghman (license 14) Tested by: andrew ........ ................ 2009-05-12 22:48 +0000 [r194059] Matthew Nicholson * apps/app_queue.c, /: Merged revisions 194057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194057 | mnicholson | 2009-05-12 17:32:13 -0500 (Tue, 12 May 2009) | 22 lines Merged revisions 194028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May 2009) | 16 lines This change modifies app_queue to properly generate CDR records in failure situations. This involves setting a proper cdr disposition coresponding to the given failure condition and ensuring the proper information is stored in the cdr record. (closes issue #13691) Reported by: dferrer Tested by: mnicholson (closes issue #13637) Reported by: atis Tested by: atis ........ ................ 2009-05-12 20:51 +0000 [r193962] Mark Michelson * /, channels/chan_sip.c: Merged revisions 193954 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 | mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 lines Update spiral support in trunk and 1.6.X to match what is in 1.4. In 1.4, a SIP spiral is treated the same way as a call forward. This works much better than what is currently in trunk and 1.6.X. The code in trunk and 1.6.X did not create a new call to the recipient of the spiral, instead trying to continue the same call. In addition to just being plain wrong, this also had the side effect of only being able to spiral calls to other SIP channels. With this in place, as long as call forwards are honored, SIP spirals will work properly. This means that it will work for outbound calls made by the Queue, Dial, and Page applications. For originated calls and spool calls, however, the spiral will not work properly until a generic call forward mechanism is introduced into Asterisk. (relates to issue #13630) ........ 2009-05-12 20:42 +0000 [r193823-193959] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 193956 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193956 | tilghman | 2009-05-12 15:40:22 -0500 (Tue, 12 May 2009) | 13 lines Merged revisions 193955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009) | 6 lines Avoid initializing routines if the authentication fails. Fixes a crash (RR) issue. (closes issue #14508) Reported by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license 377) ........ ................ * apps/app_voicemail.c, /: Merged revisions 193870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193870 | tilghman | 2009-05-12 12:29:33 -0500 (Tue, 12 May 2009) | 2 lines Convert a THREADSTORAGE object into a simple malloc'd object (as suggested by Russell on -dev) ........ * apps/app_voicemail.c, /: Recorded merge of revisions 193756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193756 | tilghman | 2009-05-11 17:50:47 -0500 (Mon, 11 May 2009) | 25 lines Recorded merge of revisions 193755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009) | 18 lines Move 300 bytes around on the stack, to make more room for an extension buffer. This allows more concurrent extensions to be copied for a single voicemail, without creating a possibility of upsetting existing users, where a dialplan could run out of stack space where it had run fine before. Alternatively, we could have allocated off the heap, but that is a larger change and would have increased the chance for instability introduced by this change. This is really solved starting in 1.6.0.11, as the use of an ast_str buffer allows an unlimited number of extensions (up to available memory). We additionally create a new warning message when the buffer length is exceeded, permitting administrators to see an issue after the fact, whereas previously the list was silently truncated. (closes issue #14739) Reported by: p_lindheimer Patches: 20090417__bug14739.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer ........ ................ 2009-05-11 22:12 +0000 [r193719] Russell Bryant * /, res/res_timing_timerfd.c: Merged revisions 193718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193718 | russell | 2009-05-11 17:04:40 -0500 (Mon, 11 May 2009) | 12 lines Fix some timer state corruption. In res_timer_timerfd, handle the case that set_rate gets called while a timer is still in continuous mode. In this case, we want to remember the configured rate, but not actually set it until continuous mode has been disabled. Thanks to dvossel for finding and helping to debug the problem. (closes issue #15080) Reported by: dvossel Tested by: dvossel ........ 2009-05-11 19:17 +0000 [r193617] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 193614 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500 (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ ................ 2009-05-11 18:59 +0000 [r193612] Leif Madsen * /, funcs/func_channel.c: Update CHANNEL(transfercapabilities) documentation. (closes issue #15073) Reported by: pkempgen Patches: 20090511__issue15073__trunk.diff.txt uploaded by tilghman (license 14) 2009-05-10 17:08 +0000 [r193503] Joshua Colp * main/bridging.c, /: Merged revisions 193502 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193502 | file | 2009-05-10 14:07:46 -0300 (Sun, 10 May 2009) | 2 lines Fix a bug where receiving a control frame of subclass -1 would cause certain channels to get hung up. ........ 2009-05-09 11:33 +0000 [r193462] Russell Bryant * include/asterisk/event.h, /: Merged revisions 193461 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193461 | russell | 2009-05-09 06:33:09 -0500 (Sat, 09 May 2009) | 2 lines Minor documentation update for ast_event_queue(). ........ 2009-05-08 20:52 +0000 [r193390] David Vossel * /, channels/chan_sip.c: Merged revisions 193387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 | dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines TCP not matching valid peer. find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it. Review: http://reviewboard.digium.com/r/236/ ........ 2009-05-08 19:51 +0000 [r193350] Mark Michelson * apps/app_queue.c, /: Merged revisions 193349 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193349 | mmichelson | 2009-05-08 14:50:44 -0500 (Fri, 08 May 2009) | 12 lines Reset the members' call counts when resetting queue statistics. This helps to prevent odd scenarios where a queue will claim to have taken 0 calls, but the members appear to have taken a non-zero amount. (closes issue #15068) Reported by: sum Patches: patchreset.patch uploaded by sum (license 766) Tested by: sum ........ 2009-05-08 15:36 +0000 [r193336] Sean Bright * funcs/func_devstate.c, /: Merged revisions 193274 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193274 | seanbright | 2009-05-08 11:18:40 -0400 (Fri, 08 May 2009) | 2 lines Fix the spelling of UNAVAILABLE in func_devstate CLI completion. ........ 2009-05-08 14:55 +0000 [r193266] David Vossel * channels/misdn_config.c, /: Merged revisions 193263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500 (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines "misdn show config" segfaults asterisk, if no MSN lists (closes issue #14976) Reported by: alecdavis Patches: misdn_config.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, FabienToune ........ ................ 2009-05-08 14:12 +0000 [r193197] Kevin P. Fleming * configs/logger.conf.sample, /, main/logger.c: Merged revisions 193194 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May 2009) | 13 lines Merged revisions 193193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines Make absolute paths for logger channels work properly (Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. ........ ................ 2009-05-07 23:44 +0000 [r193123] Tilghman Lesher * main/pbx.c, /: Merged revisions 193120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009) | 26 lines Merged revisions 193119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) | 19 lines Fix Background within a Macro for FreePBX. If the single digit DTMF is an extension in the specified context, then go there and signal no DTMF. Otherwise, we should exit with that DTMF. If we're in Macro, we'll exit and seek that DTMF as the beginning of an extension in the Macro's calling context. If we're not in Macro, then we'll simply seek that extension in the calling context. Previously, someone complained about the behavior as it related to the interior of a Gosub routine, and the fix (#14011) inadvertently broke FreePBX (#14940). This change should fix both of these situations, but with the possible incompatibility that if a single digit extension does not exist (but a longer extension COULD have matched), it would have previously gone immediately to the "i" extension, but will now need to wait for a timeout. (closes issue #14940) Reported by: p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer ........ ................ 2009-05-07 22:51 +0000 [r193080] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 193077 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500 (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines Give a more helpful message when an incoming call's dialed extension does not match. Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. ........ ................ 2009-05-07 17:53 +0000 [r192936-193008] Tilghman Lesher * /, funcs/func_odbc.c: Merged revisions 193006 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193006 | tilghman | 2009-05-07 12:51:13 -0500 (Thu, 07 May 2009) | 7 lines Second result should not contain data from the first result. (closes issue #15039) Reported by: jims Patches: 20090506__issue15039.diff.txt uploaded by tilghman (license 14) Tested by: jims ........ * channels/chan_unistim.c, /: Merged revisions 192938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009) | 6 lines Send DTMF frame before playing back audio. (closes issue #14858) Reported by: barryf Patches: 20090507__bug14858.diff.txt uploaded by tilghman (license 14) ........ * /, channels/chan_sip.c: Merged revisions 192933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) | 17 lines Merged revisions 192932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines Eliminate repetition of fullcontact during reconstruction. If the fullcontact field appears in both the sippeers and the sipregs table, then during reconstruction of the field, it will otherwise be doubled. (closes issue #14754) Reported by: Alexei Gradinari Patches: 20090506__bug14754.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ ................ 2009-05-07 Leif Madsen * Release Asterisk 1.6.2.0-beta2 2009-05-06 22:20 +0000 [r192874] Jeff Peeler * /, main/features.c: Merged revisions 192861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009) | 17 lines Merged revisions 192858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines Make ParkedCall application stop execution of the dialplan after hang up Just changed park_exec to always return non-zero. I really wasn't entirely sure at first if this was a bug. Decided it was since it would be surprising when not using ParkedCall in the dialplan to hang up and have dialplan execution continue. (closes issue #14555) Reported by: francesco_r ........ ................ 2009-05-06 17:57 +0000 [r192813] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 190946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) | 1 line Make sure that we do not clear the down flag on the BRI during PTMP link transients. Also refix SS7 audio that the early media patch broke. ........ 2009-05-06 17:41 +0000 [r192637-192810] Joshua Colp * channels/chan_iax2.c, /: Merged revisions 192808 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) | 10 lines Fix a bug where a timer would be created but not acknowledged. This scenario crept up if chan_iax2 was loaded with no configuration file present. It would create a timer and tell it to go at an interval but the thread that normally acknowledges it would not be created because no configuration file was present. The timer will now be closed if no configuration file is present. (closes issue #15014) Reported by: madkins ........ * res/res_clialiases.c, /: Merged revisions 192736 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192736 | file | 2009-05-06 13:09:27 -0300 (Wed, 06 May 2009) | 4 lines Make the code that prevents an infinite loop from happening into a case insensitive check. (thanks eliel) ........ * res/res_clialiases.c, /: Merged revisions 192700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192700 | file | 2009-05-06 11:35:47 -0300 (Wed, 06 May 2009) | 5 lines Fix an infinite loop with tab completion of CLI aliases that reference themselves. (closes issue #15020) Reported by: junky ........ * /, channels/chan_sip.c: Merged revisions 192634 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) | 14 lines Merged revisions 192633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled. (closes issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded by dimas (license 88) ........ ................ 2009-05-05 20:02 +0000 [r192528] Sean Bright * /, static-http/astman.js: Merged revisions 192525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192525 | seanbright | 2009-05-05 15:57:49 -0400 (Tue, 05 May 2009) | 18 lines Merged revisions 192524 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May 2009) | 11 lines Fix Javascript error when using astman.js in Internet Explorer. Internet Explorer (tested with 7.0) does not like trailing commas on constructs like object initializers, so get rid of them to avoid some errors. (closes issue #15026) Reported by: rajnishgiri Patches: bug15026.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ 2009-05-05 18:27 +0000 [r192402-192480] Joshua Colp * /, main/features.c: Merged revisions 192462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) | 15 lines Merged revisions 192454 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines Fix an incorrect assumption that certain values on the channel will always exist when they may not. The CDR code involved with bridges wrongly assumed that the currently executing application and data values will always exist. It is possible for this to be false when call forwarding is involved. (closes issue #14984) Reported by: gincantalupo ........ ................ * apps/app_followme.c, /: Merged revisions 192430 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192430 | file | 2009-05-05 14:46:51 -0300 (Tue, 05 May 2009) | 12 lines Merged revisions 192429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5 lines Fix a bug where the followme application would continue trying numbers after the caller hung up. (closes issue #13624) Reported by: sgenyuk ........ ................ * /, channels/chan_sip.c: Merged revisions 192387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 | file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines Fix a bug with setting t38pt_udptl at the user or peer level. If an incoming call authenticated as a user or peer and t38pt_udptl was not set to yes in general then no UDPTL session would be present and any T38 related things would fail. This commit changes it so that if after authenticating T38 is enabled but no UDPTL session is present one will be created. (issue AST-215) ........ 2009-05-05 13:43 +0000 [r192298-192360] Kevin P. Fleming * main/astobj2.c, include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 192357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 | kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5 lines Correct some flaws in the memory accounting code for stringfields and ao2 objects Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that. ........ * main/astobj2.c, main/datastore.c, main/channel.c, /, include/asterisk/astobj2.h, include/asterisk/datastore.h, include/asterisk/channel.h: Merged revisions 192318 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May 2009) | 5 lines Properly account for memory allocated for channels and datastores As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself. ........ * include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 192279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 | kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5 lines Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function. ........ 2009-05-04 22:48 +0000 [r192217] David Vossel * channels/chan_iax2.c, /: Merged revisions 192214 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192214 | dvossel | 2009-05-04 17:44:51 -0500 (Mon, 04 May 2009) | 17 lines Merged revisions 192213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines global mohinterpret setting is ignored mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers. (closes issue #14728) Reported by: dimas Patches: v1-14728.patch uploaded by dimas (license 88) Tested by: dimas, dvossel ........ ................ 2009-05-04 19:34 +0000 [r192175] Kevin P. Fleming * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 192059 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192059 | kpfleming | 2009-05-04 18:24:16 +0200 (Mon, 04 May 2009) | 5 lines Ensure that astobj2 memory allocations are properly accounted for when MALLOC_DEBUG is used This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function. ........ 2009-05-04 19:31 +0000 [r192135-192173] Tilghman Lesher * /, configure, res/res_agi.c: Merged revisions 192171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192171 | tilghman | 2009-05-04 14:29:13 -0500 (Mon, 04 May 2009) | 8 lines Restore 'asyncagi break' command to 1.6.1 and higher. (closes issue #14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt uploaded by tilghman (license 14) 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: nikkk ........ * autoconf/ast_ext_tool_check.m4, /: Merged revisions 192132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04 May 2009) | 6 lines Pass libraries in LIBS, not LDFLAGS. (closes issue #14671) Reported by: Chainsaw Patches: asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by Chainsaw (license 723) ........ 2009-05-04 17:45 +0000 [r192097] Leif Madsen * apps/app_forkcdr.c, /: Merged revisions 192096 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192096 | lmadsen | 2009-05-04 13:42:56 -0400 (Mon, 04 May 2009) | 4 lines Commit documentation changes related to issue #14801. (issue #14801) ........ 2009-05-04 15:54 +0000 [r192033] Eliel C. Sardanons * /, main/xml.c: Merged revisions 192032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192032 | eliel | 2009-05-04 11:35:35 -0400 (Mon, 04 May 2009) | 3 lines Do not re-define _POSIX_C_SOURCE if it was already defined. ........ 2009-05-04 10:01 +0000 [r191958] Kevin P. Fleming * /, configs/modules.conf.sample: Merged revisions 191955 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04 May 2009) | 8 lines Ensure that by default only one console channel driver is loaded This configuration file was changed to ensure that only one console channel driver (chan_oss) is loaded by default, but the change would only work if chan_console was not built. Now it will work as expected; if chan_alsa or chan_console are built and installed, they will not be loaded unless explicity requested. ........ 2009-05-03 14:06 +0000 [r191885] Russell Bryant * Makefile, /: Merged revisions 191884 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191884 | russell | 2009-05-03 09:05:10 -0500 (Sun, 03 May 2009) | 2 lines Remove unnecessary compiler flag ........ 2009-05-02 18:48 +0000 [r191779] Kevin P. Fleming * /, main/logger.c: Merged revisions 191775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191775 | kpfleming | 2009-05-02 20:39:48 +0200 (Sat, 02 May 2009) | 5 lines Fix an error in queue_log file rotation optimization code This code was copy-and-pasted without properly changing references to event_rotate into queue_rotate, so under some conditions the log rotation would rotate queue_log even though it was not necessary. ........ 2009-05-02 15:52 +0000 [r191703] Sean Bright * main/asterisk.c, /: Merged revisions 191700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191700 | seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1 line Update copyright year to 2009 ........ 2009-05-01 20:02 +0000 [r191554-191563] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 191560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009) | 13 lines Merged revisions 191559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1. (closes issue #14993) Reported by: BigJimmy Patches: causepatch uploaded by BigJimmy (license 371) ........ ................ * channels/chan_iax2.c, /: Merged revisions 191494 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009) | 4 lines Set debug message back to DEBUG level. (closes issue #15007) Reported by: hulber ........ 2009-05-01 18:20 +0000 [r191508] Jeff Peeler * main/channel.c, /: Merged revisions 191489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009) | 15 lines Merged revisions 191488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) | 9 lines Fix DTMF not being sent to other side after a partial feature match This fixes a regression from commit 176701. The issue was that ast_generic_bridge never exited after the feature digit timeout had elapsed, which prevented the queued DTMF from being sent to the other side. This issue was reported to me directly. ........ ................ 2009-04-30 17:46 +0000 [r191224-191370] Tilghman Lesher * main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 191367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 | tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines Detect eaccess (or euidaccess) before using it. Reported by Andrew Lindh via the -dev list. ........ * main/asterisk.c, /: Merged revisions 191283 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191283 | tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11 lines Change working directory to / under certain conditions. If backgrounding and no core will be produced, then changing the directory won't break anything; likewise, if the CWD isn't accessible by the current user, then a core wasn't possible anyway. (closes issue #14831) Reported by: chris-mac Patches: 20090428__bug14831.diff.txt uploaded by tilghman (license 14) 20090430__bug14831.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac ........ * /, channels/h323/ast_h323.cxx, channels/chan_h323.c: Merged revisions 191219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 | tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines Make H.323 compile with FDLEAK detection code enabled ........ 2009-04-29 18:40 +0000 [r191139] David Brooks * pbx/pbx_config.c, /: Merged revisions 191136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191136 | dbrooks | 2009-04-29 13:32:58 -0500 (Wed, 29 Apr 2009) | 3 lines Removing crufty code that is no longer necessary. Code cleanup. ........ 2009-04-29 08:59 +0000 [r190994] Russell Bryant * main/indications.c, /: Merged revisions 190993 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190993 | russell | 2009-04-29 03:58:39 -0500 (Wed, 29 Apr 2009) | 7 lines Log an error message if indications.conf is not found. (closes issue #14990) Reported by: tzafrir Patches: indications_err.diff uploaded by tzafrir (license 46) ........ 2009-04-29 06:38 +0000 [r190985] TransNexus OSP Development * apps/app_osplookup.c, /: Merged revisions 190830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190830 | transnexus | 2009-04-28 17:10:42 +0800 (Tue, 28 Apr 2009) | 2 lines Updated for OSP Toolkit 3.5. ........ 2009-04-28 17:33 +0000 [r190907] Tilghman Lesher * doc/tex/cdrdriver.tex, /: Merged revisions 190904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190904 | tilghman | 2009-04-28 12:31:43 -0500 (Tue, 28 Apr 2009) | 2 lines UniqueID column has a maximum size of 150 ........ 2009-04-28 14:17 +0000 [r190732-190869] Kevin P. Fleming * Makefile, /: Merged revisions 190865 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190865 | kpfleming | 2009-04-28 09:15:47 -0500 (Tue, 28 Apr 2009) | 5 lines Build XML documention from *only* the source files that have docs in them Change the build process so that doc/core-en_US.xml is dependent solely on the source files that have documentation in them, not on all source files. ........ * /, Makefile.rules: Merged revisions 190861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190861 | kpfleming | 2009-04-28 09:12:09 -0500 (Tue, 28 Apr 2009) | 5 lines Remove Makefile rules for bison and flex sources We never, ever want these files to processed automatically, because we store the output files in Subversion and users should never need to rebuild them. ........ * /, configure, include/asterisk/autoconfig.h.in: Merged revisions 190725 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr 2009) | 13 lines Merged revisions 190721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines Fix 'inconsistent line endings' when autoconf 2.63 is used Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway. ........ ................ 2009-04-27 19:36 +0000 [r190729] Tilghman Lesher * main/pbx.c, /: Merged revisions 190726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190726 | tilghman | 2009-04-27 14:34:48 -0500 (Mon, 27 Apr 2009) | 4 lines Don't warn on pipe in the System call. (closes issue #14979) Reported by: pj ........ 2009-04-27 19:15 +0000 [r190666] Russell Bryant * res/res_smdi.c, /: Merged revisions 190663 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009) | 22 lines Merged revisions 190661-190662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) | 9 lines Resolve a crash in res_smdi when used with chan_dahdi. When chan_dahdi goes to get an SMDI message, it provides no search criteria. It just grabs the next message that arrives. This code was written with the SMDI dialplan functions in mind, since that is now the preferred method of using SMDI. However, this broke support of it being used from chan_dahdi. (closes AST-212) ........ r190662 | russell | 2009-04-27 14:03:59 -0500 (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661. ........ ................ 2009-04-27 16:28 +0000 [r190625] Mark Michelson * apps/app_queue.c, /: Merged revisions 190622 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190622 | mmichelson | 2009-04-27 11:26:14 -0500 (Mon, 27 Apr 2009) | 3 lines Update warning message to not have pipes and contain all options. ........ 2009-04-23 21:23 +0000 [r190383] Russell Bryant * /, channels/chan_sip.c: Merged revisions 190371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ 2009-04-23 20:44 +0000 [r190355] Tilghman Lesher * main/pbx.c, /: Merged revisions 190352 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190352 | tilghman | 2009-04-23 15:42:11 -0500 (Thu, 23 Apr 2009) | 7 lines Labels are sometimes (most of the time?) NULL for extensions. (closes issue #14895) Reported by: chris-mac Patches: 20090423__bug14895__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ 2009-04-23 19:18 +0000 [r190297] Joshua Colp * channels/chan_local.c, /: Merged revisions 190287 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r190287 | file | 2009-04-23 16:15:30 -0300 (Thu, 23 Apr 2009) | 13 lines Merged revisions 190286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 lines Fix a bug in chan_local glare hangup detection. If both sides of a Local channel were hung up at around the same time it was possible for one thread to destroy the local private structure and have the other thread immediately try to remove the already freed structure from the local channel list. ........ ................ 2009-04-23 17:47 +0000 [r190253] Mark Michelson * apps/app_queue.c, /: Merged revisions 190250 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190250 | mmichelson | 2009-04-23 12:45:35 -0500 (Thu, 23 Apr 2009) | 9 lines Fix reversed behavior of leavewhenempty option in queues.conf. (closes issue #14650) Reported by: alecdavis Patches: 14650.patch uploaded by mmichelson (license 60) Tested by: mmichelson, lmadsen ........ 2009-04-22 21:43 +0000 [r190096] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/lock.h: Merged revisions 190093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r190093 | tilghman | 2009-04-22 16:38:15 -0500 (Wed, 22 Apr 2009) | 14 lines Merged revisions 190092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) | 7 lines Detect availability of pthread_rwlock_timedwrlock() before using it. (closes issue #14930) Reported by: tilghman Patches: 20090420__bug14930.diff.txt uploaded by tilghman (license 14) Tested by: mvanbaak, tilghman ........ ................ 2009-04-22 21:18 +0000 [r189997-190066] Jeff Peeler * main/cli.c, funcs/func_groupcount.c, /, main/app.c, include/asterisk/channel.h: Merged revisions 190057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) | 9 lines Fix building of chan_h323 with gcc-3.3 There seems to be a bug with old versions of g++ that doesn't allow a structure member to use the name list. Rename list member to group_list in ast_group_info and change the few places it is used. (closes issue #14790) Reported by: stuarth ........ * channels/h323/chan_h323.h, /, channels/h323/ast_h323.cxx, channels/chan_h323.c: Merged revisions 189993 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189993 | jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines Make chan_h323 respect packetization settings and fix small reload issue. Previously, packetization settings were ignored and now they are not. A new config option 'autoframing' has been added to mirror the way chan_sip handles it. Turning on the autoframing option (available both as a global option or per peer) overrides the local settings with the remote packetization settings. Testing was performed with varying packetization levels with the following codecs: ulaw, alaw, gsm, and g729. Also, an unrelated config reload issue has been fixed in the case of the config file not changing. (closes issue #12415) Reported by: pj Patches: 2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7), modified by me ........ 2009-04-22 18:01 +0000 [r189986] Russell Bryant * /, main/features.c: Merged revisions 189951 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189951 | russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines Fix call parking callback. Pipes -> Commas. ........ 2009-04-22 16:04 +0000 [r189816-189914] Tilghman Lesher * channels/chan_unistim.c, /: Merged revisions 189911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009) | 7 lines Do not continue to receive DTMF, when the channel is hungup and about to be destroyed. (closes issue #14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt uploaded by tilghman (license 14) Tested by: barryf ........ * /, configure, configure.ac: Merged revisions 189813 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189813 | tilghman | 2009-04-22 01:33:08 -0500 (Wed, 22 Apr 2009) | 3 lines Detect liblua on SuSE, and add libm for linking for Fedora. (Reported via the -dev list, Subject: Compiling Asterisk with LUA) ........ 2009-04-21 20:45 +0000 [r189775] David Vossel * /, channels/chan_sip.c: Merged revisions 189771 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189771 | dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines Fixes segfault when switching UDP to TCP in sip.conf after reload. If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload. The problem is the socket type is changed to TCP but the fd may still be present for UDP. Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present. Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found. (closes issue #14727) Reported by: pj Tested by: dvossel Review: http://reviewboard.digium.com/r/229/ ........ 2009-04-20 22:11 +0000 [r189540] Tilghman Lesher * main/stdtime/localtime.c, /: Merged revisions 189539 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189539 | tilghman | 2009-04-20 17:10:25 -0500 (Mon, 20 Apr 2009) | 3 lines Use nanosleep instead of poll. This is not just because mmichelson suggested it, but also because Mac OS X puked on my poll(). ........ 2009-04-20 21:41 +0000 [r189536] Terry Wilson * apps/app_dial.c, /: Merged revisions 189495,189516 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189495 | twilson | 2009-04-20 16:24:34 -0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 Apr 2009) | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL ........ ................ r189516 | twilson | 2009-04-20 16:29:29 -0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009) | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is set ........ ................ 2009-04-20 21:36 +0000 [r189533] Sean Bright * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189464 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189464 | seanbright | 2009-04-20 17:09:59 -0400 (Mon, 20 Apr 2009) | 20 lines Merged revisions 189462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr 2009) | 13 lines Properly handle @s within hints in AEL. AEL was not handling the case of a device hint containing an @ symbol, which caused parking hints (e.g. hint(park:exten@context)) to error out the parser. This patch makes AEL treat the @ the same way it treats colon and ampersand now, meaning the characters are included in verbatim. (closes issue #14941) Reported by: bpgoldsb Patches: bug14941.patch uploaded by seanbright (license 71) Tested by: bpgoldsb ........ ................ 2009-04-20 17:11 +0000 [r189353] Joshua Colp * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 | file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines Fix a bug with non-UDP connections that caused dialogs to not get freed. This issue crept up because of a reference count issue on non-UDP based dialogs. The dialog reference count was increased when transmitting a packet reliably but never decreased. This caused the dialog structure to hang around despite being unlinked from the dialogs container. (closes issue #14919) Reported by: vrban ........ 2009-04-20 14:07 +0000 [r189281] Mark Michelson * main/channel.c, /: Merged revisions 189278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr 2009) | 18 lines Merged revisions 189277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr 2009) | 12 lines Move the check for chan->fdno == -1 to after the zombie/hangup check. Many users were finding that their hung up channels were staying up and causing 100% CPU usage. (issue #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch uploaded by mmichelson (license 60) Tested by: falves11, bamby ........ ................ 2009-04-18 01:42 +0000 [r189207-189208] David Vossel * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500 (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines National prefix inserted even when caller ID not available When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank. (closes issue #13207) Reported by: shawkris Patches: national_prefix.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/220/ ........ ................ * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500 (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening. (closes issue #14091) Reported by: evandro Patches: autologoff.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/225/ ........ ................ 2009-04-17 21:56 +0000 [r189140] Richard Mudgett * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 189137 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009) | 17 lines Merged revisions 188833,189134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines Modifed/added some debug messages. JIRA ABE-1835 ........ ................ 2009-04-17 20:21 +0000 [r189105] Mark Michelson * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines Prevent a crash when SIP blonde transferring an unbridged call. If one attempts to use the attended transfer button on a SIP phone to transfer an unbridged call (such as a call to an IVR) but hangs up while the target of the transfer is still ringing, we need to not crash. The problem was that ast_hangup was called from outside the channel thread. AST-211 ........ 2009-04-17 19:47 +0000 [r189081] Sean Bright * main/asterisk.c, /: Merged revisions 189077 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 | seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1 line Fix copy/paste error with 'transmit silence' flag. ........ 2009-04-17 17:31 +0000 [r189068] Matthew Nicholson * main/pbx.c, /: Merged revisions 189010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr 2009) | 12 lines Merged revisions 189009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr 2009) | 5 lines Make Busy() application set the CDR disposition to BUSY. (closes issue #14306) Reported by: cristiandimache ........ ................ 2009-04-17 14:50 +0000 [r188941-188950] Joshua Colp * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines Merged revisions 188946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines Fix a bug where a value used to create the channel name was bogus. This commit fixes the scenario where an incoming call is authenticated using a peer entry. Previously the channel name was created using either the username setting from the sip.conf entry or the IP address that the call came from. Now the channel name will be created using the peer name itself. This commit will not change the way the channel name is generated for users or friends. (closes issue #14256) Reported by: Nick_Lewis Patches: chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, file ........ ................ * channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri, 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 lines Fix a situation where the DAHDI channel private structure lock was not unlocked when it should have been. (issue AST-210) ........ ................ 2009-04-16 22:05 +0000 [r188777-188839] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009) | 14 lines Merged revisions 188835 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) | 7 lines Only update realtime, if global option rtupdate != false (closes issue #14885) Reported by: deepesh Patches: 20090413__bug14885.diff.txt uploaded by tilghman (license 14) Tested by: deepesh ........ ................ * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500 (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009) | 4 lines Umask should not be exported into global namespace. (closes issue #14912) Reported by: jcapp ........ ................ 2009-04-15 20:20 +0000 [r188474-188598] Mark Michelson * /, main/file.c: Merged revisions 188585 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr 2009) | 13 lines Merged revisions 188582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr 2009) | 7 lines Update ast_readvideo_callback to match ast_readaudio_callback. This fixes potential refcount errors that may occur on ast_filestreams. AST-208 ........ ................ * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 | mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3 lines Fix a couple of queue member reference leaks. ........ 2009-04-14 17:46 +0000 [r188259-188416] Joshua Colp * main/rtp.c, /: Merged revisions 188413 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 | file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix an incorrect clock rate when sending T140 text. (closes issue #14029) Reported by: epicac ........ * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 | file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix a bug with the change I made yesterday to outbound proxy support. Per discussion with oej on IRC we need the actual IP address, not the outbound proxy IP address, in the sa field. Upon further inspection this should make the behaviour of all other uses of the outbound proxy in the code. ........ 2009-04-14 05:47 +0000 [r188209-188213] Tilghman Lesher * main/pbx.c, /: Merged revisions 188210 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 | tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines As suggested by Russell, warn users when their dialplan arguments contain pipes, but not commas. ........ * /, utils/smsq.c: Merged revisions 188206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 | tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines Application delimiter is ',', not '|'. (closes issue #14881) Reported by: stegro Patches: smsq.patch uploaded by stegro (license 752) ........ 2009-04-13 19:33 +0000 [r188105] Mark Michelson * res/res_musiconhold.c, /: Merged revisions 188102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr 2009) | 5 lines Fix another crash related to cached realtime music on hold. This was another off-by-one problem caused by moh_register. ........ 2009-04-13 16:34 +0000 [r188070] Joshua Colp * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 | file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1. Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will be sending to. This has to be done because the logic that determines what local IP address to use in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address we are sending to. (closes issue #12006) Reported by: mnicholson ........ 2009-04-13 14:20 +0000 [r188039] Mark Michelson * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 | mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6 lines Set all queue variables on both the caller and member channels. This allows for the variables to be accessed if a member macro is run. Thanks to Grigoriy Puzankin for bringing this up on the -dev list. ........ 2009-04-10 20:28 +0000 [r187916] Jeff Peeler * channels/Makefile, /: Merged revisions 187906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 | jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines Fix module embedding for chan_h323. Include libchanh323.a in the modules.link file so that all the symbols can be resolved at link time. (closes issue #11966) Reported by: dome Patches: issue_11966.patch uploaded by kpfleming (license 421) Tested by: jpeeler ........ 2009-04-10 17:31 +0000 [r187769] Tilghman Lesher * contrib/scripts/sip-friends.sql, contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500 (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009) | 2 lines Add lastms column to the contributed table designs ........ ................ 2009-04-10 16:54 +0000 [r187724] Kevin P. Fleming * /, build_tools/embed_modules.xml: Merged revisions 187721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10 Apr 2009) | 5 lines clean up some patterns for files to remove add embedding support for bridge and test modules ........ 2009-04-10 16:05 +0000 [r187679] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 | tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines Ensure pvt is not NULL before dereferencing it. (closes issue #14784) Reported by: pj ........ 2009-04-10 16:01 +0000 [r187677] Russell Bryant * tests/test_sched.c, tests/test_heap.c, /: Merged revisions 187675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 | russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines Disable test modules by default. ........ 2009-04-10 03:57 +0000 [r187601] Tilghman Lesher * main/audiohook.c, main/bridging.c, main/channel.c, main/pbx.c, main/manager.c, /, include/asterisk/linkedlists.h, main/features.c, main/http.c, main/app.c, include/asterisk/lock.h: Merged revisions 187599 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines Modify headers and macros, according to Russell's suggestions on the -dev list ........ 2009-04-09 21:09 +0000 [r187564] Mark Michelson * /, channels/chan_sip.c: Merge revision 187488 from trunk. 2009-04-09 19:29 +0000 [r187531-187546] David Vossel * main/audiohook.c, /: Merged revisions 186379 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 | dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 6 lines audio_audiohook_write_list() did not correctly update sample size after ast_translate. audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) ........ * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) | 16 lines Merged revisions 185845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491 Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. (closes issue #12013) Reported by: alx Review: http://reviewboard.digium.com/r/213/ ........ ................ 2009-04-09 19:15 +0000 [r187496] Mark Michelson * res/res_musiconhold.c, /: Merged revisions 187421,187424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu, 09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using cached realtime moh. The moh_register function links an mohclass and then immediately unrefs the class since the container now has a reference. The problem with using realtime music on hold is that the class is allocated, registered, and started in one fell swoop. The refcounting logic resulted in the count being off by one. The same problem did not happen when using a static config because the allocation and registration of an mohclass is a separate operation from starting moh. This also did not affect non-cached realtime moh because the classes are not registered at all. I also have modified res_musiconhold to use the _t_ variants of the ao2_ functions so that more info can be gleaned when attempting to trace the refcounts. I found this to be incredibly helpful for debugging this issue and there's no good reason to remove it. (closes issue #14661) Reported by: sum ........ r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr 2009) | 3 lines Use safe macro practices even though they really aren't necessary. ........ 2009-04-09 18:55 +0000 [r187051-187487] Tilghman Lesher * main/manager.c, /, include/asterisk/linkedlists.h, include/asterisk/lock.h: Merged revisions 187483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines Race condition between ast_cli_command() and 'module unload' could cause a deadlock. Add lock timeouts to avoid this potential deadlock. (closes issue #14705) Reported by: jamessan Patches: 20090320__bug14705.diff.txt uploaded by tilghman (license 14) Tested by: jamessan ........ ................ * /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 | tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines Allow '/' in username portion of register; this is a regression. (closes issue #14668) Reported by: Netview ........ * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions 187363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) | 10 lines Merged revisions 187362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) | 3 lines Permit zero-length text messages in SIP. (Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal") ........ ................ * main/asterisk.c, agi/Makefile, build_tools/cflags.xml, utils/Makefile, include/asterisk.h, /, main/Makefile, main/file.c, main/astfd.c (added): Merged revisions 187302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines Add debugging mode for diagnosing file descriptor leaks. (Related to issue #14625) ........ r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, missed this file in the last commit. ........ ................ * /, funcs/func_odbc.c: Merged revisions 187050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187050 | tilghman | 2009-04-08 12:08:43 -0500 (Wed, 08 Apr 2009) | 7 lines If the first column is empty, output a delimiter anyway. (closes issue #14848) Reported by: john8675309 Patches: 20090408__bug14848.diff.txt uploaded by tilghman (license 14) Tested by: john8675309 ........ 2009-04-08 16:54 +0000 [r186988-187049] Mark Michelson * res/res_musiconhold.c, /: Merged revisions 187046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500 (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines Fix a small logical error when loading moh classes. We were unconditionally incrementing the number of mohclasses registered. However, we should actually only increment if the call to moh_register was successful. While this probably has never caused problems, I noticed it and decided to fix it anyway. ........ ................ * main/channel.c, /: Merged revisions 186985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr 2009) | 30 lines Merged revisions 186984 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines Make a couple of changes with regards to a new message printed in ast_read(). "ast_read() called with no recorded file descriptor" is a new message added after a bug was discovered. Unfortunately, it seems there are a bunch of places that potentially make such calls to ast_read() and trigger this error message to be displayed. This commit does two things to help to make this message appear less. First, the message has been downgraded to a debug level message if dev mode is not enabled. The message means a lot more to developers than it does to end users, and so developers should take an effort to be sure to call ast_read only when a channel is ready to be read from. However, since this doesn't actually cause an error in operation and is not something a user can easily fix, we should not spam their console with these messages. Second, the message has been moved to after the check for any pending masquerades. ast_read() being called with no recorded file descriptor should not interfere with a masquerade taking place. This could be seen as a simple way of resolving issue #14723. However, I still want to try to clear out the existing ways of triggering this message, since I feel that would be a better resolution for the issue. ........ ................ 2009-04-08 12:39 +0000 [r186929] Russell Bryant * /, channels/chan_sip.c: Merged revisions 186928 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186928 | russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines Update some comments and resolve potential memory corruption in chan_sip. While browsing chan_sip the other day, I noticed this dangerous code in dialog_needdestroy(). This function is an ao2_callback. It is absolutely _not_ okay to unlock the container from within this function. It's also not clear why it was useful. Given that it could cause memory corruption, I have removed it. There was also a TODO comment left describing a potential implementation of an improvement to the needdestroy handling. I'm not convinced that what was described is the best choice here, so I have briefly described the way that this function is used today that could be improved. ........ 2009-04-08 05:08 +0000 [r186901] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 | tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines Add lastms to the require API call. ........ 2009-04-08 00:10 +0000 [r186836-186845] Mark Michelson * formats/format_wav_gsm.c, /, formats/format_wav.c: Merged revisions 186842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr 2009) | 14 lines Merged revisions 186841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines Fix a few typos of the word "frequency." (closes issue #14842) Reported by: jvandal Patches: frequency-typo.diff uploaded by jvandal (license 413) ........ ................ * /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 | mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 lines Fix bad merge from fix for issue 13867. (closes issue #14686) Reported by: davidw ........ * main/channel.c, /: Merged revisions 186833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr 2009) | 15 lines Merged revisions 186832 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, warning sounds will not be properly played to either party of the bridge. (closes issue #14845) Reported by: adomjan ........ ................ 2009-04-07 22:33 +0000 [r186807] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 186799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) | 10 lines Merged revisions 186775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines Fix Macro documentation to match current (and intended) behavior. (See -dev mailing list) ........ ................ 2009-04-07 20:59 +0000 [r186723] Mark Michelson * main/manager.c, /: Merged revisions 186720 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr 2009) | 12 lines Merged revisions 186719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines Ensure that \r\n is printed after the ActionID in an OriginateResponse. (closes issue #14847) Reported by: kobaz ........ ................ 2009-04-03 20:21 +0000 [r186469] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later). ........ ................ 2009-04-03 20:05 +0000 [r186449] Tilghman Lesher * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 186444,186447 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) | 14 lines Merged revisions 186415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ ................ r186447 | tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines Merged revisions 186445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) | 2 lines Found a conflict in the last commit, due to multiple targets ........ ................ 2009-04-03 15:56 +0000 [r186324] Joshua Colp * include/asterisk/crypto.h, /: Merged revisions 186321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines Fix a problem with the crypto variable definitions not actually being defined properly. (closes issue #14804) Reported by: jvandal ........ ................ 2009-04-03 15:19 +0000 [r186302] Tilghman Lesher * main/stdtime/localtime.c, /: Merged revisions 186297 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186297 | tilghman | 2009-04-03 10:18:28 -0500 (Fri, 03 Apr 2009) | 4 lines Compatibility fix for glibc 2.4 (Closes issue #14820) Reported by: phsultan ........ 2009-04-03 14:34 +0000 [r186289] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr 2009) | 20 lines Fix the ability to retrieve voicemail messages from IMAP. A recent change made interactive vm_states no longer get added to the list of vm_states and instead get stored in thread-local storage. In trunk and all the 1.6.X branches, the problem is that when we search for messages in a voicemail box, we would attempt to update the appropriate vm_state struct by directly searching in the list of vm_states instead of using the get_vm_state_by_imap_user function. This meant we could not find the interactive vm_state that we wanted. (closes issue #14685) Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........ 2009-04-03 02:11 +0000 [r186233] Russell Bryant * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009) | 29 lines Merged revisions 186229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) | 21 lines Fix a memory leak in cdr_radius. I came across this while doing some testing of my ast_channel_ao2 branch. After running a test overnight that generated over 5 million calls, Asterisk had taken up about 1 GB of my system memory. So, I re-ran the test with MALLOC_DEBUG turned on. However, it showed no leaks in Asterisk during the test, even though Asterisk was still consuming it somehow. Instead, I turned to valgrind, which when run with --leak-check=full, told me exactly where the leak came from, which was from allocations inside the radiusclient-ng library. This explains why MALLOC_DEBUG did not report it. After a bit of analysis, I found that we were leaking a little bit of memory every time a CDR record was passed to cdr_radius. I don't actually have a radius server set up to receive CDR records. However, I always have my development systems compile and install all modules. In addition to making sure there are not build errors across modules, always loading modules helps find bugs like this, too, so it is strongly recommend for all developers. ........ ................ 2009-04-02 22:00 +0000 [r186178] Mark Michelson * configs/features.conf.sample, /: Merged revisions 186175 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines Fix instructions in one-step parking comment to make more sense. Changed a capital K to a lowercase k. ........ ................ 2009-04-02 17:28 +0000 [r186111] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized ........ ................ 2009-04-02 17:14 +0000 [r186022-186063] Tilghman Lesher * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 186060 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ * main/strings.c, /: Merged revisions 186021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186021 | tilghman | 2009-04-02 10:14:22 -0500 (Thu, 02 Apr 2009) | 7 lines Missed a common case for needing to extend the buffer. (closes issue #14716) Reported by: sum Patches: 20090402__bug14716.diff.txt uploaded by tilghman (license 14) Tested by: sum ........ 2009-04-02 13:54 +0000 [r185957] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior. this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized. ........ ................ 2009-04-01 22:44 +0000 [r185947] Tilghman Lesher * include/asterisk/pbx.h, include/asterisk/strings.h, main/taskprocessor.c, res/res_odbc.c, include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c, main/manager.c, /, main/tdd.c, include/asterisk/astobj2.h, main/ast_expr2f.c: Merged revisions 185912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185912 | tilghman | 2009-04-01 15:13:28 -0500 (Wed, 01 Apr 2009) | 4 lines Merge changes from str_substitution that are unrelated to that branch. Included is a small bugfix to an ast_str helper, but most of these changes are simply doxygen fixes. ........ 2009-04-01 13:51 +0000 [r185775] Russell Bryant * main/channel.c, /: Merged revisions 185772 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009) | 14 lines Merged revisions 185771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines Fix a case where DTMF could bypass audiohooks. This change fixes a situation where an audiohook that wants DTMF would not actually get it. This is in the code path where we end DTMF digit length emulation while handling a NULL frame. ........ ................ 2009-03-31 22:38 +0000 [r185667] Kevin P. Fleming * utils, /: Merged revisions 185664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 | kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line ignore copied (generated) file ........ 2009-03-31 22:13 +0000 [r185472-185605] Mark Michelson * apps/app_queue.c, /: Merged revisions 185604 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185604 | mmichelson | 2009-03-31 17:12:52 -0500 (Tue, 31 Mar 2009) | 3 lines Fix trunk's compilation. ........ * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar 2009) | 12 lines Merged revisions 185599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar 2009) | 6 lines Fix crash that would occur if an empty member was specified in queues.conf. (closes issue #14796) Reported by: pida ........ ................ * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500 (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar 2009) | 8 lines Fix Russian voicemail intro to say the word "messages" properly. (closes issue #14736) Reported by: chappell Patches: voicemail_no_messages.diff uploaded by chappell (license 8) ........ ................ 2009-03-31 17:51 +0000 [r185428] David Brooks * channels/chan_gtalk.c, /: Merged revisions 185363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: Review: http://reviewboard.digium.com/r/181/ ........ ................ 2009-03-31 14:59 +0000 [r185264] Russell Bryant * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 | russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines Don't free() an astobj2 object. (closes issue #14672) Reported by: makoto ........ 2009-03-31 14:11 +0000 [r185200] Joshua Colp * main/audiohook.c, /: Merged revisions 185197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) | 15 lines Merged revisions 185196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines Fix crash when moving audiohooks between channels. Handle the scenario where we are called to move audiohooks between channels and the source channel does not actually have any on it. (closes issue #14734) Reported by: corruptor ........ ................ 2009-03-30 20:52 +0000 [r185128-185129] Richard Mudgett * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged revisions 185123 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines Make chan_misdn BRI TE side normally defer channel selection to the NT side. Channel allocation collisions are not handled by chan_misdn very well. This patch simply avoids the problem for BRI only. For PRI, allocation collisions are still possible but less likely since there are simply more channels available and each end could use a different allocation strategy. misdn.conf options available: te_choose_channel - Use to force the TE side to allocate channels. method - Specify the channel allocation strategy. (closes issue #13488) Reported by: Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, festr ........ ................ 2009-03-30 16:52 +0000 [r185089] Mark Michelson * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar 2009) | 45 lines Merged revisions 185031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked. (This is copied and pasted from the review request I made for this patch) Asterisk has some odd behavior when queue weights are used. The current logic used when potentially calling a queue member is: If the member we are going to call is part of another queue and _that other queue has any callers in it_ and has a higher weight than the queue we are calling from, then don't try to contact that member. The issue here is what I have marked with underscores. If the higher-weighted queue has any callers in it at all, then the queue member will be unreachable from the lower-weighted queue. This has the potential to be really really bad if using a queue strategy, such as leastrecent or fewestcalls, with the potential to call the same member repeatedly. The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works well for this situation. With this set of changes, the logic used becomes: If the member we are going to call is part of another queue, the other queue has a higher weight than the queue we are calling from, and the higher weight queue has at least as many callers as available members, then do not try to contact the queue member. If the higher weighted queue has fewer callers than available members, then there is no reason to deny the call to this member since the other queue can afford to spare a member. Since the fix involved writing a generic function for determining the number of available members in the queue, I also modified the is_our_turn function to make use of the new num_available_members function to determine if it is our turn to try calling a member. There is one small behavior change. Before writing this patch, if you had autofill disabled, then if you were the head caller in a queue, you would automatically be told that it was your turn to try calling a member. This did not take into account whether there were actually any queue members available to take the call. Now we actually make sure there is at least one member available to take the call if autofill is disabled. (closes issue #13220) Reported by: garychen Review: http://reviewboard.digium.com/r/202/ ........ ................ 2009-03-30 14:43 +0000 [r184951] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines Merged revisions 184947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines Improve our handling of T38 in the initial INVITE from a device. We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ ........ ................ 2009-03-30 13:57 +0000 [r184913] Russell Bryant * channels/h323/Makefile.in, /: Merged revisions 184910 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) | 4 lines Fix build error when chan_h323 is not being built. (reported by cai1982 in #asterisk-dev) ........ 2009-03-29 05:56 +0000 [r184839-184846] Russell Bryant * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009) | 13 lines Merged revisions 184842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines Ensure targs variable is fully initialized. (closes issue #14758) Reported by: tim_ringenbach ........ ................ * channels/Makefile, /: Merged revisions 184838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines Simplify chan_h323 build to not require a second run of "make". (closes issue #14715) Reported by: jthurman Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614) Tested by: tzafrir, russell ........ 2009-03-27 19:21 +0000 [r184779] Kevin P. Fleming * channels/chan_iax2.c, main/timing.c, main/channel.c, /, bridges/bridge_softmix.c, include/asterisk/timing.h, include/asterisk/channel.h: Merged revisions 184762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines Improve timing interface to remember which provider provided a timer The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ ........ 2009-03-27 18:12 +0000 [r184707-184729] Russell Bryant * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure we use the best RNG available. ........ * apps/app_queue.c, apps/app_voicemail.c, main/cli.c, include/asterisk/app.h, /, apps/app_dumpchan.c, main/app.c: Merged revisions 184693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184693 | russell | 2009-03-27 11:21:10 -0500 (Fri, 27 Mar 2009) | 2 lines Change global_app_buf to ast_str_thread_global_buf. ........ 2009-03-27 15:58 +0000 [r184650-184678] Joshua Colp * /, bridges/bridge_softmix.c: Merged revisions 184677 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184677 | file | 2009-03-27 12:57:28 -0300 (Fri, 27 Mar 2009) | 7 lines Fix a potential timer leak in bridge_softmix. It is possible for a bridge to be created without actually being used. In that scenario a timing file descriptor would be opened and not closed. To fix this the timing file descriptor is now closed in the destroy callback, not the thread function. ........ * /, res/res_agi.c: Merged revisions 184673 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 | file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix speech structure leak in the AGI speech recognition integration. The AGI dialplan applications did not destroy the speech structure automatically if it was not destroyed by the running AGI script. They will now do this. (issue LUMENVOX-15) ........ * /, bridges/bridge_softmix.c: Merged revisions 184639 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184639 | file | 2009-03-27 11:18:40 -0300 (Fri, 27 Mar 2009) | 2 lines Remove a cast that is not needed. ........ 2009-03-27 14:09 +0000 [r184632] Russell Bryant * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /, res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions 184630 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 | russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines Change g_eid to ast_eid_default. ........ 2009-03-27 13:59 +0000 [r184612-184629] Joshua Colp * /, bridges/bridge_softmix.c: Merged revisions 184628 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184628 | file | 2009-03-27 10:57:29 -0300 (Fri, 27 Mar 2009) | 6 lines Fix a potential race condition when creating a software based mixing bridge. It was possible for no timer to become available between creating the bridge and starting it. We now open a timer when creating it and keep it open until the bridge is destroyed. ........ * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | 16 lines Merged revisions 184565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls. If calls were placed using an IP address or hostname the global nat setting was copied over but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP actions. (closes issue #14546) Reported by: acunningham ........ ................ 2009-03-27 02:35 +0000 [r184514-184552] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009) | 20 lines Fix some issues with rwlock corruption that caused deadlock like symptoms. When dvossel and I were doing some load testing last week, we noticed that we could make Asterisk trunk lock up instantly when we started generating a bunch of calls. The backtraces of locked threads were bizarre, and many were stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a number of places where a backtrace would be loaded into an invalid index of the backtrace array. It's an off by one error, which ends up writing over the rwlock itself. 2) Ensure that in the array of held locks, we NULL out an index once it is not being used so that it's not confusing when analyzing its contents. 3) Remove a bunch of logging referring to an rwlock operating being done with "deep reentrancy". It is normal for _many_ threads to hold a read lock on an rwlock. ........ * /, main/file.c: Merged revisions 184515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 | russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines Don't act surprised if we get a -1 indication. ........ * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26 Mar 2009) | 2 lines Pass more useful information through to lock tracking when DEBUG_THREADS is on. ........ 2009-03-26 22:19 +0000 [r184454] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 184448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar 2009) | 9 lines Merged revisions 184447 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar 2009) | 3 lines use new, improved 8kHz prompts ........ ................ 2009-03-25 22:15 +0000 [r184343-184346] Russell Bryant * /, main/event.c: Merged revisions 184344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 | russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines Remove unneeded AST_LIST_ENTRY() and comment on the purpose of ast_event_ref. ........ * include/asterisk/_private.h, channels/chan_iax2.c, channels/chan_dahdi.c, include/asterisk/event.h, apps/app_minivm.c, res/ais/evt.c, main/event.c, include/asterisk/strings.h, main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c, channels/chan_unistim.c, include/asterisk/devicestate.h, /, channels/chan_sip.c, main/devicestate.c: Merged revisions 184339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) | 35 lines Improve performance of the ast_event cache functionality. This code comes from svn/asterisk/team/russell/event_performance/. Here is a summary of the changes that have been made, in order of both invasiveness and performance impact, from smallest to largest. 1) Asterisk 1.6.1 introduces some additional logic to be able to handle distributed device state. This functionality comes at a cost. One relatively minor change in this patch is that the extra processing required for distributed device state is now completely bypassed if it's not needed. 2) One of the things that I noticed when profiling this code was that a _lot_ of time was spent doing string comparisons. I changed the way strings are represented in an event to include a hash value at the front. So, before doing a string comparison, we do an integer comparison on the hash. 3) Finally, the code that handles the event cache has been re-written. I tried to do this in a such a way that it had minimal impact on the API. I did have to change one API call, though - ast_event_queue_and_cache(). However, the way it works now is nicer, IMO. Each type of event that can be cached (MWI, device state) has its own hash table and rules for hashing and comparing objects. This by far made the biggest impact on performance. For additional details regarding this code and how it was tested, please see the review request. (closes issue #14738) Reported by: russell Review: http://reviewboard.digium.com/r/205/ ........ 2009-03-25 19:27 +0000 [r184266-184283] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 | file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix issue with a T38 reinvite being sent even if not configured to do so. If we receive a T38 request negotiate control frame we should only attempt to do so if the option is enabled on the dialog. ........ * main/bridging.c, /: Merged revisions 183652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183652 | file | 2009-03-22 18:00:28 -0300 (Sun, 22 Mar 2009) | 4 lines Fix a minor logic flaw with the bridge generic thread. We only want to move the channel pointers that are actually present. ........ 2009-03-25 15:33 +0000 [r184256] Eliel C. Sardanons * main/asterisk.c, /: Merged revisions 184220 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) | 19 lines Merged revisions 184188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. When moving the cursor backward and pressing TAB to autocomplete, a NULL is put in the line and we are loosing what we have already wrote after the actual cursor position. (closes issue #14373) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) Tested by: lmadsen ........ ................ 2009-03-25 14:40 +0000 [r184150-184221] Russell Bryant * main/timing.c, /: Merged revisions 184219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184219 | russell | 2009-03-25 09:33:32 -0500 (Wed, 25 Mar 2009) | 2 lines Include poll-compat.h ........ * main/timing.c, /: Merged revisions 184151 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184151 | russell | 2009-03-24 21:03:13 -0500 (Tue, 24 Mar 2009) | 2 lines Change poll() to ast_poll(). ........ * utils/Makefile, /, include/asterisk/compat.h: Merged revisions 184147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 | russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines Fix build issues on Mac OSX. (closes issue #14714) Reported by: ygor ........ 2009-03-24 22:42 +0000 [r184082] Mark Michelson * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar 2009) | 15 lines Merged revisions 184078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. ........ ................ 2009-03-24 22:02 +0000 [r184041-184044] Russell Bryant * main/channel.c, /: Merged revisions 184043 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184043 | russell | 2009-03-24 17:00:58 -0500 (Tue, 24 Mar 2009) | 2 lines Put siren7 and siren14 in ast_best_codec() just so they're in there somewhere. ........ * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low and =medium The default codec configuration for chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as the codec in some test calls, but that no longer happens after this change. ........ 2009-03-24 15:29 +0000 [r183868-183917] Tilghman Lesher * /, configs/voicemail.conf.sample: Merged revisions 183914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500 (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ ................ * /, main/http.c: Merged revisions 183865 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 | tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines Allow browsers to cache images and other static content. (This is a regression over 1.4) ........ 2009-03-23 19:00 +0000 [r183769] Mark Michelson * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar 2009) | 13 lines Merged revisions 183700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines Fix a memory leak in res_monitor.c The only way that this leak would occur is if Monitor were started using the Manager interface and no File: header were given. Discovered while reviewing the ast_channel_ao2 review request. ........ ................ 2009-03-23 18:12 +0000 [r183704] Leif Madsen * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) | 7 lines Fixes a documentation error introduced during the CLI cleanup at AstriDevCon 2008. (closes issue #14655) Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728) Tested by: lmadsen ........ 2009-03-20 17:09 +0000 [r183564] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183560 | russell | 2009-03-20 12:00:58 -0500 (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines Fix a crash in IAX2 registration handling found during load testing with dvossel. ........ ................ 2009-03-20 12:19 +0000 [r183519] Eliel C. Sardanons * channels/chan_dahdi.c, /: Merged revisions 183511 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) | 2 lines Remove duplicate inside the xml documentation. ........ 2009-03-19 19:20 +0000 [r183337] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500 (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines Delay signalling progress until a PRI channel really signals progress. (closes issue #13034) Reported by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by tilghman (license 14) patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ 2009-03-19 18:20 +0000 [r183263] Russell Bryant * main/loader.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 183242 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines Merged revisions 183241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ ................ 2009-03-19 18:12 +0000 [r183247] Mark Michelson * apps/app_queue.c, /: Merged revisions 183244 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 | mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 lines Fix a memory leak associated with queues. For every attempt that app_queue made to place an outbound call to a queue member, we would allocate a queue_end_bridge structure. When the bridge for the call had completed, we would free the structure. Unfortunately not all call attempts actually end up bridged to a member, so we need to be more selective of when to allocate the structure. With this change, the allocation occurs in an area where we can guarantee that the call will be bridged. (closes issue #14680) Reported by: caspy Patches: 14680.patch uploaded by mmichelson (license 60) Tested by: caspy ........ 2009-03-19 Leif Madsen * Release Asterisk 1.6.2.0-beta1 2009-03-19 16:11 +0000 [r183122] Mark Michelson * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 ........ ................ 2009-03-19 15:45 +0000 [r183068-183111] Joshua Colp * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ ........ * main/channel.c, /: Merged revisions 183057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. ........ 2009-03-18 21:19 +0000 [r183031] Jeff Peeler * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ 2009-03-18 14:39 +0000 [r182947] Russell Bryant * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c, configure, apps/app_mp3.c, res/res_agi.c, include/asterisk/poll-compat.h, channels/chan_alsa.c, main/asterisk.c, apps/app_nbscat.c, /, main/Makefile, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/io.h, main/utils.c, include/asterisk/channel.h: Merged revisions 182847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ 2009-03-17 20:53 +0000 [r182725] Jeff Peeler * channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /, channels/h323/ast_h323.cxx, configure, autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h, channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions 182722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ 2009-03-17 16:46 +0000 [r182592] Russell Bryant * main/channel.c, /: Merged revisions 182553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). ........ 2009-03-17 15:01 +0000 [r182528-182534] Kevin P. Fleming * main/channel.c, /: Merged revisions 182530 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 lines correct logic flaw in ast_answer() changes in r182525 ........ * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 182525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ 2009-03-17 05:54 +0000 [r182453] Tilghman Lesher * main/db.c, /: Merged revisions 182450 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) | 14 lines Merged revisions 182449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ ................ 2009-03-17 02:02 +0000 [r182409] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 182408 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) | 8 lines OPENR2 uses an incorrect string value if the extension delimiter is not present. * Fixed OPENR2 using an incorrect string value if the extension delimiter is not present in the Dial() function. This was fixed for SS7 and PRI in trunk -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others. * Removed trailing whitespace that appeared with OPENR2. ........ 2009-03-16 20:51 +0000 [r182360-182361] Russell Bryant * /: svnmerge init * / (added): Create a branch for 1.6.2 2009-03-16 20:35 +0000 [r182355] Russell Bryant * CREDITS, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, include/asterisk/autoconfig.h.in, configure.ac, CHANGES, makeopts.in: Add MFC/R2 support for chan_dahdi. This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ 2009-03-16 17:49 +0000 [r182282] David Vossel * /, channels/chan_iax2.c: Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ 2009-03-16 17:33 +0000 [r182211-182278] Tilghman Lesher * funcs/func_env.c: Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution. Previously, FILE() returned one less character than specified, due to the terminating NULL. Both the offset and length parameters now behave identically to the way variable substitution offsets and lengths also work. (closes issue #14670) Reported by: BMC * channels/chan_local.c, /: Merged revisions 182208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ 2009-03-16 13:58 +0000 [r182171] Joshua Colp * main/channel.c: Fix a memory leak in the ast_answer / __ast_answer API call. For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. 2009-03-13 21:26 +0000 [r182029-182121] Mark Michelson * apps/app_queue.c: Change faulty comparison used when announcing average hold minutes and seconds (closes issue #14227) Reported by: caspy * main/features.c: Remove ast_ prefix from functions which are not public. * /, main/features.c: Merged revisions 181990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ 2009-03-13 17:25 +0000 [r182022] Joshua Colp * channels/chan_sip.c: Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) 2009-03-13 16:55 +0000 [r181985] Kevin P. Fleming * channels/chan_sip.c: improve a bit of suboptimal code 2009-03-13 01:26 +0000 [r181899] Richard Mudgett * /: Merged revisions 181898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Just recording the v1.4 change in trunk since it originally came from here. ........ r181898 | rmudgett | 2009-03-12 20:19:29 -0500 (Thu, 12 Mar 2009) | 4 lines Use the correct branch integrated property when generating the version string. Copied the make_version file from Asterisk trunk. ........ 2009-03-12 21:43 +0000 [r181769-181846] Mark Michelson * apps/app_queue.c: Run the macro on the queue member's channel when he answers, not the caller's channel. * /, channels/chan_sip.c: Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ 2009-03-12 17:32 +0000 [r181731] Tilghman Lesher * main/translate.c: Adjust translation table column widths based upon the translation times. Previously, only 5 columns were displayed, and if a translation time exceeded 99,999 useconds, it would be displayed as 0, instead of its actual time. (closes issue #14532) Reported by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman (license 14) Tested by: pj 2009-03-12 16:56 +0000 [r181612-181665] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 181664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 lines Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. ........ * /, res/res_musiconhold.c: Merged revisions 181659-181660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in previous commit. ........ * /, res/res_musiconhold.c: Merged revisions 181655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy ........ * apps/app_dial.c: Fix crash when sleep and retries argument was not given to RetryDial application. (closes issue #14647) Reported by: sherpya 2009-03-12 01:33 +0000 [r181542-181577] Richard Mudgett * build_tools/make_version: Whitespace chages. * build_tools/make_version: Use the correct branch integrated property when generating the version string 2009-03-11 23:14 +0000 [r181499] Michiel van Baak * configs/sip.conf.sample: Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk 2009-03-11 22:25 +0000 [r181465] Russell Bryant * main/channel.c: Make handling of the BRIDGE_PLAY_SOUND variable thread-safe. 2009-03-11 22:20 +0000 [r181444] Jason Parker * /, configure, configure.ac: Merged revisions 181436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | 4 lines Allow prefix to set localstatedir (when used and different from the default). This is similar to the /etc change that was made for the non-FreeBSD case. ........ 2009-03-11 22:14 +0000 [r181424-181428] Russell Bryant * main/channel.c: Make handling of the BRIDGEPVTCALLID variable thread-safe. * main/channel.c, /: Merged revisions 181423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ 2009-03-11 17:34 +0000 [r181371] David Vossel * channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions 181340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ 2009-03-11 17:26 +0000 [r181345] Joshua Colp * /, channels/chan_sip.c: Merged revisions 181328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ 2009-03-11 17:04 +0000 [r181301] Tilghman Lesher * include/asterisk/astobj2.h: Turn off malloc debugging of astobj2, since it apparently doesn't work too well during startup. 2009-03-11 16:40 +0000 [r181296] Joshua Colp * /, channels/chan_sip.c: Merged revisions 181295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ 2009-03-11 16:19 +0000 [r181292] Russell Bryant * doc/google-soc2009-ideas.txt: Replace contents of this doc with a pointer to its new home 2009-03-11 14:28 +0000 [r181244] Mark Michelson * apps/app_queue.c: Fix segfault when dialing a typo'd queue If trying to dial a non-existent queue, there would be a segfault when attempting to access q->weight, even though q was NULL. This problem was introduced during the queue-reset merge and thus only affects trunk. (closes issue #14643) Reported by: alecdavis 2009-03-11 13:44 +0000 [r181210] Joshua Colp * apps/app_confbridge.c: Don't play the "you are about to be placed into the conference" and "the leader has left the conference" sounds if the quiet option is enabled. (reported by Vadim Lebedev on the asterisk-dev list) 2009-03-11 04:06 +0000 [r181135] Jeff Peeler * utils/Makefile, include/asterisk/utils.h, include/asterisk/astmm.h, channels/chan_sip.c, channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c, pbx/pbx_config.c: Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj 2009-03-11 02:25 +0000 [r181099] Russell Bryant * doc/google-soc2009-ideas.txt: tabs to spaces 2009-03-11 00:49 +0000 [r181032-181033] Mark Michelson * channels/chan_sip.c: Add missing comment that quotes RFC 3891 * /, channels/chan_sip.c: Merged revisions 181029,181031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines Remove unused variables. ........ 2009-03-11 00:29 +0000 [r181027-181028] Tilghman Lesher * main/strings.c, main/hashtab.c, include/asterisk/astobj2.h, main/heap.c, include/asterisk/strings.h, include/asterisk/hashtab.h, main/astobj2.c, include/asterisk/heap.h: Add MALLOC_DEBUG to various utility APIs, so that memory leaks can be tracked back to their source. (related to issue #14636) * main/pbx.c: Spacing changes only 2009-03-10 22:03 +0000 [r180944] Jason Parker * /, configure, configure.ac, autoconf/ast_prog_sed.m4, autoconf/ast_check_gnu_make.m4: Merged revisions 180941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) | 1 line Make things happier when using autoconf 2.62+ ........ 2009-03-10 22:03 +0000 [r180935-180942] Russell Bryant * doc/google-soc2009-ideas.txt: Add some notes on getting in contact with the dev community * doc/google-soc2009-ideas.txt: Remove difficulty and language specifiers * doc/google-soc2009-ideas.txt: Expand upon documentation of manager event project 2009-03-10 21:15 +0000 [r180898] Michiel van Baak * CHANGES: list the move of the astvarrundir from /var/run to /var/run/asterisk (actually its $(localstatedir)/run/asterisk Makes setups with asterisk as non-root easier to manage because you can setup permissions on this dir instead of touching a file and setting permissions on that. Files that come to mind are asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell 2009-03-10 19:36 +0000 [r180859-180862] Russell Bryant * doc/google-soc2009-ideas.txt: add more projects * doc/google-soc2009-ideas.txt: add more project ideas 2009-03-10 14:40 +0000 [r180800] Joshua Colp * main/manager.c: Reset the thread local string buffer when handling the UserEvent action. (closes issue #14593) Reported by: JimDickenson 2009-03-09 22:00 +0000 [r180750] Russell Bryant * doc/google-soc2009-ideas.txt: Add current mentors list, and first pass on a project list broken out of "PineMango" I will work on adding projects that have been sent to be via email tomorrow. 2009-03-09 20:58 +0000 [r180719] Jeff Peeler * include/asterisk/rtp.h, include/asterisk/extconf.h, main/devicestate.c, include/asterisk/tcptls.h, main/enum.c, include/asterisk/callerid.h, include/asterisk/doxyref.h, include/asterisk/event.h, include/asterisk/audiohook.h, include/asterisk/dsp.h, include/asterisk/timing.h, include/asterisk/udptl.h, include/asterisk/dlinkedlists.h, include/asterisk/utils.h, include/asterisk/devicestate.h, include/asterisk/taskprocessor.h, include/asterisk/enum.h, include/asterisk/astobj2.h, include/asterisk/config.h, include/asterisk/channel.h, include/asterisk/manager.h, include/asterisk/heap.h, include/asterisk/logger.h, include/asterisk/http.h, include/asterisk/res_odbc.h, include/asterisk/app.h, main/tcptls.c, include/asterisk/linkedlists.h, include/asterisk/sched.h, include/asterisk/datastore.h, include/asterisk/lock.h, include/asterisk/pbx.h, include/asterisk/dnsmgr.h: Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 Copied from my review board description: This is a continuation of the API changes documentation started for describing changes between releases. Most of the API changes were pretty simple needing only to be brought to attention via the new "Asterisk API Changes" list. However, if you see anything that needs further explanation feel free to supplement what is there. The current method of documenting is to add (in the header file): \version and then to add the function to the change list in doxyref.h on the AstAPIChanges page. I also made sure all the functions that were newly added were tagged with \since 1.6.1. I think this is a good habit to start both for the historical aspect as well as for the future ability to easily add a "New Asterisk API" page. Review: http://reviewboard.digium.com/r/190/ 2009-03-09 14:14 +0000 [r180684] Russell Bryant * doc/google-soc2009-ideas.txt (added): Add skeleton for GSoC ideas list 2009-03-07 15:36 +0000 [r180641] Russell Bryant * contrib/asterisk-ng-doxygen: Make some minor updates to the doxygen configuration - add bridges directory to be processed - add some res/ subdirs - alphabetize subdirs - use consistent indentation 2009-03-06 18:25 +0000 [r180579] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 180567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when IMAP storage is enabled. ........ 2009-03-06 17:26 +0000 [r180534] David Vossel * /, main/enum.c: Merged revisions 180532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ ........ 2009-03-05 23:26 +0000 [r180383-180465] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 180464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ * /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ 2009-03-05 19:05 +0000 [r180382] Michiel van Baak * Makefile: Make sure we terminate the first s| command so we can actually produce correct files. 2009-03-05 18:29 +0000 [r180373] Kevin P. Fleming * main/frame.c, /, include/asterisk/frame.h, main/rtp.c: Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ 2009-03-05 18:18 +0000 [r180369] Joshua Colp * channels/chan_bridge.c (added), main/Makefile, bridges/bridge_simple.c, bridges/bridge_softmix.c, include/asterisk/channel.h, bridges/bridge_multiplexed.c, CHANGES, Makefile, include/asterisk/bridging_technology.h (added), bridges (added), bridges/bridge_builtin_features.c, include/asterisk/bridging_features.h (added), include/asterisk/bridging.h (added), apps/app_confbridge.c (added), main/bridging.c (added), bridges/Makefile: Merge phase 1 support for the new bridging architecture. This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ 2009-03-05 06:21 +0000 [r180304-180334] Tilghman Lesher * contrib/editors/asterisk.vim: Also highlight the preamble and postamble * contrib/editors/ael.vim (added), contrib/editors/asterisk.vim (added), contrib/editors (added), contrib/editors/asteriskvm.vim (added): Add syntax coloring files for Vim, including a new one for AEL 2009-03-04 21:01 +0000 [r180261] Russell Bryant * channels/chan_sip.c: Resolve object matching issues related to the removal of the sip_user object. Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ 2009-03-04 20:48 +0000 [r180259] Tilghman Lesher * main/aescrypt.c, main/abstract_jb.c, main/acl.c, main/app.c, main/alaw.c: Spacing changes only 2009-03-04 19:24 +0000 [r180195] Joshua Colp * /, main/callerid.c: Merged revisions 180194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 lines Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) ........ 2009-03-04 17:03 +0000 [r180155] Mark Michelson * channels/chan_sip.c, configs/sip.conf.sample: Allow for "magic" pickups to work when we wish to ignore the context When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis 2009-03-04 14:39 +0000 [r180120] Joshua Colp * apps/app_dial.c: Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) 2009-03-03 23:35 +0000 [r180079] Steve Murphy * utils/Makefile: My bad! left check_expr2 in the ALL_UTILS list by mistake. Already done to 1.6.x 2009-03-03 23:21 +0000 [r180032] David Vossel * main/channel.c, include/asterisk/app.h, apps/app_read.c, main/app.c: app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ 2009-03-03 22:49 +0000 [r180007] Mark Michelson * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ 2009-03-03 22:12 +0000 [r179973] Steve Murphy * utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c, main/ast_expr2.fl, main/ast_expr2.c: Merged revisions 179807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some work to do to port these changes to trunk; the check_expr stuff hasn't been updated here for quite some time, it appears. I added some more tests to the check_expr2 suite. I had to play around with the makefile a bit, etc. I added STANDALONE2 #ifdefs to ast_expr2.y so as not to conflict structure with aelparse. ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. ........ 2009-03-03 22:01 +0000 [r179972] David Vossel * apps/app_meetme.c: app_meetme not setting filename and fileformat correctly for realtime When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set. Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. (closes issue #14545) Reported by: dalbaech Patches: app_meetme-realtime5.patch uploaded by dvossel (license 671) Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705) Tested by: dvossel, dalbaech Review: http://reviewboard.digium.com/r/180/ 2009-03-03 20:59 +0000 [r179937] Mark Michelson * res/res_timing_dahdi.c, doc/timing.txt (added): Add documentation for timing modules used in Asterisk This document specifies the timing modules available in Asterisk beginning with Asterisk 1.6.1. The document goes into detail about the differences between each and gives a general overview of what timing is used for in Asterisk. There is also a section which can be used to help customize your setup or to troubleshoot timing issues you may have. I also added messages to the DAHDI timing test used in res_timing_dahdi.c that points to this new documentation if people experience problems. Big thanks to all who contributed comments on this. (closes issue #14490) Reported by: mmichelson Patches: timing.txt uploaded by mmichelson (license 60) Review: http://reviewboard.digium.com/r/164/ 2009-03-03 20:02 +0000 [r179903] Brian Degenhardt * apps/app_directed_pickup.c: fix a leaked channel lock (and future deadlock) when we try to pick up our own channel 2009-03-03 18:28 +0000 [r179841] Joshua Colp * /, main/features.c: Merged revisions 179840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ 2009-03-03 17:03 +0000 [r179745] Mark Michelson * pbx/pbx_spool.c: Convert pbx_spool to use string fields instead of statically-sized buffers. In tests run after making this conversion, I noticed an approximate 85% reduction in memory usage for call file processing. Review: http://reviewboard.digium.com/r/168/ 2009-03-03 16:47 +0000 [r179742] Russell Bryant * main/channel.c, /: Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ 2009-03-03 15:13 +0000 [r179675] Olle Johansson * channels/chan_sip.c: Please prefix default values with DEFAULT 2009-03-03 14:40 +0000 [r179672] Joshua Colp * main/channel.c, /: Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines Move where fdno is set to the default value to *after* the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ 2009-03-03 13:54 +0000 [r179609] Russell Bryant * main/channel.c, /: Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ 2009-03-03 00:01 +0000 [r179537] Jeff Peeler * main/channel.c, /: Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ 2009-03-02 23:36 +0000 [r179533] Russell Bryant * /, apps/app_meetme.c: Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ 2009-03-02 23:10 +0000 [r179469] Tilghman Lesher * /, main/app.c: Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ 2009-03-02 23:06 +0000 [r179462-179465] Russell Bryant * res/res_timing_timerfd.c: Fix a reference leak in timerfd_set_rate(). (found during a debugging session with dvossel and mmichelson.) * main/channel.c, /: Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ 2009-03-02 20:16 +0000 [r179396] Jason Parker * /, main/editline/configure, main/editline/np/unvis.c, main/editline/sys.h, main/editline/configure.in: Merged revisions 179395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ 2009-03-02 17:18 +0000 [r179361] Tilghman Lesher * cdr/cdr_sqlite3_custom.c: Backport 1.6.0 fix to trunk (failsafe if db is not loaded) 2009-03-02 14:28 +0000 [r179291-179323] Joshua Colp * channels/chan_iax2.c: Do not try to remove a registration scheduled item if the scheduler context has already been destroyed. (closes issue #14580) Reported by: alecdavis * main/audiohook.c: Fix issue where changing the volume of both directions of audio did not work. (closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) 2009-03-01 23:25 +0000 [r179219-179254] Mark Michelson * apps/app_speech_utils.c: Swap reversed timevals. This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! * channels/chan_sip.c: Properly free memory and remove scheduler entries when a transmission failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis 2009-02-27 21:47 +0000 [r179164] Russell Bryant * res/res_ais.c, doc/distributed_devstate.txt, configs/ais.conf.sample: Mark res_ais as experimental, as the binary event format is subject to change. 2009-02-27 21:32 +0000 [r179161] Tilghman Lesher * cdr/cdr_sqlite3_custom.c: If config file is blank, don't load module. (Closes issue #14563) 2009-02-27 21:23 +0000 [r179154] Russell Bryant * UPGRADE.txt: Add a note about the ordering of entries in sip.conf in 1.6.1. 2009-02-27 20:34 +0000 [r179122] Michiel van Baak * channels/chan_skinny.c: Add reload support to chan_skinny. Special thanks goes to DEA who had to redo this patch twice because we first put unload/load support in and later redid the way we configure devices and lines. (closes issue #10297) Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA (license 3) With mods by me based on feedback from wedhorn and Russell and seanbright Tested by: DEA, mvanbaak, pj Review: http://reviewboard.digium.com/r/130/ 2009-02-27 19:04 +0000 [r179057] Jason Parker * doc/tex/channelvariables.tex: Update documentation for DIALEDTIME and ANSWEREDTIME variables. (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) 2009-02-27 15:51 +0000 [r179021] Russell Bryant * sounds/Makefile: Fix downloading SIREN7 and SIREN14 sound packages. In passing, also fix downloading SLIN16 extra sound packages. (closes issue #14565) Reported by: jtodd 2009-02-27 03:45 +0000 [r178986] Steve Murphy * /, main/features.c, configs/features.conf.sample: Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ 2009-02-26 18:41 +0000 [r178919] Tilghman Lesher * main/features.c, CHANGES, configs/features.conf.sample: Sound confirmation of call pickup success. (closes issue #13826) Reported by: azielke Patches: pickupsound2-trunk.patch uploaded by azielke (license 548) __20081124_bug_13826_updated.patch uploaded by lmadsen (license 10) Tested by: lmadsen 2009-02-26 17:46 +0000 [r178871] David Vossel * channels/chan_iax2.c: IAX2 prune realtime, minor tweak to last fix A return statement was missing which caused unexpected cli output. issue #14479 2009-02-26 17:45 +0000 [r178828-178870] Steve Murphy * apps/app_osplookup.c, apps/app_rpt.c: These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x. * /, main/features.c: Merged revisions 178804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom ........ 2009-02-26 16:42 +0000 [r178801] Joshua Colp * main/file.c: Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed. (closes issue #14541) Reported by: grant 2009-02-26 15:50 +0000 [r178767] David Vossel * channels/chan_iax2.c: IAX2 prune realtime fix Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ 2009-02-26 15:40 +0000 [r178764] Joshua Colp * main/indications.c: Ensure there is a valid tone part before trying to play tones. (closes issue #14558) Reported by: alecdavis 2009-02-26 15:02 +0000 [r178733] Olle Johansson * configs/res_snmp.conf.sample: Clarifications on the different models and reference to further docs. 2009-02-26 13:39 +0000 [r178703-178704] Kevin P. Fleming * README: another minor commit to test post-commit script changes (now testing post-revprop-change as well, third try) * README: minor commit to test post-commit script changes 2009-02-25 19:49 +0000 [r178573-178607] Tilghman Lesher * main/stdtime/localtime.c: Picky, picky buildbots * configure, include/asterisk/autoconfig.h.in, configure.ac, main/stdtime/localtime.c: Use notification when timezone files change and re-scan then. (closes issue #14300) Reported by: jamessan Patches: 20090127__bug14300.diff.txt uploaded by tilghman (license 14) 20090224__bug14300.diff uploaded by jamessan (license 246) Tested by: jamessan Review: http://reviewboard.digium.com/r/136/ * res/res_odbc.c: Oops, wrong direction of command 2009-02-25 12:45 +0000 [r178509] Russell Bryant * /, main/asterisk.c: Merged revisions 178508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines Update the copyright year for the main page of the doxygen documentation. ........ 2009-02-24 23:27 +0000 [r178375-178446] Tilghman Lesher * /, configs/extensions.conf.sample: Merged revisions 178445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines Add section about the #exec command in configuration files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) ........ * main/asterisk.c: Apparently, a void cast doesn't override warn_unused_result. * main/asterisk.c: The 3 possible errors with pipe(2) are all impossible in this situation. 2009-02-24 20:39 +0000 [r178374] Russell Bryant * /, main/rtp.c: Merged revisions 178373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell ........ 2009-02-24 20:06 +0000 [r178303-178342] Tilghman Lesher * utils/astcanary.c, main/asterisk.c: Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init. * utils/astcanary.c: Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary. Also, add some documentation supporting the use of astcanary. (closes issue #14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545) 2009-02-24 17:42 +0000 [r178300] David Vossel * doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Allows manager command to see if IAX link is trunked and encrypted. Displays what kind of encryption is enabled as well. Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not. (closes issue #14427) Reported by: snuffy Patches: iax_show_trunks.diff uploaded by snuffy (license 35) 2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review: http://reviewboard.digium.com/r/173/ 2009-02-24 15:18 +0000 [r178213] Joshua Colp * /, channels/chan_sip.c: Merged revisions 178205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ 2009-02-23 23:11 +0000 [r178142] Russell Bryant * /, main/rtp.c: Merged revisions 178141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras ........ 2009-02-23 21:02 +0000 [r178107] Tilghman Lesher * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Permit emailsubject and emailbody to be set per mailbox. (closes issue #14372) Reported by: fhackenberger Patches: voicemail_individual_subject_and_body_1.6.1 uploaded by fhackenberger (license 592) with additional fixes by Corydon76 (license 14) 2009-02-23 18:23 +0000 [r178061] Michiel van Baak * channels/chan_skinny.c: update the new manager commands in chan_skinny to match chan_sip's headers. requested by oej. 2009-02-23 17:59 +0000 [r178030] David Vossel * channels/chan_iax2.c: Changes the way keyrotation is enabled by default Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled. (closes issue #14523) Reported by: mvanbaak 2009-02-23 17:48 +0000 [r178027] Michiel van Baak * CHANGES: list the addition of the SKINNY manager actions in the CHANGES file. 2009-02-23 17:29 +0000 [r178022] Russell Bryant * tests/test_sched.c, main/sched.c: Fix a regression in scheduler entry ordering, and add a regression test for it. (closes issue #14522) Reported by: pj Tested by: russell 2009-02-22 23:04 +0000 [r177988] Michiel van Baak * doc/manager_1_1.txt, channels/chan_skinny.c: Add a couple of manager commands to chan_skinny Added: SKINNYdevices SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521) Reported by: mvanbaak Review: http://reviewboard.digium.com/r/170/ 2009-02-21 15:59 +0000 [r177944] Tilghman Lesher * channels/chan_sip.c: On update, test against the existence of sipregs. 2009-02-21 14:37 +0000 [r177913] Michiel van Baak * main/asterisk.c: add extra check for sysinfo/sysctl (closes issue #14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff uploaded by snuffy (license 35) 2009-02-21 14:16 +0000 [r177884] Sean Bright * main/hashtab.c, include/asterisk/hashtab.h: Trailing whitespace, minor coding guideline fixes, and start beefing up the hashtab documentation a bit. 2009-02-21 13:17 +0000 [r177855] Russell Bryant * include/asterisk/indications.h: Fix build issues on Solaris and OpenBSD. (closes issue #14512) Reported by: snuffy 2009-02-21 13:13 +0000 [r177849-177852] Michiel van Baak * Makefile, contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.archlinux.asterisk, contrib/scripts/safe_asterisk: set ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path When running asterisk as non-root and without this patch the pidfile wants to go into /var/run/asterisk.pid. This directory is not writable for the non-root user and changing permissions is not an option. Putting it in /var/run/asterisk/asterisk.pid makes it possible to set permissions on the /var/run/asterisk dir so everything works as it should be. Patched committed is based on pabelanger's patch. (closes issue #13153) Reported by: pabelanger Patches: 2009012900_bug13153-nonrootscripts.diff.txt uploaded by mvanbaak (license 7) Review: http://reviewboard.digium.com/r/139/ * channels/chan_sip.c: make chan_sip.c compile on OpenBSD again. 2009-02-20 23:02 +0000 [r177732-177787] Tilghman Lesher * main/pbx.c, /: Merged revisions 177786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines Don't print the CR-NL combination when we aren't outputting to the manager. An embedded CR-NL in a CLI command screws up several AMI parsers that don't expect to see that combination in the middle of output. (Closes issue #14305) Reported by: martins Patch by: tilghman ........ * /, include/asterisk/threadstorage.h: Merged revisions 177701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed. Fixed for snuff-home on -dev channel. ........ 2009-02-20 20:29 +0000 [r177699] Dwayne M. Hubbard * apps/app_fax.c: Make app_fax compatible with spandsp-0.0.6pre4 Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred integer to indicate the number of pages transferred (so far) during the fax session. The spandsp-0.0.6pre4 release removed the pages_transferred integer and replaced it with two different integers - pages_tx and pages_rx. This revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards compatibility for previous spandsp releases. 2009-02-20 17:29 +0000 [r177661-177664] Tilghman Lesher * include/asterisk/app.h, main/app.c, apps/app_system.c: Allow semicolons to be escaped, when passing arguments to the System command. (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert * apps/app_voicemail.c: Oops, merge broke trunk 2009-02-20 00:35 +0000 [r177624] Jeff Peeler * channels/chan_sip.c: Set sip_request ast_str data to NULL so ast_str_copy allocates space properly in copy_request (issue #14478) Reported by: erik_dedecker 2009-02-19 23:56 +0000 [r177595] Steve Murphy * /, main/Makefile, main/ast_expr2f.c: Merged revisions 177540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was already pretty 8-bit clean; but I'm still removing the --full from the flex command so everything is uniform. ........ r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines This patch fixes a problem with 8-bit input to the ast_expr2 scanner. The real culprit was the --full argument to flex in the Makefile! This causes a 7-bit scanner to be generated. I reviewed the rules and found one rule where I needed to specifically include 8-bit chars for a token. I tested against the text supplied by ibercom, and all looks very well. This has been there a surprisingly long time! (closes issue #14498) Reported by: ibercom Patches: 14498.patch uploaded by murf (license 17) Tested by: murf ........ 2009-02-19 22:33 +0000 [r177506-177537] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 177536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads. (closes issue #14253) Reported by: Skavin Patches: 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14) Tested by: Skavin ........ * doc/database_transactions.txt (added): Document how to use database transactions 2009-02-19 16:45 +0000 [r177387] Jeff Peeler * include/asterisk/channel.h: Fix another merge error from 176708 2009-02-19 16:38 +0000 [r177384] Joshua Colp * /, apps/app_speech_utils.c: Merged revisions 177383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines If we are able to create a speech structure unset the ERROR variable in case it was previously set. (issue #LUMENVOX-13) ........ 2009-02-19 15:56 +0000 [r177356] Jeff Peeler * main/features.c: Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev on the asterisk-dev mailing list. Thanks! 2009-02-19 00:26 +0000 [r177320] Tilghman Lesher * include/asterisk/res_odbc.h, funcs/func_odbc.c, CHANGES, res/res_odbc.c, configs/res_odbc.conf.sample: ODBC transaction support 2009-02-19 00:08 +0000 [r177291] Joshua Colp * CHANGES: Update CHANGES file to include MWI subscription support that was added some time ago. 2009-02-18 23:51 +0000 [r177287] Tilghman Lesher * main/strings.c: Handle negative length and eliminate a condition that is always true. 2009-02-18 23:50 +0000 [r177286] Steve Murphy * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | 34 lines This patch fixes a regression of sorts that was introduced in rev 24425. It basically fixes AST-190/ABE-1782. What was wrong: the user has 6000 extensions in one context; and then 6000 contexts, one per extension. The parser could only handle about 4893 of the 6000 extens in the single context. This was due to the regression I mentioned. To get rid of shift/reduce conflicts, Luigi set up right-recursive lists for globals, context elements, switch lists, and statements. Right recursive lists got rid of the warnings, but instead, they use up a tremendous amount of stack space when the lists are long. I saw this a few years back, and resolved not to fix it until someone complained. That day has arrived! After the changes were made, I ran the regression test suite, and there were no problems. I took the test case the user provided, and added 100,000 extensions to the single context, that already had 6,000 extens in it. (I'll see your 6, and raise you 100!) It takes a few minutes to read it all in, check it and generate code for it, but no problems. So, I think I can say that fundamentally, there are no longer any limits on the number of items you can place in contexts, statement blocks, switches, or globals, beyond your virt mem constraints. ........ 2009-02-18 23:09 +0000 [r177229] Kevin P. Fleming * main/frame.c: fix two very minor bugs: if anyone ever uses SLINEAR16 as a format in RTP, ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps. 2009-02-18 22:51 +0000 [r177226] David Vossel * main/features.c: Locking issue in action_bridge and bridge_exec action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it. issue# 14296 Review: http://reviewboard.digium.com/r/167/ 2009-02-18 20:11 +0000 [r177162] Jeff Peeler * channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h, channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx, channels/h323/ast_ptlib.h (added), configure, channels/h323/compat_h323.h, configure.ac, channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4, channels/h323/ast_h323.h, channels/h323/chan_h323.h, channels/h323/cisco-h225.cxx: Modify h323 to build against PTLib as well as the older PWLib Several changes in PTLib have occurred requiring build time detection. Changes accounted for include the library name change, config option change, install location change, and a boolean type change which is handled by ast_ptlib.h. Also, the sed check has been modified to properly work with autoconf >= 2.62. (closes issue #14224) Reported by: bergolth Patches: asterisk-autoconf-sed.patch uploaded by bergolth (license 661) asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested by: jpeeler 2009-02-18 19:12 +0000 [r177101] Russell Bryant * apps/app_meetme.c: Re-add 'o' option to MeetMe, reverting rev 62297. Enabling this option by default proved to be a bad idea, as the talker detection is not very reliable. So, make it optional again, and off by default. (issue #13801) Reported by: justdave 2009-02-18 19:05 +0000 [r177098] Tilghman Lesher * /, include/asterisk/config.h: Merged revisions 177096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) | 2 lines Document the return value of the update method (as requested on -dev list) ........ 2009-02-18 17:24 +0000 [r177035] Doug Bailey * main/utils.c: Fixed error where a check for an zero length, terminated string was needed. 2009-02-18 17:11 +0000 [r177005] Joshua Colp * channels/chan_sip.c: Fix ordering of output for a ChannelUpdate manager event. (closes issue #14497) Reported by: vinsik Patches: chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) 2009-02-18 16:09 +0000 [r176948] Doug Bailey * main/utils.c: Need to take into account the \0 terminator of the old string to determine the amount available. 2009-02-18 15:35 +0000 [r176943] Steve Murphy * main/pbx.c: This patch fixes merge_contexts_and_delete so it does not deadlock when hints are present. Reason: when I re-engineered the merge_and_delete func to reduce its lock time, I failed to notice that the functions it calls still also do locking as before. This leads to deadlocks on dialplan reloads, when there are actually living, subscribed hints registered in the system. While the reporter come across this problem while using AEL, I might note that these deadlocks should also happen if extensions.conf were used. Here I added these routines to pbx.c: ast_add_extension_nolock add_pri_lockopt ast_add_extension2_lockopt find_context add_hint_nolock All of the above routines are static and restricted to be used only within pbx.c, and more specifically within the merge_contexts_and_delete routine. They are pretty much the same as their counterparts except they don't lock contexts or hints. Most of them now do the real work of their name-alike, with optional locking via extra arguments, and are called by their name-alike. The goal was to have the original functions so they would behave exactly as before. Both PJ and I tested these fixes, and the deadlocking problem is no longer encountered. (closes issue #14357) Reported by: pj Patches: 14357.diff uploaded by murf (license 17) Tested by: pj, murf 2009-02-18 06:14 +0000 [r176901-176904] Russell Bryant * include/asterisk/heap.h: Add example code for a heap traversal. * main/pbx.c: Fix a number of incorrect uses of strncpy(). The big problem here is that the 3rd argument provided in these uses of strncpy() did not reserve a byte for the null terminator, leaving the potential for writing one byte past the end of the buffer. Aside from this, there were coding guidelines violations with regards to spacing, as well as hard coded lengths being used instead of sizeof(). 2009-02-18 02:55 +0000 [r176869] Dwayne M. Hubbard * channels/chan_sip.c: T38 faxdetect should jump to the 'fax' extension for incoming calls only The previous implementation of T38 faxdetect resulted in both sides of the call jumping to a fax extension when both sides had 't38pt_udptl=yes' and 'faxdetect=yes' in sip.conf and a 'fax' extension in the current context. This revision will jump to a 'fax' extension on incoming calls only. 2009-02-18 02:02 +0000 [r176841] Kevin P. Fleming * main/rtp.c: suppress smoothers for Siren codecs as well as Speex and G.723.1 2009-02-17 22:52 +0000 [r176771] Russell Bryant * apps/app_milliwatt.c: Remove a dependency that no longer exists. 2009-02-17 22:28 +0000 [r176760] Shaun Ruffell * codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice with G723. This commit brings in the changes that were living out on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. codec_dahdi.c now always uses signed linear as the simple codec so that a soft g729 codec will not end up being preferred to the hardware codec. There are also changes to allow codec_dahdi.c to feed packets to the hardware in the native sample size of the codec. This solves problems with choppy audio when using G723. 2009-02-17 22:08 +0000 [r176708] Jeff Peeler * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ 2009-02-17 22:02 +0000 [r176706] Mark Michelson * tests/test_sched.c: Use constants from inttypes.h to clear up 32-bit compilation errors 2009-02-17 21:59 +0000 [r176705] Dwayne M. Hubbard * channels/chan_sip.c: create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38 This is required to create a UDPTL structure in create_addr_from_peer() to handle the scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but is defined the peer's context. I tested this patch by enabling t38pt_udptl in the [general] section on one system and only enabling t38pt_udptl in a peer's context on the system sending a fax. Without the patch, the sending system will fail to initiate T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure". When this patch is applied the sending side will successfully initiate T38 negotiation. 2009-02-17 21:40 +0000 [r176697] Mark Michelson * include/asterisk/frame.h: Clear up documentation of AST_FRIENDLY_OFFSET in frame.h 2009-02-17 21:23 +0000 [r176669] Tilghman Lesher * /: Recorded merge of revisions 176661 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009) | 9 lines Backport change to 1.4: Prior to masquerade, move the group definitions to the channel performing the masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ 2009-02-17 21:22 +0000 [r176666] Russell Bryant * main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c, res/res_timing_timerfd.c, include/asterisk/timing.h, main/timing.c: Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ 2009-02-17 21:14 +0000 [r176642] Tilghman Lesher * channels/chan_local.c: Prior to masquerade, move the group definitions to the channel performing the masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma 2009-02-17 21:04 +0000 [r176632-176639] Russell Bryant * tests/test_sched.c (added), main/sched.c: Significantly improve scheduler performance under high load. This patch changes the scheduler to use a max-heap to store pending scheduler entries instead of a fully sorted doubly linked list. When the number of entries in the scheduler gets large, this will perform much better. For much more detailed information on this change, see the review request. Review: http://reviewboard.digium.com/r/160/ * tests/test_heap.c (added): Add a test module for the heap implementation. Review: http://reviewboard.digium.com/r/160/ * main/Makefile, main/heap.c (added), include/asterisk/heap.h (added): Add an implementation of the heap data structure. A heap is a convenient data structure for implementing a priority queue. Code from svn/asterisk/team/russell/heap/. Review: http://reviewboard.digium.com/r/160/ 2009-02-17 20:50 +0000 [r176631] Olle Johansson * include/asterisk/config.h: Typo 2009-02-17 20:41 +0000 [r176627] Russell Bryant * channels/chan_unistim.c, main/pbx.c, apps/app_read.c, configs/indications.conf.sample, apps/app_playtones.c (added), include/asterisk/indications.h, apps/app_readexten.c, apps/app_disa.c, UPGRADE.txt, include/asterisk/channel.h, include/asterisk/_private.h, main/indications.c, main/loader.c, main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_channel.c, res/snmp/agent.c, main/app.c, res/res_indications.c (removed), main/asterisk.c: Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ 2009-02-17 18:49 +0000 [r176592] Tilghman Lesher * funcs/func_odbc.c, res/res_odbc.c: Add assertions in the quest to track down a refcount leak. (closes issue #14485) Reported by: davevg 2009-02-17 17:33 +0000 [r176557] Russell Bryant * main/pbx.c, apps/app_queue.c: Fix a race condition that caused device states to become incorrect for hints. The problem here is that the hint processing code was subscribed to the wrong event type. So, it started processing state for a hint too soon, before the device state cache had been updated. Also, fix a similar bug in app_queue, as it was also subscribed to the wrong event type. (closes issue #14461) Reported by: alecdavis 2009-02-17 17:28 +0000 [r176513-176556] Olle Johansson * configs/extconfig.conf.sample: Typo * main/config.c: If there are no realtime engines, there's no reason to check for realtime families 2009-02-17 14:39 +0000 [r176360-176501] Tilghman Lesher * channels/chan_sip.c: In this version, we can combine the queries, because we support dropping nonexistent columns. * /, channels/chan_sip.c: Merged revisions 176426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines After a 'sip reload', qualifies for realtime peers weren't immediately restarted, instead waiting until the next registration. We're now caching the qualify across a reload/restart and starting the qualify immediately upon loading the peer. (closes issue #14196) Reported by: pdf Patches: 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) Tested by: pdf ........ * main/strings.c: Might want to update the buffer pointer after a realloc (or we crash) (closes issue #14485) Reported by: davevg 2009-02-16 23:37 +0000 [r176356] Kevin P. Fleming * sounds/sounds.xml: add support for Siren7 and Siren14 flavors of prompts and music on hold 2009-02-16 23:33 +0000 [r176355] David Vossel * /, channels/chan_iax2.c: Merged revisions 176354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that. issue #13749 ........ 2009-02-16 23:14 +0000 [r176320] Tilghman Lesher * channels/chan_skinny.c: Use the correct list macros for deleting an item from the middle of a list. (issue #13777) Reported by: pj Patches: 20090203__bug13777.diff.txt uploaded by Corydon76 (license 14) Tested by: pj 2009-02-16 21:45 +0000 [r176255] Kevin P. Fleming * /, main/utils.c, include/asterisk/stringfields.h: Merged revisions 176216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ 2009-02-16 21:40 +0000 [r176253] Mark Michelson * /, apps/app_meetme.c: Merged revisions 176249,176252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it to be nonblocking atomically Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys ........ r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines Remove unused variable and make dev-mode compilation happy ........ 2009-02-16 21:30 +0000 [r176248] David Vossel * channels/chan_iax2.c: Merged revisions 175597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ 2009-02-16 18:25 +0000 [r176174] Mark Michelson * main/logger.c: Assist proper thread synchronization when stopping the logger thread. I was finding that on my dev box, occasionally attempting to "stop now" in trunk would cause Asterisk to hang. I traced this to the fact that the logger thread was waiting on a condition which had already been signalled. The logger thread also need to be sure to check the value of the close_logger_thread variable. The close_logger_thread variable is only checked when the list of logmessages is empty. This allows for the logger thread to print and free any pending messages before exiting. 2009-02-16 17:44 +0000 [r176138] Tilghman Lesher * channels/chan_dahdi.c: Can't set debug level 2 (intense debugging) unless the syntax matches 2009-02-16 17:09 +0000 [r176100] Russell Bryant * channels/chan_features.c (removed): Remove chan_features. Review: http://reviewboard.digium.com/r/161/ 2009-02-16 15:36 +0000 [r176030] Joshua Colp * /, channels/chan_sip.c: Merged revisions 176029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) ........ 2009-02-16 02:54 +0000 [r175983] Russell Bryant * main/channel.c: Make the causes array static, and remove the type name as it is not needed. 2009-02-16 00:26 +0000 [r175952] Michiel van Baak * channels/chan_unistim.c, /, channels/chan_sip.c, include/asterisk/manager.h, doc/unistim.txt: Merged revisions 175921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ 2009-02-15 21:27 +0000 [r175829-175882] Russell Bryant * include/asterisk/sched.h, main/sched.c: Make ast_sched_report() and ast_sched_dump() thread safe. * channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: Fix a number of problems with ast_sched_report(). 1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. 2009-02-15 20:39 +0000 [r175783-175827] Olle Johansson * formats/format_ilbc.c, /: Merged revisions 175825 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb 2009) | 2 lines format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta! ........ * formats/format_ilbc.c, /: Merged revisions 175792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb 2009) | 2 lines Disable format_ilbc.so by default, like codec_ilbc.so ........ * /, channels/chan_sip.c: Merged revisions 175777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2 lines Make sure that the debug line is not printed on debug level 0 ........ 2009-02-13 20:57 +0000 [r175655-175663] Mark Michelson * doc/manager_1_1.txt, CHANGES, apps/app_queue.c: Merge queue-reset branch to Asterisk From a user point-of-view, this adds new CLI commands and Manager Actions to better facilitate the reloading of queues and the resetting of their statistics. The new CLI commands are the "queue reload" and "queue reset stats" commands. The new manager actions are the QueueReload and QueueReset commands. Review: http://reviewboard.digium.com/r/115 * doc/manager_1_1.txt, apps/app_chanspy.c: Add manager events for chanspy starting or stopping (closes issue #14469) Reported by: caio1982 Patches: chanspy_events2.diff uploaded by caio1982 (license 22) 2009-02-13 20:26 +0000 [r175623-175636] Russell Bryant * res/res_jabber.c: fix a few more XML documentation problems * main/pbx.c: add missing 2009-02-13 20:11 +0000 [r175597] David Vossel * configs/iax.conf.sample, channels/iax2.h, channels/chan_iax2.c: Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. Review: http://reviewboard.digium.com/r/159/ 2009-02-13 19:49 +0000 [r175591] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 175590 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) ........ 2009-02-13 16:41 +0000 [r175549] Joshua Colp * apps/app_record.c: Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian 2009-02-13 13:41 +0000 [r175508-175512] Kevin P. Fleming * CHANGES: document G.722.1/.1C support * main/frame.c, channels/chan_sip.c, include/asterisk/rtp.h, channels/chan_h323.c, include/asterisk/frame.h, formats/format_siren14.c (added), main/rtp.c, formats/format_siren7.c (added): Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14) This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ 2009-02-13 04:22 +0000 [r175411-175475] Dwayne M. Hubbard * CHANGES: add 'faxbuffers' configuration option information to CHANGES * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add dynamic fax buffer configuration option to chan_dahdi.conf When the 'faxdetect' configuration option is used, one may also want to use the 'faxbuffers' configuration option in chan_dahdi.conf. This option will dynamically use the configured 'faxbuffers' buffer policy on a channel for the life of the call following the detection of fax tones. The faxbuffers buffer policy will be reverted during call teardown. An example use of 'faxbuffers' is below. This example would switch to using 6 buffers with a full buffer policy. faxbuffers=>6,full 2009-02-12 21:41 +0000 [r175368] Russell Bryant * channels/chan_sip.c: Remove useless string copy, and make sscanf safe again 2009-02-12 21:27 +0000 [r175344] David Vossel * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds force encryption option to iax.conf This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well. (closes issue #13285) Reported by: sgofferj Tested by: russell Review: http://reviewboard.digium.com/r/150/ 2009-02-12 21:25 +0000 [r175334] Tilghman Lesher * main/udptl.c, /: Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ 2009-02-12 20:48 +0000 [r175298] Jeff Peeler * /, main/features.c: Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ 2009-02-12 20:45 +0000 [r175255-175295] Russell Bryant * channels/chan_sip.c: Avoid using ast_strdupa() in a loop. * build_tools/cflags.xml: Don't enable something by default that has a dependency on something _not_ enabled by default. menuselect was not happy with this. 2009-02-12 18:48 +0000 [r175250] Kevin P. Fleming * channels/chan_iax2.c: correct warning message to not refer specifically to DAHDI 2009-02-12 18:00 +0000 [r175188] Jeff Peeler * /, main/features.c: Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ 2009-02-12 17:07 +0000 [r175127] David Vossel * channels/chan_iax2.c: Setting key rotation to be off by default Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off. 2009-02-12 16:57 +0000 [r175125] Russell Bryant * /, main/rtp.c: Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ 2009-02-12 16:28 +0000 [r175121] Mark Michelson * include/asterisk/astobj2.h, main/astobj2.c: Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. 2009-02-12 14:25 +0000 [r175058-175089] Philippe Sultan * channels/chan_gtalk.c: Issue a warning message if our candidate's IP is the loopback address. (closes issue #13985) Reported by: jcovert Tested by: phsultan * /, channels/chan_gtalk.c: Merged revisions 175029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ 2009-02-11 23:12 +0000 [r174945-174951] Mark Michelson * apps/app_queue.c: Fix a bit of odd logic for announcing position. Sync with 1.6.0's logic * apps/app_queue.c: Fix odd "thank you" sound playing behavior in app_queue.c If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy * apps/app_dial.c, main/channel.c, main/pbx.c, apps/app_dictate.c, apps/app_waitforsilence.c, include/asterisk/channel.h: Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 2009-02-11 14:44 +0000 [r174844] Joshua Colp * main/channel.c: Tell the device state core a change happened when a channel is freed but not a specific state. We need to do this because while we know that the freeing of the channel may cause something to become not in use we do not know this for sure. There may be another channel that is still up which would cause it to be in use. (closes issue #13238) Reported by: kowalma Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis 2009-02-10 23:17 +0000 [r174764-174805] Mark Michelson * apps/app_chanspy.c: Fix potential for stack overflows in app_chanspy.c When using the 'g' or 'e' options, the stack allocations that were used could cause a stack overflow if a spyer stayed on the line long enough without actually successfully spying on anyone. The problem has been corrected by using static buffers and copying the contents of the appropriate strings into them instead of using functions like alloca or ast_strdupa * main/manager.c: Fix an fd leak that would occur in HTTP AMI sessions The explanation behind this fix is a bit complicated, and I've already typed it up in the code as a huge comment inside of manager.c, so I'll give the abridged version here. We needed a way to separate action-specific data from session-specific data. Unfortunately, the only way to maintain API compatibility and to not have to change every single manager action was to rename the current mansession structure and wrap it inside a new mansession structure which actually contains action- specific data. (closes issue #14364) Reported by: awk Patches: 14364_better.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/148/ 2009-02-10 20:15 +0000 [r174710] Joshua Colp * channels/chan_sip.c: Only decrease inringing count if above zero. (issue #13238) Reported by: kowalma 2009-02-10 19:38 +0000 [r174705] Kevin P. Fleming * main/slinfactory.c, include/asterisk/slinfactory.h: improve slinfactory API to remove implicit sample rate and require explicit sample rate selection by creator of the slinfactory 2009-02-10 18:16 +0000 [r174584] Matthew Nicholson * /, main/jitterbuf.c: Merged revisions 174583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines Improve behavior of jitterbuffer when maxjitterbuffer is set. This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ ........ 2009-02-10 17:48 +0000 [r174543-174580] Joshua Colp * channels/chan_sip.c: Set the type for the peer structure to be a peer as the default. (closes issue #14447) Reported by: triccyx * channels/chan_sip.c: Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported. (closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma 2009-02-10 07:06 +0000 [r174470-174503] Tilghman Lesher * apps/app_stack.c, apps/app_voicemail.c: Fix0ring build * apps/app_stack.c: Remove the usage of the KeepAlive app, as it no longer exists. 2009-02-10 04:49 +0000 [r174370-174435] Steve Murphy * apps/app_rpt.c: This patch removes the use of AST_PBX_KEEPALIVE from app_rpt.c. (closes issue #14435) Reported by: D_McNaul * apps/app_rpt.c: More intptr_t work. * /, apps/app_rpt.c: Merged revisions 174369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines This patch solves some compiler complaints in both 32 and 64-bit environments. ........ 2009-02-09 17:27 +0000 [r174327] Mark Michelson * channels/chan_sip.c: Fix something I messed up in the merge I just did 2009-02-09 17:26 +0000 [r174325] David Vossel * apps/app_externalivr.c: Fixes issue with hangups not being sent and external process never terminating. The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued. If the external script expects to be notified of a hangup and never receives one, it runs indefinitely. (closes issue #14251) Reported by: chris-mac Tested by: dvossel 2009-02-09 17:20 +0000 [r174301] Mark Michelson * /, channels/chan_sip.c: Merged revisions 174282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ 2009-02-09 14:49 +0000 [r174219] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 174218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off. (closes issue #14407) Reported by: mostyn ........ 2009-02-07 16:16 +0000 [r174149] Russell Bryant * /, res/snmp/agent.c: Merged revisions 174148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines Fix a race condition that could cause a crash. ........ 2009-02-06 23:51 +0000 [r174084] Dwayne M. Hubbard * /, channels/chan_sip.c: Merged revisions 174082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ 2009-02-06 20:12 +0000 [r174046] David Vossel * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds immediate yes/no option to iax.conf This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well. (closes issue #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk uploaded by jcovert (license 551) iax.conf.sample.patch uploaded by jcovert (license 551) Tested by: jcovert, dvossel Review: http://reviewboard.digium.com/r/143/ 2009-02-06 19:28 +0000 [r173974-174041] Joshua Colp * channels/chan_dahdi.c: Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription. (closes issue #14322) Reported by: amessina * /, channels/chan_sip.c: Merged revisions 173967-173968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ 2009-02-06 16:28 +0000 [r173952] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 173917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ 2009-02-06 15:59 +0000 [r173902] Joshua Colp * main/audiohook.c, apps/app_chanspy.c: Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached. (closes issue #14414) Reported by: bluecrow76 2009-02-06 10:55 +0000 [r173848-173858] Russell Bryant * include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c: Add a common implementation of a scheduler context with a dedicated thread. This commit expands the Asterisk scheduler API to include a common implementation of a scheduler context being processed by a dedicated thread. chan_iax2 has been updated to use this new code. Also, as a result, this resolves some race conditions related to the previous chan_iax2 scheduler handling. Related to rev 171452 which resolved the same issues in 1.4. Code from team/russell/sched_thread2 Review: http://reviewboard.digium.com/r/129/ * main/manager.c: Resolve a memory leak that would occur on an invalid channel given to Action: Status 2009-02-05 23:48 +0000 [r173773-173776] Mark Michelson * configs/extensions.conf.sample: Update extensions.conf.sample to be correct. In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up * apps/app_voicemail.c: Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage (closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) 2009-02-05 21:00 +0000 [r173697] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 173696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ 2009-02-05 20:30 +0000 [r173693] Mark Michelson * /, apps/app_queue.c: Merged revisions 173692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ 2009-02-05 19:36 +0000 [r173657] Tilghman Lesher * res/res_config_sqlite.c: Change the first field, or we don't get the necessary field separation. 2009-02-05 18:48 +0000 [r173507-173593] Mark Michelson * /, apps/app_mixmonitor.c: Merged revisions 173592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor ........ * /, apps/app_mixmonitor.c: Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ * apps/app_queue.c: Fix some areas where the incorrect interface was passed to ast_device_state I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r 2009-02-04 21:26 +0000 [r173503] Tilghman Lesher * res/res_jabber.c: Add XML documentation for the applications and functions in res_jabber (closes issue #14405) Reported by: snuffy Patches: xml_jabber.diff uploaded by snuffy (license 35) 2009-02-04 21:25 +0000 [r173502] David Vossel * channels/iax2-parser.h, channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing off calls. Reverts changes in 116884 Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. The changes reverted in 116884 caused backwards compatibility issues involving iax2 transfer with 1.6.0, 1.4, and 1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel 2009-02-04 21:17 +0000 [r173500] Jeff Peeler * /, main/features.c, include/asterisk/features.h: Merged revisions 173211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ 2009-02-04 18:48 +0000 [r173458] Tilghman Lesher * main/tcptls.c: When using a socket as a FILE *, the stdio functions will sometimes try to do an fseek() on the stream, which is an invalid operation for a socket. Turning off buffering explicitly lets the stdio functions know they cannot do this, thus avoiding a potential error. (closes issue #14400) Reported by: fnordian Patches: tcptls.patch uploaded by fnordian (license 110) 2009-02-04 17:45 +0000 [r173354-173397] Mark Michelson * /, apps/app_chanspy.c: Merged revisions 173396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines Revert my previous change because it was stupid ........ * /, apps/app_chanspy.c: Merged revisions 173392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever matter, but it's needed. ........ * main/file.c: Fix a problem where file playback would cause fds to remain open forever The problem came from the fact that a frame read from a format interpreter was not freed. Adding a call to ast_frfree fixed this. The explanation for why this caused the problem is a bit complex, but here goes: There was a problem in all versions of Asterisk where the embedded frame of a filestream structure was referenced after the filestream was freed. This was fixed by adding reference counting to the filestream structure. The refcount would increase every time that a filestream's frame pointer was pointing to an actual frame of data. When the frame was freed, the refcount would decrease. Once the refcount reached 0, the filestream was freed, and as part of the operation, the open files were closed as well. Thus it becomes more clear why a missing ast_frfree would cause a reference leak and cause the files to not be closed. You may ask then if there was a frame leak before this patch. The answer to that is actually no! The filestream code was "smart" enough to know that since the frame we received came from a format interpreter, the frame had no malloced data and thus didn't need to be freed. Now, however, there is cleanup that needs to be done when we finish with the frame, so we do need to call ast_frfree on the frame to be sure that the refcount for the filestream is decremented appropriately. (closes issue #14384) Reported by: fiddur Patches: 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, putnopvut 2009-02-04 00:43 +0000 [r173311] Tilghman Lesher * main/pbx.c, pbx/pbx_config.c: Ensure that commas placed in the middle of extension character classes do not interfere with correct parsing of the extension. Also, if an unterminated character class DOES make its way into the pbx core (through some other method), ensure that it does not crash Asterisk. (closes issue #14362) Reported by: Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 2009-02-03 17:35 +0000 [r173169] Richard Mudgett * channels/chan_dahdi.c: Broke up the large conditional blocks so it is easy to see if a function is compiled. 2009-02-03 00:29 +0000 [r173104-173130] Tilghman Lesher * configure, include/asterisk/autoconfig.h.in, configure.ac, main/xml.c, include/asterisk/compiler.h, apps/app_stack.c, include/asterisk/optional_api.h: 1. Make OS X compile cleanly with app_stack. 2. Use curl to download sound files, as curl is installed natively on OS X, whereas wget and fetch are not. (closes issue #14332) Reported by: oej Tested by: Corydon76 * /, configs/extensions.conf.sample: Merged revisions 173070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ 2009-02-02 23:57 +0000 [r173067] Terry Wilson * /, main/features.c: Merged revisions 173066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines Fix a feature inheritance bug I added after code review ........ 2009-02-02 23:21 +0000 [r173028-173047] Mark Michelson * main/manager.c, CHANGES: Reverting commit number 173028 as there are some potential issues * main/manager.c, CHANGES: Add a CLI command to log out a manager user (closes issue #13877) Reported by: eliel Patches: cli_manager_logout.patch.txt uploaded by eliel (license 64) Tested by: eliel, putnopvut 2009-02-02 20:40 +0000 [r172963] Richard Mudgett * /: Recorded merge of revisions 172962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009) | 11 lines channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. These changes are already in trunk -r172400 ........ 2009-02-02 19:02 +0000 [r172929] Steve Murphy * apps/app_dial.c, main/features.c, CHANGES, include/asterisk/features.h: This reverts the changes I made for 11583; will reviewboard this before committing again... reopened 11583 until all Russell's issues are resolved. 2009-02-02 18:13 +0000 [r172894] Leif Madsen * configs/res_ldap.conf.sample: Update the res_ldap.conf file with a better working example. (closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert 2009-02-02 17:37 +0000 [r172890] Steve Murphy * apps/app_dial.c, main/features.c, CHANGES, include/asterisk/features.h: This change allows the disconnect feature (as in "one-touch" in features.c) to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf 2009-02-02 16:42 +0000 [r172855] Russell Bryant * channels/chan_sip.c: Fix a spelling mistake. 2009-02-02 10:46 +0000 [r172816-172818] Olle Johansson * channels/chan_sip.c: Add a todo. I do need to really check what's going on with this kill-the-user business ;-) Why do we suddenly have two flags to set peer type? * channels/chan_sip.c: Small formatting change * channels/chan_sip.c: Add some well-needed improvements to the wishlist in the code, so that we can close some bug reports. 2009-02-02 01:41 +0000 [r172778] Sean Bright * channels/chan_sip.c: The CID lookup feature wasn't actually working properly with dialog-info+xml supporting devices. The devices (snoms, specifically) need to receive a SIP URI instead of just an extension. This adds that functionality. 2009-02-01 02:44 +0000 [r172706-172741] Tilghman Lesher * apps/app_voicemail.c: Blank argument crashes Asterisk (closes issue #14377) Reported by: amorsen * funcs/func_strings.c: Don't increment the loop, now that incrementing is taken care of by the decoder function. (closes issue #14363) Reported by: andrew53 Patches: func_strings_filter.patch uploaded by andrew53 (license 519) 2009-01-30 22:22 +0000 [r172598] Mark Michelson * include/asterisk/channel.h: Fix redefinition of flag in channel.h 2009-01-30 21:50 +0000 [r172580-172581] Terry Wilson * configs/features.conf.sample: Remove incorrect line from sample config * apps/app_dial.c, main/global_datastores.c, main/features.c, include/asterisk/global_datastores.h, CHANGES, configs/features.conf.sample: Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ 2009-01-30 18:36 +0000 [r172441-172548] Tilghman Lesher * funcs/func_aes.c: Parameter position reversed in documentation * /, autoconf/ast_func_fork.m4, configure, main/app.c, apps/app_rpt.c, main/asterisk.c: Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ 2009-01-29 23:15 +0000 [r172370-172440] Richard Mudgett * main/cli.c: Remove tabs from comment * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. * include/asterisk/say.h: Fixed some doxygen comments 2009-01-29 17:10 +0000 [r172318-172319] Olle Johansson * channels/chan_local.c: Revert two lines that was extra, but only on fridays. * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c, include/asterisk/causes.h, apps/app_queue.c: Fix "cancel answered elsewhere" through app_queue with members in chan_local. Also, implement a private cause code (as suggested by Tilghman). This works with chan_sip, but doesn't propagate through chan_local. 2009-01-29 16:48 +0000 [r172315] Tilghman Lesher * configs/func_odbc.conf.sample: Better document mode=multirow, based upon a conversation with Jared. 2009-01-29 13:47 +0000 [r172271] Leif Madsen * contrib/scripts/realtime_pgsql.sql: The realtime_pgsql.sql script is missing a couple of fields. closes issue #14339) Reported by: fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678) 2009-01-29 13:24 +0000 [r172173-172270] Olle Johansson * configs/sip.conf.sample, CHANGES: Update documentation * include/asterisk/app.h, channels/chan_sip.c, main/app.c: - Make sure we set setvar= variables on outbound calls too, not only inbound calls. - Also, change a function in app.c to return a userful value instead of always returning 0. Patch by fnordian, changed by Corydon76 and myself. This does not close the bug report, as fnordian had an additional change we're still discussing. (related to issue #14059) Reported by: fnordian Patches: chan_sip_hfield.patch uploaded by fnordian (license 110) 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14) Tested by: fnordian, Corydon76, oej * channels/chan_sip.c: Make sure register= line supports both port and expiry at the same time. (closes issue #14185) Reported by: Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657) Tested by: Nick_Lewis * /, channels/chan_sip.c: Merged revisions 172169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 ........ 2009-01-28 22:52 +0000 [r172132] Steve Murphy * channels/chan_misdn.c: A further correction: cast the sizeof to an int. 2009-01-28 22:48 +0000 [r172131] Tilghman Lesher * res/res_config_odbc.c: Fix how we skip fields (to avoid fields which don't exist) when doing an UPDATE. (closes issue #14205) Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage 2009-01-28 21:48 +0000 [r172063-172099] Steve Murphy * channels/chan_misdn.c: my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2 didn't like the \%ld and issued a warning, breaking my dev-mode build. This fixes it. * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ 2009-01-28 17:27 +0000 [r171964] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 171963 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines Clarify log message (suggested by manxpower on #asterisk-dev) ........ 2009-01-28 14:39 +0000 [r171838-171925] Olle Johansson * CHANGES: Yep. Documentation is important. * apps/app_queue.c: Add final part of previously committed work for answered elsewhere in queue - the missing piece that started with app_dial() earlier on. This is to avoid having the list and counter of missed calls being touched by queue calls. Add the C option to queue() and nothing will be logged on phones that support the Reason: header on SIP cancel, like the SNOM phones. * configs/sip.conf.sample: Add some more notes about device matching. * /, configs/sip.conf.sample: Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ 2009-01-28 00:17 +0000 [r171797] Mark Michelson * funcs/func_aes.c: Fix some signedness problems in func_aes.c 2009-01-27 23:28 +0000 [r171793] Matthew Fredrickson * channels/chan_dahdi.c: Don't complain about lack of D-channels on PTMP connections 2009-01-27 22:43 +0000 [r171757] David Vossel * funcs/func_aes.c (added), CHANGES: Adding AES_ENCRYPT and AES_DECRYPT dialplan functions. (closes issue #14301) Reported by: amorsen review: http://reviewboard.digium.com/r/128/ 2009-01-27 21:58 +0000 [r171618-171691] Mark Michelson * channels/chan_agent.c: Merged revisions 171689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines Fix devicestate problems for "always-on" agent channels A revision to chan_agent attempted to "inherit" the device state of the underlying channel in order to report the device state of an agent channel more accurately. The problem with the logic here is that it makes no sense to use this for always-on agents. If the agent is logged in, then to the underlying channel, the agent will always appear to be "in use," no matter if the agent is on a call or not. The reason is that to the underlying channel, the channel is currently in use on a call to the AgentLogin application. The most common cause that I found for this issue to occur was for a SIP channel to be the underlying channel type for an Agent channel. If the SIP phone re-registers, then the registration will cause the device state core to query the device state of the SIP channel. Since the SIP channel is in use, the Agent channel would also inherit this status. Once the agent channel was set to "in use" there was no way that the device state could change on that channel unless the agent logged out. The solution for this problem is a bit different in 1.4 than it is in the other branches. In 1.4, there will be a one-line fix to make sure that only callback agents will inherit device state from their underlying channel type. For the other branches of Asterisk, since callback support has been removed, there is also no need for device state inheritance in chan_agent, so I will simply be removing it from the code. In addition, the 1.4 source is getting a new comment to help the next person who edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be used to determine if the agent is a callback agent or not. (closes issue #14173) Reported by: nathan Patches: 14173.patch uploaded by putnopvut (license 60) Tested by: nathan, aramirez ........ * /, main/slinfactory.c: Merged revisions 171621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines Prevent a crash from occurring when a jitter buffer interpolated frame is removed from a slinfactory slinfactory used the "samples" field of an ast_frame in order to determine the amount of data contained within the frame. In certain cases, such as jitter buffer interpolated frames, the frame would have a non-zero value for "samples" but have NULL "data" This caused a problem when a memcpy call in ast_slinfactory_read would attempt to access invalid memory. The solution in use here is to never feed frames into the slinfactory if they have NULL "data" (closes issue #13116) Reported by: aragon Patches: 13116.diff uploaded by putnopvut (license 60) ........ * apps/app_queue.c: Fix queue crashes that would occur after the calling channel was masqueraded. The data passed to the end_bridge_callback was assumed to be data which was still stack'd. The problem was that with some call features, attended transfers in particular, a new bridge thread is started once the feature completes, meaning that when the end_bridge_callback is called, the end_bridge_callback_data was invalid. To fix this problem, there are two measures taken 1. Instead of pointing to stacked data, we now used heap-allocated data for passing to the end_bridge_callback in app_queue 2. Since bridges can end multiple times on a single logical call, we wait until the final bridge is broken to actually set any queue variables. This is accomplished through reference-counting and the use of an end_bridge_callback_data_fixup function in app_queue.c (closes issue #14260) Reported by: ccesario Patches: 14260.patch uploaded by putnopvut (license 60) Tested by: ccesario 2009-01-27 15:23 +0000 [r171558] Doug Bailey * channels/chan_dahdi.c: Handle new VMWI ioctl structure (Now there are two VMWI ioctl calls.) (issue #14104) Reported by: alecdavis Tested by: dbailey 2009-01-27 15:00 +0000 [r171263-171528] Olle Johansson * /, channels/chan_sip.c: Solving the same issue, but a bit different in trunk... Merged revisions 171527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines Use the same branch tag in CANCEL as in INVITE Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now. I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems. Thanks Fredrik for pointing out where the bug in the SIP messaging was. (closes issue #14346) Reported by: oej Patches: bug14346.diff uploaded by oej (license 306) Tested by: oej ........ * channels/chan_sip.c: Moving generic setting to friends * channels/chan_sip.c: Continue to move variables into the sip_cfg structure to make them easier to handle in the future as a group of settings for a group of devices. At some point, I want one sip_cfg per domain handled, so we can have "group" settings. * channels/chan_sip.c: Just moving around variable declarations so that we have all globals in the same place. Default setting is set before we activate the channel or at reloads, not where we declare the variable. * /, channels/chan_sip.c: Merged revisions 171264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ * main/channel.c: Add extensions and context on manager event when new channel is created. 2009-01-25 23:58 +0000 [r171188] Tilghman Lesher * /, channels/chan_oss.c: Merged revisions 171187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines Correctly track the hookstate (closes issue #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt uploaded by Corydon76 (license 14) ........ 2009-01-25 16:50 +0000 [r171043-171081] Michiel van Baak * channels/chan_skinny.c: dont segfault when a MWI event occurs on a line without a registered device * configs/skinny.conf.sample: Make the sample skinny.conf work (closes issue #14325) Reported by: DEA Patches: skinny.conf.sample-trunk.txt uploaded by DEA (license 3) 2009-01-25 13:35 +0000 [r170980] Sean Bright * /, apps/app_page.c: Merged revisions 170979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines Resolve a logic error that was causing Page() to crash when more than one channel was specified. (closes issue #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt uploaded by seanbright (license 71) Tested by: kc0bvu ........ 2009-01-25 02:49 +0000 [r170902-170943] Russell Bryant * include/asterisk/utils.h: Change ARRAY_LEN() to be more C++ safe. When the second part of this macro is written as 0[a] instead of a[0], it will force a failure if the macro is used on a C++ object that overloads the [] operator. * res/res_agi.c: Add a todo to finish the XML docs in this module 2009-01-24 13:55 +0000 [r170837] Tilghman Lesher * /, configs/res_odbc.conf.sample: Merged revisions 170836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009) | 2 lines Remove superfluous implementation note (closes issue #14319) ........ 2009-01-23 23:10 +0000 [r170794] Richard Mudgett * doc/tex/Makefile: Fix asterisk.pdf generation if branch name has an underscore in it. 2009-01-23 22:58 +0000 [r170790] Russell Bryant * doc/tex/Makefile: Don't blow up if a branch name has an underscore in it 2009-01-23 20:56 +0000 [r170677-170720] Mark Michelson * /, configs/res_odbc.conf.sample: Merged revisions 170719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan 2009) | 8 lines Add notes to the idlecheck explanation in res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000 Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65) ........ * /, contrib/i18n.testsuite.conf: Merged revisions 170671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use deprecated syntax * Convert Wait,1 to Wait(1) * Convert SetLanguage to Set(CHANNEL(language)) * Use 'n' for all priorities beyond the first Also added test for Chinese numbers, too. (closes issue #14320) Reported by: dant Patches: i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license 670) ........ 2009-01-23 20:18 +0000 [r170652] Joshua Colp * main/channel.c, /: Merged revisions 170648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them. (closes issue #14249) Reported by: RadicAlish ........ 2009-01-23 19:25 +0000 [r170608] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 170588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23 Jan 2009) | 2 lines Additions to AST-2009-001 ........ 2009-01-23 19:09 +0000 [r170505-170569] Joshua Colp * apps/app_dial.c, /: Merged revisions 170568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself. (closes issue #14310) Reported by: RadicAlish ........ * /, channels/chan_sip.c: Merged revisions 170504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold. (closes issue #14295) Reported by: klaus3000 ........ 2009-01-23 17:46 +0000 [r170501] Michiel van Baak * channels/chan_h323.c: let's use SENTINEL where needed 2009-01-23 17:32 +0000 [r170498] Joshua Colp * apps/app_voicemail.c: Reset the ast_str used for escape substitution. We need to do this since it is a thread local variable that may contain the value of a previous substitution. (closes issue #14312) Reported by: pj 2009-01-23 17:03 +0000 [r170463] Matthew Fredrickson * channels/chan_dahdi.c: We should not do restart messages if we're in PTMP mode 2009-01-23 16:57 +0000 [r170460] Michiel van Baak * channels/chan_skinny.c: Dont clear the display of skinny phones when not needed. (closes issue #13182) Reported by: pj Patches: 2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak, pj 2009-01-23 16:35 +0000 [r170457] Doug Bailey * channels/chan_dahdi.c: MWI messages included in CID spill was not being properly handled and prevented the call from being processed (issue #14313) Reported by: seandarcy Tested by: dbailey 2009-01-23 15:44 +0000 [r170393] Mark Michelson * main/channel.c, /: Merged revisions 170392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines Fix broken call pickup There was a subtle change in ast_do_masquerade which resulted in failed attempts to pickup calls. The problem was that the value of the AST_FLAG_OUTGOING flag was copied from the clone to the original channel. In the case of call pickup, this meant that the AST_FLAG_OUTGOING flag ended up being cleared on the channel that was attempting to execute the pickup. Because this flag was not set, when ast_read came across an answer frame, it ignored it. The result of this was that the calling channel was never properly answered. This fix changes the behavior in ast_do_masquerade to set the flags on the original channel to the union of the flags on the clone channel. This way, if the AST_FLAG_OUTGOING flag is set on either of the two channels involved in the masquerade, the resulting channel will have the flag set as well. (closes issue #14206) Reported by: francesco_r Patches: 14206.patch uploaded by putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut ........ 2009-01-22 23:23 +0000 [r170351] Matthew Fredrickson * channels/chan_dahdi.c: Make sure we don't set the channel to be inalarm for a D-channel drop on PTMP connections 2009-01-22 21:25 +0000 [r170307] Tilghman Lesher * main/abstract_jb.c: Create logfile safely. (closes issue #14160) Reported by: tzafrir Patches: 20090104__bug14160.diff.txt uploaded by Corydon76 (license 14) 2009-01-22 20:04 +0000 [r170240] Joshua Colp * /, main/rtp.c: Merged revisions 170239 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 lines Don't crash if RTCP is not enabled on an RTP structure but statistics are output. (closes issue #14234) Reported by: jcovert Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551) rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........ 2009-01-22 17:19 +0000 [r170165] Tilghman Lesher * /, pbx/pbx_config.c: Merged revisions 170158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009) | 6 lines Allow global variables after substitution to be as long as other variables. (closes issue #14263) Reported by: markd Patches: 20090120__bug14263.diff.txt uploaded by Corydon76 (license 14) ........ 2009-01-22 16:52 +0000 [r170148] Joshua Colp * /, apps/app_meetme.c: Merged revisions 170147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists. (closes issue #14282) Reported by: cheesegrits ........ 2009-01-22 15:49 +0000 [r170112] Doug Bailey * channels/chan_dahdi.c, configure, include/asterisk/autoconfig.h.in, configure.ac: change VMWI to use new DAHDI_VMWI ioctl call. Change configure script to detect the new ioctl call data structure. (issue #14104) Reported by: alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded by dbailey (license ) Tested by: alecdavis, dbailey 2009-01-22 15:14 +0000 [r170047-170051] Joshua Colp * main/pbx.c, /: Merged revisions 170050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6 lines Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness. (closes issue #14011) Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga (license 665) ........ * apps/app_parkandannounce.c: Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop. (closes issue #14304) Reported by: jcovert 2009-01-22 02:10 +0000 [r170007] Richard Mudgett * channels/chan_dahdi.c: * Adjust some conditionals to balance curly braces. * Other minor changes. 2009-01-22 00:44 +0000 [r169944] Tilghman Lesher * /, include/asterisk/linkedlists.h: Merged revisions 169943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really wanted to ask is whether autoconf detected a static initializer value. This fixes rwlocks on all such platforms (mainly, Mac OS X). (closes issue #13767) Reported by: jcovert Patches: 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14) Tested by: jcovert, Corydon76 ........ 2009-01-22 00:23 +0000 [r169910] Richard Mudgett * channels/chan_dahdi.c: Whitespace changes only 2009-01-21 23:25 +0000 [r169869] Joshua Colp * main/pbx.c, /: Merged revisions 169867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4 lines Read lock the contexts to maintain the locking order when we are notified that the state of a device has changed. (closes issue #13839) Reported by: mcallist ........ 2009-01-21 23:20 +0000 [r169794-169866] Mark Michelson * channels/chan_dahdi.c: Test commit for test issue #14303 * main/say.c: Fix a crash when saying certain numbers in Chinese This commit fixes a crash that was occurring when attempting to say a number between 10000 and 100000 due to dividing by 0. This also removes some places where a "zero" is spoken when it should not be. (closes issue #14291) Reported by: dant Patches: say.c-14291.diff uploaded by dant (license 670) Tested by: dant 2009-01-21 22:04 +0000 [r169793] Michiel van Baak * doc/tex/extensions.tex: remove duplicated sentence. 2009-01-21 21:53 +0000 [r169791] Mark Michelson * channels/chan_sip.c: Further fix some oddities in sip show users and sip show peers logic ccesario on IRC pointed out that his sip peers were not displayed properly when he would issue the command "sip show peers." The problem was that the onlymatchonip field was used to determine if the endpoint was a "peer" or "user." The tricky part is that a "friend" is supposed to be treated as both a "user" and a "peer" but the logic would not allow "friends" to show up as "peers" since onlymatchonip was set to FALSE for friends. I have modified the sip_peer structure to more explicitly keep track of what type endpoint it is so that the various manager and CLI commands will display the expected information Reported by ccesario via IRC Tested by ccesario 2009-01-21 21:03 +0000 [r169723] Tilghman Lesher * /, main/asterisk.c: Merged revisions 169722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) | 8 lines Extra NULLs in the output cause some terminal types to abort in the middle of a color code, causing terminal weirdness. (closes issue #14130) Reported by: coolmig Patches: 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, coolmig ........ 2009-01-21 17:21 +0000 [r169673] Steve Murphy * utils/refcounter.c: This patch corrects a segfault reported in 14289, due to a null ptr being refd. Yes, seanbright is right in the bug comments, that is the fix. Sorry for this oversight; I guess my personal usage didn't have this happen! murf (closes issue #14289) Reported by: jamesgolovich 2009-01-21 10:49 +0000 [r169620-169625] Russell Bryant * /: Remove properties that erroneously got merged into trunk * main/tcptls.c: Fix a regression in TCP support. This patch fixes a problem that caused chan_sip to think that every open TCP session was to a remote address of 0.0.0.0:0. (closes issue #14287) Reported by: jamesgolovich Patches: bug-14287.diff.txt uploaded by jamesgolovich (license 176) 2009-01-21 00:33 +0000 [r169557-169611] Mark Michelson * apps/app_queue.c: Fix device state parsing issues for channel names with multiple slashes The fix being applied is a bit different for trunk and the 1.6.X branches. For trunk, we only wish to strip off the characters beyond the second slash if the channel is a Local channel (i.e. we are removing the /n from the device name). Other channel technologies with multiple slashes (e.g. DAHDI) need the information after the second slash in order to get the proper device state information. In addition to this fix, the 1.6.X branches are receiving a much more important fix as well. The problem in 1.6.X is that the member's device name was being directly changed instead of having a copy changed. This meant that we would strip off the second slash and trailing characters and then leave the member's device name like that permanently thereafter. (closes issue #14014) Reported by: kebl0155 Patches: 14014_number2.patch uploaded by putnopvut (license 60) Tested by: kebl0155 * apps/app_queue.c: Use the default timeout for a queue instead of -1 (closes issue #14272) Reported by: timking * /, channels/chan_sip.c: Convert the character pointers in a sip_request to be pointer offsets When an ast_str expands to hold more data, any pointers that were pointing to the data prior to the expansion will be pointing at invalid memory. This change makes such pointers used in chan_sip.c instead be offsets from the beginning of the string so that the same math may be applied no matter where in memory the string resides. To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added to chan_sip.c so that given a sip_request and an offset, the string at that offset is returned. (closes issue #14220) Reported by: riksta Tested by: putnopvut Review http://reviewboard.digium.com/r/126/ 2009-01-20 19:22 +0000 [r169486-169510] Terry Wilson * main/features.c: Make a proper builtin attended transfer to parking work This is an ugly hack from 1.4 that allows the timeout callback from a parked call to use the right channel name for the callback when the park is done with a builtin attended transfer (that isn't completed early). This hasn't ever worked in trunk and no one has complained yet, so eh. * /, main/features.c: Merged revisions 169485 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009) | 6 lines Don't play audio to the channel if we've masqueraded (closes issue #14066) Reported by: bluefox Tested by: otherwiseguy, bluefox ........ 2009-01-19 21:42 +0000 [r169438] Kevin P. Fleming * include/asterisk/res_odbc.h, funcs/func_odbc.c, include/asterisk/strings.h, res/res_odbc.c: ast_str_SQLGetData is *not* part of the ast_str API, it's part of the ast_odbc API and just happens to use an ast_str as the buffer; move all of it to res_odbc.c and res_odbc.h, renaming appropriately along the way fix some minor coding style issues in strings.h and add some attribute_pure annotations to functions in the ast_str API 2009-01-19 20:14 +0000 [r169367-169369] Michiel van Baak * main/asterisk.c: fix assignment in swapmode plug. Spotted and fix provided by ys (closes issue #14129) Reported by: ys Tested by: ys * channels/chan_skinny.c: Redo the event-based MWI in chan_skinny. Dan saw regular segfaults with the old implementation and rewrote it to make it really eventbased. I altered it to be trunk compatible and wedhorn gave some feedback and ideas how to make it even better. (closes issue #13821) Reported by: DEA Patches: chan_skinny-mwi-events.txt uploaded by DEA (license 3) Tested by: mvanbaak, DEA "no probs by me" from wedhorn 2009-01-19 20:05 +0000 [r169365] Tilghman Lesher * main/manager.c, /, apps/app_userevent.c: Merged revisions 169364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) | 4 lines Truncate userevents at the end of a line, when the command exceeds the buffer. (closes issue #14278) Reported by: fnordian ........ 2009-01-19 18:36 +0000 [r169327] Michiel van Baak * main/asterisk.c: Make asterisk compile on non-amd64 versions of OpenBSD. The HW_PHYSMEM64 is only available in latest OpenBSD and/or amd64 versions of OpenBSD. Use HW_PHYSMEM when HW_PHYSMEM64 is not available. (closes issue #14129) Reported by: ys Patches: 2009011600_physmem64.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak, jtodd 2009-01-19 18:22 +0000 [r169277-169325] Doug Bailey * channels/chan_dahdi.c: Get rid of magic number and replace with DAHDI_VMWI_NUMBER_MASK when determining the number of messages pending for MWI call * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add enhanced MWI generation to take advantage of new dahdi line reversal MWI ability. (closes issue #14104) Reported by: alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey (license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, dbailey 2009-01-19 15:54 +0000 [r169211] Mark Michelson * channels/chan_local.c, /: Merged revisions 169210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a potential NULL pointer dereference Move the check for if both channels on a local_pvt have generators to below where p->chan is checked for NULLity (NULLness?). This prevents a crash from occurring if p->chan is NULL. (closes issue #14189) Reported by: sascha Patches: 14189.patch uploaded by putnopvut (license 60) Tested by: sascha ........ 2009-01-17 18:26 +0000 [r169153] Doug Bailey * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add discriminator for when ring pulse alert signal is used to preface MWI spills This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up 2009-01-17 02:52 +0000 [r169116] Sean Bright * pbx/pbx_dundi.c: Change intializer types. Found while working on asterisk-cpp. I have a new favorite error message from g++: pbx_dundi.c:4580: sorry, unimplemented: non-trivial designated initializers not supported I like it when compilers are apologetic. 2009-01-17 01:56 +0000 [r169044-169080] Terry Wilson * main/tcptls.c, main/http.c, include/asterisk/tcptls.h: Fix qualify for TCP peer (closes issue #14192) Reported by: pabelanger Patches: asterisk-bug14192.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich * channels/chan_sip.c: Fix port :0 added to SIP INVITE URI when outboundproxy used (closes issue #14233) Reported by: chris-mac Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy 2009-01-16 22:43 +0000 [r168976] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168975 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines Account for possible NULL pointer when we receive a 408 in response to a REGISTER It may be that by the time we receive a reply to a REGISTER request, the attempt has timed out and thus the registry structure pointed to by the corresponding sip_pvt has gone away. This situation was handled properly for a 200 OK response, but the 408 case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash This commit fixes this assumption and prints out a message to the console if we should receive a late 408 response to a REGISTER (closes issue #14211) Reported by: aborghi Patches: 14211.diff uploaded by putnopvut (license 60) Tested by: aborghi ........ 2009-01-16 22:16 +0000 [r168941] Terry Wilson * /, main/features.c: Merged revisions 168716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) | 12 lines Convert call to park_call_full to masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded parking, otherwise we will try to call ast_hangup() in __pbx_run() and in do_parking_thread() and then promptly crash. (closes issue #14215) Reported by: waverly360 Tested by: otherwiseguy (closes issue #14228) Reported by: kobaz Tested by: otherwiseguy ........ 2009-01-16 19:54 +0000 [r168898] Mark Michelson * res/res_timing_timerfd.c: Fix a logic error that occur when using the timerfd interface This sequence of events posed a problem timerfd_timer_open timerfd_timer_enable_continuous timerfd_timer_set_rate timerfd_timer_disable_continuous The reason was that the timing module was written under the assumption that timerfd_timer_set_rate would not be called between enabling and disabling continuous mode. What happened in this situation was that timerfd_timer_enable_continuous saved off our previously set timer (in this situation a 0 timer, meaning it never runs out). Then timerfd_timer_disable_continuous would restore this 0 timer, even though it logically should set the timer to be whatever was set in timerfd_timer_set_rate. Now the behavior in timerfd_timer_set_rate is to overwrite the saved timer that may or may not have been set in timerfd_timer_enable_continuous. Even if timerfd_timer_enable_continuous has not been previously called, this will not harm the operation. Thanks to Terry Wilson for discovering the problem and giving me a really great debug capture that pointed out the problem clearly 2009-01-16 18:49 +0000 [r168832] Tilghman Lesher * /, main/say.c, include/asterisk/say.h, apps/app_voicemail.c: Merged revisions 168828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) ........ 2009-01-16 17:09 +0000 [r168759-168760] Russell Bryant * CHANGES: Fix a spelling mistake. * channels/chan_misdn.c: build in dev mode 2009-01-16 00:34 +0000 [r168737-168746] Steve Murphy * res/ael/pval.c, /: Merged revisions 168745 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | 14 lines This patch fixes a problem where a goto (or jump, in this case) fails a consistency check because it can't find a matching extension. The problem was a missing instruction to end the range notation in the code where it converts the pattern into a regex and uses the regex code to determine the match. I tested using the AEL code the user supplied, and now, the consistency check passes. (closes issue #14141) Reported by: dimas ........ * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: This patch allows null args in ast_expr2 func calls, and fixes commas being converted to pipes, which was 1.4 type stuff. If the user says count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); then it won't complain about the empty arg (c,,...) and fabled's patch won't let it swap the commas for pipes. Ran it thru my dialplan and no complaints. (closes issue #14169) Reported by: fabled Patches: function-argument-separator-fix.diff uploaded by fabled (license 448) 2009-01-15 20:18 +0000 [r168734] Kevin P. Fleming * res/res_config_odbc.c, build_tools/menuselect-deps.in, configure, funcs/func_odbc.c, configure.ac, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, makeopts.in, res/res_odbc.c, apps/app_voicemail.c: remove the PBX_ODBC logic from the configure script, and add GENERIC_ODCB logic that includes copying the relevant LIB and INCLUDE data from either UnixODBC or iODBC, based on which was found; if both were found, prefer UnixODBC this stops modules from being linked against both sets of libraries on systems that have both installed 2009-01-15 20:00 +0000 [r168725-168732] Mark Michelson * channels/chan_sip.c: Add missing brace * channels/chan_sip.c: Fix the compactheaders option in sip.conf * channels/chan_sip.c: Remove an unneeded condition for line addition to a SIP request/response In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically allocated buffer to hold the text of the request. There was a check in the add_line function to not attempt to write the line into the buffer if we did not have room for it. In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str structure is used to hold the text. Since it may grow to fit an arbitrarily sized string, this check in add_line is no longer valid. I found this oddity while attempting to fix issue #14220; however, I do not believe that this is the fix for that issue since the output supplied by the reporter did not contain the warning message that would be printed had this condition been satisfied. 2009-01-15 18:47 +0000 [r168722] Olle Johansson * /, configs/extconfig.conf.sample: Merged revisions 168721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 lines Meetme actually has realtime but wasn't documented ........ 2009-01-15 18:39 +0000 [r168719] Tilghman Lesher * include/asterisk/strings.h: Resolve issue with negative vs non-negative length parameters. (closes issue #14245) Reported by: dveiga 2009-01-15 18:08 +0000 [r168711-168712] Olle Johansson * channels/chan_sip.c: Make sure that we have the same terminology in sip.conf.sample and the source code warning. Thanks Nick Lewis for pointing this out in the bug tracker. * configs/sip.conf.sample: Clarify some misunderstandings and make it even more clear that you can refer to a peer in the register= line. 2009-01-15 15:33 +0000 [r168705] Sean Bright * apps/app_meetme.c: Add a missing unlock and properly handle the 'maxusers' setting on MeetMe conferences. We were using the 'user number' field to compare against the maximum allowed users, which works assuming users with lower user numbers didn't leave the conference. (closes issue #14117) Reported by: sergedevorop Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright (license 71) Tested by: sergedevorop 2009-01-15 13:37 +0000 [r168636-168639] Olle Johansson * CREDITS, CHANGES: Related to issue #14246 Update changes for SIPRemoveHeader() * channels/chan_sip.c: Add capability to remove added SIP headers *before* INVITE is generated. (closes issue #14246) Reported by: klaus3000 Patches: 2patch_chan_sip_SIPRemoveHeader_trunk.txt uploaded by klaus3000 (license 65) * apps/app_queue.c: Add support for setting the Reason header when cancelling a call in the queue because someone else answered. Previously, only dial() was supported. EDV-102 2009-01-15 00:14 +0000 [r168629] Mark Michelson * /, apps/app_queue.c: Merged revisions 168628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines Fix some crashes from bad datastore handling in app_queue.c * The queue_transfer_fixup function was searching for and removing the datastore from the incorrect channel, so this was fixed. * Most datastore operations regarding the queue_transfer datastore were being done without the channel locked, so proper channel locking was added, too. (closes issue #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by putnopvut (license 60) Tested by: ZX81, festr ........ 2009-01-14 23:10 +0000 [r168626] Sean Bright * main/cli.c: Don't crash when typing 'core set verbose' or 'core set debug' by themselves. (closes issue #14219) Reported by: jamesgolovich Patches: asterisk-setverbosecrash.diff.txt uploaded by jamesgolovich (license 176) 2009-01-14 21:51 +0000 [r168623] Richard Mudgett * /, channels/misdn/isdn_lib.c: Merged revisions 168622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009) | 4 lines * Fixed create_process() allocation of process ID values. The allocated process IDs could overflow their respective NT and TE fields. Affects outgoing calls. ........ 2009-01-14 21:19 +0000 [r168619] Doug Bailey * channels/chan_dahdi.c: This fixes a problem where MWI FSK spills were being injected onto off hook fxs lines. (closes issue #14143) Reported by: alecdavis Patches: chan_dahdi-14143.patch.txt uploaded by dbailey (license ) Tested by: alecdavis 2009-01-14 20:58 +0000 [r168615] Sean Bright * /, contrib/scripts/autosupport: Merged revisions 168614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan 2009) | 9 lines Update autosupport script to supply info for both Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x and trunk instead of zttest. (closes issue #14132) Reported by: dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded by dsedivec (license 638) ........ 2009-01-14 20:51 +0000 [r168613] Steve Murphy * /, apps/app_page.c: Merged revisions 168608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 line app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning. ........ 2009-01-14 20:13 +0000 [r168610] Mark Michelson * channels/chan_sip.c: Restore the "sip show users" and "sip show user" CLI commands (closes issue #14180) Reported by: amorsen Patches: sip_show_users_161v3.diff uploaded by putnopvut (license 60) Tested by: blitzrage, amorsen 2009-01-14 19:36 +0000 [r168609] Michiel van Baak * main/asterisk.c: Fix compilation on FreeBSD and OSX This started as work to fix the 'core show sysinfo' CLI command but while working on it oej pointed out that read_credentials did not compile neither. So while being there, fix that as well. Thanks for all the testing oej! (closes issue #14129) Reported by: ys Tested by: oej, mvanbaak 2009-01-14 19:11 +0000 [r168601-168604] Tilghman Lesher * main/udptl.c, /: Merged revisions 168603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) | 7 lines Don't read into a buffer without first checking if a value is beyond the end. (closes issue #13600) Reported by: atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76 (license 14) Tested by: atis ........ * channels/chan_misdn.c: Mostly spacing changes; no functionality change at all. 2009-01-14 02:00 +0000 [r168594] Terry Wilson * /, apps/app_page.c: Merged revisions 168593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines Don't overflow when paging more than 128 extensions The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbitrary limit. This issue would have special importance to anyone who is dynamically creating the argument passed to the Page application and allowing more than 128 extensions to be added by an outside user via some external interface. The patch posted by a_villacis was slightly modified for some coding guidelines and other cleanups. Thanks, a_villacis! (closes issue #14217) Reported by: a_villacis Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660) Tested by: otherwiseguy ........ 2009-01-13 23:57 +0000 [r168591] Tilghman Lesher * channels/chan_misdn.c: Janitor patch for chan_misdn (make channel variable access safe) (closes issue #12887) Reported by: pputman Patches: chan_misdn_threadsafe.patch uploaded by pputman (license 81) 2009-01-13 23:05 +0000 [r168585-168588] Terry Wilson * res/res_http_post.c: Fully overwrite a same-named file when uploading (closes issue #14190) Reported by: timking * Makefile, include/asterisk/options.h, main/asterisk.c: Add option to hide console connect messages (closes issue #14222) Reported by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded by jamesgolovich (license 176) Tested by: otherwiseguy 2009-01-13 22:30 +0000 [r168579] Mark Michelson * apps/app_queue.c: Clarify a message that app_queue prints and change to a debug-level message The "No one is answering..." verbose message contained 3 numbers that were not explained in any way to whoever was viewing the message. It is more helpful now since the message explains what the numbers mean. Also, the message has been downgraded to "DEBUG" level. (closes issue #14172) Reported by: caio1982 Patches: queue_answering_debug.diff uploaded by caio1982 (license 22) 2009-01-13 22:22 +0000 [r168578] Terry Wilson * /, channels/chan_sip.c: Merged revisions 168551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009) | 7 lines Don't pass a value with a side effect to a macro (closes issue #14176) Reported by: paraeco Patches: chan_sip.c.diff uploaded by paraeco (license 658) ........ 2009-01-13 21:18 +0000 [r168575] Mark Michelson * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Allow specifying a port number in the user portion of a register => line in sip.conf With this commit, a register => line in sip.conf may contain a port number in the "user" section of the line. Please see CHANGES and sip.conf.sample for more details regarding this. (closes issue #14198) Reported by: Nick_Lewis Patches: chan_sip.c-domainport2.patch uploaded by Nick (license 657) Tested by: Nick_Lewis 2009-01-13 19:22 +0000 [r168562] Russell Bryant * channels/chan_unistim.c, main/pbx.c, apps/app_read.c, /, include/asterisk/indications.h, apps/app_readexten.c, apps/app_disa.c, include/asterisk/channel.h, main/indications.c, main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_channel.c, main/app.c, res/snmp/agent.c, res/res_indications.c: Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ 2009-01-13 17:51 +0000 [r168547] Tilghman Lesher * /, funcs/func_logic.c: Merged revisions 168546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009) | 6 lines If either conditional is NULL, don't try copying it. (closes issue #14226) Reported by: caspy Patches: 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14) ........ 2009-01-13 16:02 +0000 [r168539] Dwayne M. Hubbard * main/taskprocessor.c: correct a CLI description 2009-01-12 23:45 +0000 [r168526] Tilghman Lesher * /, channels/chan_alsa.c: Merged revisions 167095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 Dec 2008) | 5 lines Repeat attempts to write when we receive -EAGAIN from the driver, as detailed in the ALSA sample code (see http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32) Reported by: Jerry Geis (via the -users list) Fixed by: me (license 14) ........ 2009-01-12 23:12 +0000 [r168523] Mark Michelson * main/srv.c: bump the verbosity of a message in srv.c up by one. It used to be at this level prior to a large patch merge which converted ast_verbose calls to ast_verb (closes issue #14221) Reported by: jcovert Patches: srv.c.patch uploaded by jcovert (license 551) 2009-01-12 23:06 +0000 [r168522] Tilghman Lesher * configure, include/asterisk/autoconfig.h.in, configure.ac, main/app.c: Some platforms (notably, the BSDs) have a more efficient implementation called closefrom(3). 2009-01-12 21:51 +0000 [r168508-168517] Jeff Peeler * /, res/res_agi.c: Merged revisions 168516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) | 5 lines (closes issue #13881) Reported by: hoowa Update the app CDR field for AGI commands that are not executing an application via "exec". ........ * /, channels/chan_agent.c: Merged revisions 168507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG Tested by: denisgalvao This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock. Review: http://reviewboard.digium.com/r/35/ ........ 2009-01-12 16:31 +0000 [r168497] Olle Johansson * apps/app_minivm.c: Better to use the proper app name 2009-01-12 15:00 +0000 [r168485] Mark Michelson * channels/chan_sip.c: Merged revisions 168482 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan 2009) | 5 lines I am reverting the fix made in revision 168128 (and its upward merges) after being contacted by Olle Johansson and being shown how this fix is incorrect. Thanks to Olle for clearing this up for me. ........ 2009-01-12 14:57 +0000 [r168481] Russell Bryant * /, configs/indications.conf.sample: Merged revisions 168480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines s/ringdance/ringcadence/ for Bulgaria ........ 2009-01-12 14:35 +0000 [r168479] Olle Johansson * main/asterisk.c: Don't include swap.h unless we have swapctl 2009-01-10 01:42 +0000 [r168334] Tilghman Lesher * channels/chan_sip.c: sizeof for a stringfield is 4. Kinda low for reconstructing a field value. 2009-01-09 23:16 +0000 [r168270] Kevin P. Fleming * /, sounds/Makefile: Merged revisions 168267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan 2009) | 1 line update to use new sound file packages that include license files ........ 2009-01-09 23:15 +0000 [r168269] Richard Mudgett * channels/chan_misdn.c: Spacing change 2009-01-09 23:04 +0000 [r168265] Michiel van Baak * contrib/scripts/sip_nat_settings (added), CHANGES: Add a script to find out the correct settings for Asterisk behind NAT (closes issue #13065) Reported by: tzafrir Patches: sip_nat_settings uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and moi 2009-01-09 22:21 +0000 [r168200] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 168198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 Jan 2009) | 2 lines Make this compile for mvanbaak ........ 2009-01-09 21:53 +0000 [r168193] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168128 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan 2009) | 13 lines Add check_via calls to more request handlers INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not checking the topmost Via to determine where to send the response. Adding check_via calls to those request handlers solves this. (closes issue #13071) Reported by: baron Patches: check_via.patch uploaded by baron (license 531) Tested by: baron ........ 2009-01-09 21:43 +0000 [r168192] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 168191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 * Miscellaneous doxygen comments added. ........ 2009-01-09 20:25 +0000 [r168142] Terry Wilson * res/res_phoneprov.c: Don't leak memory if phoneprov.conf does not exist (closes issue #14203) Reported by: jamesgolovich Patches: asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich (license 176) 2009-01-09 18:30 +0000 [r168090] Tilghman Lesher * res/res_agi.c, include/asterisk/strings.h: When using ast_str with a non-ast_str-enabled API, we need to update the buffer or otherwise, we cannot use ast_str_strlen(). 2009-01-09 18:01 +0000 [r168014-168054] Matthew Nicholson * main/logger.c: Added a comment to logger.c about where to put includes * main/logger.c: Use ast_safe_system() in logger.c instead of system() (closes issue #14194) Reported by: pabelanger 2009-01-09 01:15 +0000 [r167935-167973] Terry Wilson * apps/app_originate.c: Set ORIGINATE_STATUS instead of OUTGOING_STATUS to match the documentation * apps/app_dial.c: Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set 2009-01-08 22:37 +0000 [r167894] Tilghman Lesher * /, res/res_agi.c: Merged revisions 167840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009) | 6 lines Don't truncate database results at 255 chars. (closes issue #14069) Reported by: evandro Patches: 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14) ........ 2009-01-08 22:34 +0000 [r167888] Mark Michelson * channels/chan_sip.c: Revert chan_sip changes which were accidentally committed in revision 167792 2009-01-08 21:40 +0000 [r167835-167837] Tilghman Lesher * apps/app_minivm.c: Fix variables to comply with documentation changes * apps/app_minivm.c: Textual changes, consistency in status variable naming, and other minor bugs. (closes issue #13943) Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded by Marquis (license 32) 2009-01-08 19:48 +0000 [r167792] Mark Michelson * channels/chan_sip.c, CHANGES, apps/app_queue.c: Add the average talk time for a queue This patch adds the functionality to app_queue of calculating the average amount of time that channels are bridged for a queue. The algorithm used to calculate the average is the same exponential average currently used to calculate the average holdtime. See the CHANGES file to see the methods you may use to view this information. (closes issue #13960) Reported by: coolmig Patches: app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621) 2009-01-08 19:44 +0000 [r167791] Tilghman Lesher * channels/chan_dahdi.c, CHANGES: Convert dialplan application DAHDISendCallreroutingFacility to use commas. (closes issue #13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded by eliel (license 64) 2009-01-08 17:26 +0000 [r167700-167720] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 167714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan 2009) | 1 line remove an unnecessary argument to queue_request() ........ * channels/chan_sip.c: Merged revisions 167620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan 2009) | 5 lines When a SIP request or response arrives for a dialog with an associated Asterisk channel, and the lock on that channel cannot be obtained because it is held by another thread, instead of dropping the request/response, queue it for later processing when the channel lock becomes available. http://reviewboard.digium.com/r/123/ ........ 2009-01-08 14:27 +0000 [r167662] Leif Madsen * contrib/scripts/sip-friends.sql: Oops... fix the fieldname I changed yesterday to be right. 2009-01-07 22:36 +0000 [r167542-167569] Russell Bryant * /, main/file.c: Merged revisions 167566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) | 2 lines Fix the last couple of places where free() was improperly used directly. ........ * /, main/file.c: Merged revisions 167554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) | 2 lines Don't fclose() the file early, the filestream destructor will handle it. ........ * /, main/file.c: Merged revisions 167545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) | 2 lines Only try to close the file if one was actually opened ........ * /, main/file.c: Merged revisions 167541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) | 4 lines Don't use free() directly. This caused a crash since ast_filestream is now an ao2 object. Reported by JunK-Y on IRC, #asterisk-dev ........ 2009-01-07 18:20 +0000 [r167478] BJ Weschke * apps/app_followme.c: Answer the channel if it has not already been answered and we've already found a valid profile for followme. (closes issue #14140) Reported by: dimas Patches: 14140.patch uploaded by dimas 2009-01-07 18:18 +0000 [r167477] Leif Madsen * configs/queues.conf.sample: Update queues.conf.sample documentation. Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so. (closes issue #14179) Reported by: CrashHD Tested by: CrashHD 2009-01-07 17:35 +0000 [r167442] Russell Bryant * /, main/indications.c: Merged revisions 167432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009) | 4 lines Treat an empty string the same way as a NULL country argument. In passing, simplify the handling of returning a default tone zone. ........ 2009-01-07 17:05 +0000 [r167416] Doug Bailey * channels/chan_dahdi.c: Cleanup fsk spill if off hook is detected during mwi spill. Correct logic error in handling events when sending mwi spill (closes issue #14143) Reported by: alecdavis Patches: chan_dahdi.handle_init_event2.diff.txt uploaded by dbailey 2009-01-07 14:26 +0000 [r167373] Leif Madsen * contrib/scripts/sip-friends.sql: Update the sip-friends.sql file to use the non-deprecated 'defaultname' instead of 'username' and remove an extra comma that would cause the script to fail as-is 2009-01-06 21:36 +0000 [r167301] Mark Michelson * /, main/db.c: Merged revisions 167299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan 2009) | 8 lines Use the correct variable when creating the format string (closes issue #14177) Reported by: nic_bellamy Patches: asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic (license 299) ........ 2009-01-06 21:02 +0000 [r167265] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 167260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r167260 | tilghman | 2009-01-06 14:48:05 -0600 (Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 Jan 2009) | 2 lines Security fix AST-2009-001. ........ ................ 2009-01-05 16:59 +0000 [r167180] Mark Michelson * /, channels/chan_sip.c: Merged revisions 167179 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan 2009) | 41 lines A couple of changes to T.38 SDP attribute handling There are some boolean attributes for T.38 such as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and T38FaxTranscodingJBIG. By simply being present, we should treat these as a "true" value. The current code, however, was requiring a 1 or 0 as the value of the attribute in order to parse it. This is due to the fact that there are some T.38 endpoints and gateways that also transmit this information incorrectly. This patch follows the "be liberal in what you accept and strict in what you send" philosophy by accepting both the correctly- and incorrectly-formatted attributes, but only sending information as it is supposed to be sent. It was also discovered that a particular type of T.38 gateway sends some non-standard T.38 SDP attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate, it used T38MaxDatagram and T38FaxMaxRate respectively. We now will properly accept these attributes as well. Note that there are a lot of patches cited in the below commit message template. This is because the person who submitted these patches is an awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes issue #13976) Reported by: linulin Patches: chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648) Tested by: arcivanov ........ 2009-01-05 16:44 +0000 [r167176] Tilghman Lesher * UPGRADE-1.6.txt: More clearly explain that quote marks are no longer necessary. (closes issue #13718) Reported by: davidw Patches: 20081020__bug13718.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage 2009-01-03 20:29 +0000 [r167125] Jeff Peeler * main/asterisk.c: When parsing environment variable ASTERISK_PROMPT, make sure to proceed to the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit. 2008-12-31 23:07 +0000 [r167061] Sean Bright * doc/CODING-GUIDELINES, include/asterisk.h, channels/h323/README: Mostly just whitespace, but also convert 'CVS' to 'SVN' in a couple places and fix a few typos I found in the CODING_GUIDELINES. 2008-12-31 22:53 +0000 [r167057] Terry Wilson * main/xmldoc.c: Don't forget to free typename 2008-12-31 21:52 +0000 [r167021] Mark Michelson * channels/chan_dahdi.c: Change some incorrect syntax for pri set debug and correct an off-by-one error in ss7 set debug command 2008-12-31 19:39 +0000 [r166954-166958] Tilghman Lesher * main/ast_expr2.h, main/ast_expr2.c: That was weird... * channels/chan_local.c, /, main/ast_expr2.h, main/ast_expr2.c: Merged revisions 166953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) | 5 lines Also inherit the musiconhold class. (Closes #14153) Reported by: Jerry Geis, via the users list. Patch by: me (license 14) ........ 2008-12-30 20:50 +0000 [r166908] Terry Wilson * res/res_phoneprov.c, doc/sip-retransmit.txt, doc/tex/phoneprov.tex, res/res_http_post.c, phoneprov/polycom_line.xml, doc/realtimetext.txt: Fix some svn:keywords 2008-12-29 18:04 +0000 [r166861] Mark Michelson * apps/app_dial.c, apps/app_queue.c: Update app_queue to deal with the removal of AST_PBX_KEEPALIVE When placing a call to a queue which ran a gosub on the member's channel, Asterisk would crash every time, stemming from the fact that the member's channel was being hung up unexpectedly when the Gosub completed. The necessary change was pretty much copied and pasted from app_dial's similar changes made last week. I also took the opportunity to change a LOG_DEBUG message in app_dial to use ast_debug. I am guessing this was due to a direct merge from 1.4 that was not corrected to use trunk's preferred syntax. 2008-12-28 15:36 +0000 [r166823] Eliel C. Sardanons * funcs/func_audiohookinherit.c: Fix a typo in the XML documentation of the AUDIOHOOK_INHERIT dialplan function. 2008-12-28 15:15 +0000 [r166773] Russell Bryant * /, channels/misdn_config.c: Merged revisions 166772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) | 4 lines Use strncat() instead of an sprintf() in which source and target buffers overlap http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html ........ 2008-12-24 15:10 +0000 [r166731] Terry Wilson * channels/chan_sip.c: There is no section 22.2.2 in rfc 3261. I believe 26.2.2 is what was meant: Note that in the SIPS URI scheme, transport is independent of TLS, and thus "sips:alice@atlanta.com;transport=tcp" and "sips:alice@atlanta.com;transport=sctp" are both valid (although note that UDP is not a valid transport for SIPS). The use of "transport=tls" has consequently been deprecated, partly because it was specific to a single hop of the request. This is a change since RFC 2543. 2008-12-23 20:47 +0000 [r166696] Tilghman Lesher * channels/chan_sip.c: Allow semicolons and extended characters in user-specified SIP headers. (closes issue #14110) Reported by: gork Patches: 20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14) Tested by: gork, putnopvut 2008-12-23 18:13 +0000 [r166665] Steve Murphy * apps/app_dial.c, main/pbx.c, /, main/features.c, apps/app_macro.c, include/asterisk/pbx.h, apps/app_queue.c, include/asterisk/features.h: Merged revisions 166093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ 2008-12-23 16:04 +0000 [r166625] Russell Bryant * CHANGES: Fix spelling error. 2008-12-23 15:17 +0000 [r166569] Mark Michelson * main/channel.c, /: Merged revisions 166568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines Fix a crash resulting from a datastore with inheritance but no duplicate callback The fix for this is to simply set the newly created datastore's data pointer to NULL if it is inherited but has no duplicate callback. (closes issue #14113) Reported by: francesco_r Patches: 14113.patch uploaded by putnopvut (license 60) Tested by: francesco_r ........ 2008-12-23 04:32 +0000 [r166533] Tilghman Lesher * main/channel.c, /: Merged revisions 166509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines Use the integer form of condition for integer comparisons. (closes issue #14127) Reported by: andrew ........ 2008-12-22 23:25 +0000 [r166470] Mark Michelson * res/res_agi.c: Always use the value of the AGISIGHUP when running an AGI. Prior to this patch, the value of AGISIGUP was not always honored when set on a channel. (closes issue #13711) Reported by: fmueller Patches: 13711.patch uploaded by putnopvut (license 60) 2008-12-22 21:45 +0000 [r166436] Russell Bryant * res/res_musiconhold.c: Cosmetic change - don't mix struct initializer styles. 2008-12-22 21:08 +0000 [r166382] Mark Michelson * channels/chan_dahdi.c, /: Merged revisions 166380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks and autoservice It has been discovered that if a channel is locked prior to a call to ast_autoservice_stop, then it is likely that a deadlock will occur. The reason is that the call to ast_autoservice_stop has a check built into it to be sure that the thread running autoservice is not currently trying to manipulate the channel we are about to pull out of autoservice. The autoservice thread, however, cannot advance beyond where it currently is, though, because it is trying to acquire the lock of the channel for which autoservice is attempting to be stopped. The gist of all this is that a channel MUST NOT be locked when attempting to stop autoservice on the channel. In this particular case, the channel was locked by a call to ast_read. A call to ast_exists_extension led to autoservice being started and stopped due to the existence of dialplan switches. It may be that there are future commits which handle the same symptoms but in a different location, but based on my looks through the code, it is very rare to see a construct such as this one. (closes issue #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded by putnopvut (license 60) Tested by: rtrauntvein Review: http://reviewboard.digium.com/r/107/ ........ 2008-12-22 20:26 +0000 [r166273-166377] Russell Bryant * res/res_musiconhold.c: Fix a bad typo. * main/astobj2.c: Remove some error messages. This is the default handler that is valid to use. * /, main/utils.c: Merged revisions 166297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) | 2 lines Fix up timeout handling in ast_carefulwrite(). ........ * include/asterisk/utils.h, main/manager.c, main/utils.c: Introduce ast_careful_fwrite() and use in AMI to prevent partial writes. This patch introduces a function to do careful writes on a file stream which will handle timeouts and partial writes. It is currently used in AMI to address the issue that has been reported. However, there are probably a few other places where this could be used. (closes issue #13546) Reported by: srt Tested by: russell http://reviewboard.digium.com/r/104/ * res/res_musiconhold.c: Re-work ref count handling of MoH classes using astobj2 to resolve crashes. (closes issue #13566) Reported by: igorcarneiro Tested by: russell Review: http://reviewboard.digium.com/r/106/ 2008-12-22 16:08 +0000 [r166268] Joshua Colp * main/dnsmgr.c: Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port. (closes issue #13628) Reported by: pananix Patches: bug13628.patch uploaded by jpeeler (license 325) Tested by: file, blitzrage 2008-12-22 16:07 +0000 [r166267] Mark Michelson * funcs/func_timeout.c, main/file.c: Fix a file playback crash and explicitly initialize values in func_timeout.c A crash was brought up on the bugtracker. The first run through valgrind was full of legitimate complaints of uninitialized values in func_timeout when setting a response timeout. These were fixed but the crash persisted. A second run through showed the real problem. The reference counting used for filestreams was incorrect because there were some missing increments when a frame was read from a format module. (closes issue #14118) Reported by: blitzrage Patches: 14118v2.patch uploaded by putnopvut (license 60) Tested by: blitzrage 2008-12-22 14:16 +0000 [r166258] Russell Bryant * res/res_agi.c: Remove AST_PBX_KEEPALIVE usage from res_agi. This patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The only usage was for the AGI command, "asyncagi break". This patch removes this feature. Normally, a feature would not be removed like this. However, this code is broken and usage of it will result in a memory leak. Usage of this feature will make the AGI code return a result of AST_PBX_KEEPALIVE. The PBX handler assumes that another thread has assumed ownership of the channel. The channel thread will exit without destroying the channel. Unfortunately, _no_ thread has ownership of the channel at this point. There are a couple of serious problems here: 1) The only way to recover the caller is to issue a channel redirect. This will work, but this will be done with a masquerade, and the old ast_channel structure will be lost. 2) Until the channel redirect happens, there is no code servicing the channel. That means nothing is reading audio or handling events coming from the channel. This is very bad. The recommended way to get this same "break" functionality is to issue the redirect while the channel is still being handled by the AGI code. That way, there will be no memory leak, and there will be no period of time that the channel is not being serviced. 2008-12-20 01:37 +0000 [r166219] Russell Bryant * include/asterisk/doxyref.h: Make a note about formatting the review URL in commit messages 2008-12-19 23:45 +0000 [r166092-166162] Mark Michelson * main/audiohook.c: Get rid of an extra space. I don't know how this crept back in when I had already fixed it earlier * funcs/func_audiohookinherit.c: Remove the verbatim tag from the author line I could have sworn I already did that before, though... * main/channel.c, funcs/func_audiohookinherit.c (added), include/asterisk/audiohook.h, main/audiohook.c, CHANGES: Adding a new dialplan function AUDIOHOOK_INHERIT This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ 2008-12-19 21:44 +0000 [r166058] Matthew Fredrickson * channels/chan_dahdi.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Add configuration support for half_full DAHDI buffer policy 2008-12-19 18:20 +0000 [r165954] Eliel C. Sardanons * apps/app_record.c: Fix the XML documentation for Record(). tags inside elements must have CDATA and no another XML node. 2008-12-19 15:05 +0000 [r165801-165890] Russell Bryant * /, apps/app_chanspy.c: Merged revisions 165889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines Ensure that the chanspy datastore is fully initialized. This patch resolved some random crash issues observed by a user on a BSD system (closes issue #14111) Reported by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) ........ * include/asterisk/doxyref.h: Disable some automatic links generated by doxygen. * include/asterisk/doxyref.h: Introduce commit message formatting guidelines. This documents the recommended outline to use for commit message. It also covers information on special tags that can be used in commit messages, as well as the template functionality that is available on bugs.digium.com. Review: http://reviewboard.digium.com/r/96/ * /, main/utils.c: Merged revisions 165796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines Make ast_carefulwrite() be more careful. This patch handles some additional cases that could result in partial writes to the file description. This was done to address complaints about partial writes on AMI. (issue #13546) (more changes needed to address potential problems in 1.6) Reported by: srt Tested by: russell Review: http://reviewboard.digium.com/r/99/ ........ 2008-12-18 21:43 +0000 [r165798] Jeff Peeler * main/manager.c: (closes issue #13993) Reported by: mika Add ActionID response to ping if sent with request. 2008-12-18 21:41 +0000 [r165797] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 165767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) | 8 lines Add mutexes around accesses to the IMAP library interface. This prevents certain crashes, especially when shared mailboxes are used. (closes issue #13653) Reported by: howardwilkinson Patches: asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590) Tested by: jpeeler ........ 2008-12-18 21:21 +0000 [r165792] Joshua Colp * channels/chan_dahdi.c, channels/chan_misdn.c, channels/chan_sip.c, pbx/pbx_ael.c, apps/app_queue.c, channels/chan_oss.c: Numerous documentation updates. (closes issue #13970) Reported by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10) 2008-12-18 19:34 +0000 [r165724] Mark Michelson * res/res_odbc.c: Fix crashes in res_odbc. The variable "class" was being set NULL just prior to being dereferenced in an ao2_link call. I have moved the setting of the variable to NULL until after the ao2_link call. 2008-12-18 19:33 +0000 [r165662-165723] Russell Bryant * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ * /, res/res_musiconhold.c: Merged revisions 165661 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008) | 7 lines Set the process group ID on the MOH process so that all children will get killed (closes issue #14099) Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license 645) ........ 2008-12-18 18:36 +0000 [r165658] Tilghman Lesher * apps/app_voicemail.c: Fix 2 resource leaks and fix another pipe-to-comma conversion 2008-12-18 17:13 +0000 [r165599] Joshua Colp * /, main/rtp.c: Merged revisions 165591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us. (closes issue #13545) Reported by: davidw ........ 2008-12-18 16:36 +0000 [r165541] Tilghman Lesher * res/res_odbc.c: Fix reference counts of the class and add an assertion to the end. 2008-12-18 15:25 +0000 [r165502] Eliel C. Sardanons * main/strings.c, include/asterisk/strings.h: Remove duplicate code from the ast_str API. We now use __AST_STR_* to access 'struct ast_str' members, but this must only be used inside the API implementation. (closes issue #14098) Reported by: eliel Patches: ast_str.patch uploaded by eliel (license 64) 2008-12-18 14:23 +0000 [r165433-165469] Russell Bryant * apps/app_originate.c: Add a \todo note for app_originate. Jared Smith suggested that we add a way to be able to set variables and functions on the outbound channel. I think that it's a great idea, so I have added it as a todo so that it gets done at some point. * apps/app_originate.c (added), CHANGES: Add a new application, Originate. (closes issue #14075) Reported by: rcasas Patches: app_originate.c uploaded by rcasas (license 641), heavily modified by me Tested by: russell Review: http://reviewboard.digium.com/r/95/ 2008-12-17 23:39 +0000 [r165397] Tilghman Lesher * apps/app_record.c: Add RECORD_STATUS variable, as requested on the -users list. Patch by me (license 14) 2008-12-17 21:46 +0000 [r165326-165330] Mark Michelson * res/res_odbc.c: Fix a refcount leak in res_odbc * apps/app_meetme.c, res/res_realtime.c: Fix the build 2008-12-17 21:28 +0000 [r165319-165325] Tilghman Lesher * apps/app_macro.c: Oops, broke trunk * /, apps/app_macro.c: Merged revisions 165317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) | 4 lines Reverse the fix from issue #6176 and add proper handling for that issue. (Closes issue #13962, closes issue #13363) Fixed by myself (license 14) ........ 2008-12-17 21:17 +0000 [r165318] Mark Michelson * apps/app_meetme.c, res/res_realtime.c, apps/app_directory.c, apps/app_queue.c, apps/app_voicemail.c: Merged revisions 165255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines Fix some memory leaks found while looking at how realtime configs are handled. Also cleaned up some coding guidelines violations in app_realtime.c, mostly related to spacing ........ 2008-12-17 20:50 +0000 [r165254] Steve Murphy * utils/extconf.c: This patch is here committed to satisfy the buildbot, who has a problem with the const. 2008-12-17 19:55 +0000 [r165219] Terry Wilson * res/res_phoneprov.c: Polycom phones close the connection after reading a little bit of the firmware files, we should stop sending in that case. Also, make that case print out a debug statement instead of a scary WARNING. 2008-12-17 19:52 +0000 [r165216] Joshua Colp * channels/chan_sip.c: Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly! (closes issue #14055) Reported by: chris-mac 2008-12-17 18:49 +0000 [r165180] Matthew Nicholson * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc 2008-12-17 17:56 +0000 [r165145] Russell Bryant * doc/appdocsxml.dtd: argsep is used as an attribute for an argument, as well 2008-12-17 17:53 +0000 [r165142-165143] Mark Michelson * apps/app_voicemail.c: And actually assign the function to a pointer... * apps/app_voicemail.c: Use the create_vm_state_from_user function in a place where it was not being used before. Also, I've moved the urgent folder check in messagecount() up a bit so that the flow is a bit better. This was something I noticed while taking a look at issue #13973, although I don't think this is the underlying cause of the issue. 2008-12-17 16:41 +0000 [r165108] Kevin P. Fleming * utils: ignore this copied file 2008-12-17 05:04 +0000 [r165039-165071] Steve Murphy * utils/Makefile, pbx/pbx_ael.c, utils/ael_main.c, utils/extconf.c, utils/conf2ael.c, utils/check_expr.c: A possibly "horrible fix" for a "horribly broken" situation. As stuff shifts around in the asterisk code, the miscellaneous inclusions from the standalone stuff gets broken. There's no easy fix for this situation. I made sure that everything in utils builds without problem ***AND*** that aelparse runs the regressions correctly with the following make menuselect options both on and off: DONT_OPTIMIZE DEBUG_THREADS DEBUG_CHANNEL_LOCKS MALLOC_DEBUG MTX_PROFILE DEBUG_SCHEDULER DEBUG_THREADLOCALS DETECT_DEADLOCKS CHANNEL_TRACE I think from now on, I'm going to #undef all these features in the various utils native files; I guess I could do the same for the copied-in files, surrounded by STANDALONE ifdef. A standalone isn't going to care about threads, mutexes, etc. * pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-vtest13: fixed the regressions 2008-12-16 23:06 +0000 [r164978] Mark Michelson * /, channels/chan_sip.c: Merged revisions 164977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec 2008) | 7 lines After looking through SIP registration code most of the day, this is one of the few things I could find that was just plain wrong. Even though it probably isn't possible for it to happen, it seems weird to have code that checks if a pointer is NULL and then immediately dereferences that pointer if it was NULL. ........ 2008-12-16 22:57 +0000 [r164976] Tilghman Lesher * main/pbx.c, doc/api-1.6.2-changes.txt (added), funcs/func_logic.c, include/asterisk/pbx.h, utils/extconf.c, CHANGES, configs/extensions.conf.sample: Add timezone to the possible fields in a timespec. (closes issue #14028) Reported by: mostyn Patches: timezone-v2.patch uploaded by mostyn (license 398) (with additional code guideline fixes and a memory leak fix by me - license 14) 2008-12-16 22:45 +0000 [r164942] Jeff Peeler * apps/app_record.c: (closes issue #13669) Reported by: pj Delete file recording if recording terminated from a hangup. 2008-12-16 22:31 +0000 [r164941] Terry Wilson * channels/chan_sip.c: Make a note of the feature request in bug #11157 as per the reporter and oej, and suspend the bug since no one seems to be keen on implementing it any time soon. 2008-12-16 21:39 +0000 [r164821-164882] Russell Bryant * /, main/utils.c: Merged revisions 164881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008) | 9 lines Fix an issue where DEBUG_THREADS may erroneously report that a thread is exiting while holding a lock. If the last lock attempt was a trylock, and it failed, it will still be in the list of locks so that it can be reported. (closes issue #13219) Reported by: pj ........ * /, apps/app_macro.c: Merged revisions 164876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has been returned. This is a bug I noticed while looking at the code for app_macro. This return code means that another thread has assumed ownership of the channel and it can no longer be touched. (I hate this return code with a passion, by the way.) ........ * main/asterisk.c: Fix build issues on Linux after sysinfo related changes 2008-12-16 20:47 +0000 [r164809-164814] Joshua Colp * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Qualify trumps poke per lmadsen. * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add configuration options for finer control over how Asterisk handles having to poke all peers at seemingly the same time. (closes issue #13217) Reported by: cervajs 2008-12-16 20:41 +0000 [r164807] Russell Bryant * main/manager.c, /: Merged revisions 164806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) | 9 lines Add "restart gracefully" to the AMI blacklist of CLI commands. "module unload" was already identified as a command that can not be used from the AMI. "restart gracefully" effectively unloads all modules, and will run in to the same problems. (closes issue #13894) Reported by: kernelsensei ........ 2008-12-16 20:08 +0000 [r164802] Michiel van Baak * configure, include/asterisk/autoconfig.h.in, configure.ac, main/asterisk.c: introduce 'core show sysinfo' for systems that dont have the Linux-ish sysinfo stuff but do have sysctl. (closes issue #13433) Reported by: mvanbaak Patches: 2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license 7) with two free calls replaced with ast_free based on feedback on reviewboard Review: http://reviewboard.digium.com/r/91/ 2008-12-16 20:04 +0000 [r164801] Steve Murphy * main/pbx.c: (closes issue #14076) Reported by: toc Tested by: murf OK, Well this issue has had its share of flip-flopping. I found the following: 1. the code in question, in ext_cmp1 in pbx.c, would not allow two extensions that vary only by any dashes contained within them, to be defined in the same context. 2. for input dialstrings, dashes are NOT ignored. So, skipping them when sorting patterns seemed a bit silly. Thus, you might declare ext 891 in a context, but if you try dialing 8-9-1, it will NOT match 891. So, I proposed to remove the code from ext_cmp1 to skip the spaces and dashes. Just kept us from declaring 891 and 8-9-1 in the same context, forcing users to generate otherwise uselessly obfuscated dialplan code to get the same effect. Then, I tried out 1.4, and found that: 1. you can declare 891 and 8-9-1 in the same context! 2. You can't define 891, and have 8-9-1 match it! Nor can you define 8-9-1, and have 891 match it! So, it appears that my proposal simply restores the pbx to behaving as it did in 1.4. 2008-12-16 19:54 +0000 [r164798] Tilghman Lesher * contrib/scripts/safe_asterisk: Set up umask as a possible configuration option. (closes issue #13753) Reported by: irroot 2008-12-16 17:14 +0000 [r164737] Russell Bryant * /, main/threadstorage.c, include/asterisk/threadstorage.h: Merged revisions 164736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was being used within the context of the thread local data destructors. We would go off and allocate more thread local data while the pthread lib was in the middle of destroying it all. This led to a memory leak. Another issue was an invalid argument being provided to the the object_add API call. (closes issue #13678) Reported by: ys Tested by: Russell ........ 2008-12-16 16:50 +0000 [r164733] Joshua Colp * pbx/pbx_config.c: Be more detailed about why the include did not get included. (closes issue #14071) Reported by: kshumard Patches: pbx_config.patch.improvederroroutput.txt uploaded by kshumard (license 92) 2008-12-16 16:00 +0000 [r164675] Russell Bryant * /, channels/chan_sip.c: Merged revisions 164672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines Fix a memory leak related to the use of the "setvar" configuration option. The problem was that these variables were being appended to the list of vars on the sip_pvt every time a re-registration or re-subscription came in. Since it's just a waste of memory to put them there unless the request was an INVITE, then the fix is to check the request type before copying the vars. (closes issue #14037) Reported by: marvinek Tested by: russell ........ 2008-12-16 15:44 +0000 [r164659] Joshua Colp * channels/chan_sip.c: When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself. (closes issue #13634) Reported by: performer 2008-12-16 15:31 +0000 [r164648] Steve Murphy * main/pbx.c, /: Merged revisions 164634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5 lines I added a sentence to clarify why - and ' ' are ignored in patterns as per bug 14076. Leif says he'll put some stuff about it in the extensions.conf sample, etc. ........ 2008-12-16 15:00 +0000 [r164602-164623] Russell Bryant * apps/app_minivm.c: Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable that was not needed. (closes issue #14081) Reported by: pkempgen * /, res/res_musiconhold.c: Merged revisions 164605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16 Dec 2008) | 5 lines Don't try to change working directory if a directory was not configured. (closes issue #14089) Reported by: caspy ........ * channels/chan_dahdi.c: Fix usage of the DAHDI_VMWI ioctl. (closes issue #14090) Reported by: alecdavis Patches: chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license 585) 2008-12-16 01:52 +0000 [r164565] Sean Bright * doc/tex/odbcstorage.tex: Use tables instead of ASCII art. Also change a bit of minor formatting. 2008-12-15 22:25 +0000 [r164519-164525] Russell Bryant * channels/chan_iax2.c: Open a timer before loading configuration so that the trunking configuration option will take effect. (closes issue #14082) Reported by: seandarcy * channels/chan_iax2.c: Fix log message to refer to the generic timing interface, not DAHDI specifically (inspired by issue #14082) * main/frame.c: Make sure we handle a uint32_t payload in ast_frdup() (closes issue #14080) Reported by: fnordian Patches: frame.patch uploaded by fnordian (license 110) 2008-12-15 21:17 +0000 [r164485] Tilghman Lesher * configs/extconfig.conf.sample, pbx/pbx_realtime.c, CHANGES: Allow disabling pattern match searches within the Realtime dialplan switch. (closes issue #13698) Reported by: fhackenberger Patches: 20081211__bug13698.diff.txt uploaded by Corydon76 (license 14) Tested by: fhackenberger 2008-12-15 20:07 +0000 [r164419-164428] Mark Michelson * apps/app_page.c: Add an 'i' option to app_page. This option works the same as the 'i' options for app_dial and app_queue, in that they will ignore any attempts by phones to forward the call. (closes issue #13977) Reported by: putnopvut Patches: page_ignore_forwards.patch uploaded by putnopvut (license 60) Tested by: putnopvut, acunningham * /, include/asterisk/pbx.h: Merged revisions 164422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec 2008) | 3 lines Add the deadlock note to ast_spawn_extension as well ........ * /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged revisions 164416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) | 4 lines Add notes to autoservice and pbx doxygen regarding a potential deadlock scenario so that it is avoided in the future ........ 2008-12-15 19:48 +0000 [r164417] Tilghman Lesher * channels/chan_sip.c, include/asterisk/strings.h: Revert ast_str opacity in chan_sip for now, since something wasn't quite right in the merge. 2008-12-15 19:42 +0000 [r164415] Steve Murphy * include/asterisk/strings.h: I was getting this warning during a compile on a 64-bit machine running ubuntu server 8.10, and gcc-4.3.2: [CXXi] chan_vpb.ii -> chan_vpb.oo cc1plus: warnings being treated as errors In file included from /home/murf/asterisk/trunk/include/asterisk/utils.h:671, from chan_vpb.cc:46: /home/murf/asterisk/trunk/include/asterisk/strings.h: In function ‘char* ast_str_truncate(ast_str*, ssize_t)’: /home/murf/asterisk/trunk/include/asterisk/strings.h:479: error: comparison between signed and unsigned integer expressions make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2 which this fix silences 2008-12-15 18:12 +0000 [r164351] Joshua Colp * /, channels/chan_sip.c: Merged revisions 164350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 lines Do not try to unlock a non-existant channel if the transfer fails. (closes issue #13800) Reported by: dwagner Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608) ........ 2008-12-15 18:09 +0000 [r164349] Tilghman Lesher * cdr/cdr_pgsql.c: When querying for the structure of the CDR table, remove the schema, if it exists. (Closes issue #14058) 2008-12-15 17:24 +0000 [r164312] Joshua Colp * main/file.c: Use ast_seekstream to return the file stream back to the beginning instead of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module. (closes issue #14079) Reported by: elguero 2008-12-15 17:21 +0000 [r164272-164309] Russell Bryant * channels/h323/ast_h323.cxx, include/asterisk/strings.h: Fix a couple more build issues related to ast_str_opaque * pbx/pbx_dundi.c: When a reload is issued, always process the configuration for dundi.conf. The reason is that a reload can be used to refresh DNS lookups for defined peers. Even if the config file hasn't changed, we want to process it for that purpose. (closes issue #13776) Reported by: kombjuder 2008-12-15 16:16 +0000 [r164268-164270] Mark Michelson * apps/app_queue.c: Fix a compile warning and a logic error that could have been bad for non-realtime queues * apps/app_queue.c: Fix up a few issues with regards to queues * Fix reference counting used in the __queues_show function * Add code to be sure that the "queue show" command does not print information for a realtime queue which has been deleted from the backend * Add a missing unref to the realtime queue loading function for the case where a queue is in the module's container but has been deleted from the realtime backend (closes issue #14033) Reported by: cristiandimache Patches: 14033.patch uploaded by putnopvut (license 60) Tested by: cristiandimache 2008-12-15 15:41 +0000 [r164208-164257] Joshua Colp * configure, include/asterisk/autoconfig.h.in, apps/app_fax.c, configure.ac: Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret. (closes issue #14073) Reported by: seandarcy * main/utils.c: Update to work with new ast_str changes. 2008-12-15 14:40 +0000 [r164202-164203] Russell Bryant * main/channel.c, /, main/features.c: Merged revisions 164201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines Handle a case where a call can be bridged to a channel that is still ringing. The issue that was reported was about a case where a RINGING channel got redirected to an extension to pick up a call from parking. Once the parked call got taken out of parking, it heard silence until the other side answered. Ideally, the caller that was parked would get a ringing indication. This patch fixes this case so that the caller receives ringback once it comes out of parking until the other side answers. The fixes are: - Make sure we remember that a channel was an outgoing channel when doing a masquerade. This prevents an erroneous ast_answer() call on the channel, which causes a bogus 200 OK to be sent in the case of SIP. - Add some additional comments to explain related parts of code. - Update the handling of the ast_channel visible_indication field. Storing values that are not stateful is pointless. Control frames that are events or commands should be ignored. - When a bridge first starts, check to see if the peer channel needs to be given ringing indication because the calling side is still ringing. - Rework ast_indicate_data() a bit for the sake of readability. (closes issue #13747) Reported by: davidw Tested by: russell Review: http://reviewboard.digium.com/r/90/ ........ * apps/app_jack.c: Fix build WRT ast_str_opaque 2008-12-14 18:16 +0000 [r164168] Tilghman Lesher * include/asterisk/strings.h: Don't pass a negative to an unsigned type and expect things to work correctly. 2008-12-14 15:26 +0000 [r164054-164137] Sean Bright * doc/tex/cdrdriver.tex: Use a \picture instead of ASCII art. * res/snmp/agent.c: Use ast_str_strlen() instead of recalculating the string length. 2008-12-13 13:26 +0000 [r164028] Michiel van Baak * res/snmp/agent.c: nuke another use of the ast_str internals. 2008-12-13 08:36 +0000 [r163991] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, funcs/func_strings.c, utils/hashtest.c, cdr/cdr_adaptive_odbc.c, main/utils.c, apps/app_chanisavail.c, include/asterisk/tcptls.h, cdr/cdr_pgsql.c, res/res_http_post.c, apps/app_followme.c, res/res_config_sqlite.c, main/config.c, main/cli.c, main/cdr.c, channels/chan_dahdi.c, res/res_config_odbc.c, main/manager.c, configure, funcs/func_odbc.c, res/res_agi.c, apps/app_dumpchan.c, main/logger.c, main/http.c, main/app.c, apps/app_externalivr.c, res/res_config_ldap.c, include/asterisk/threadstorage.h, cdr/cdr_manager.c, res/res_clialiases.c, utils/refcounter.c, res/res_config_pgsql.c, main/strings.c (added), main/pbx.c, channels/chan_sip.c, main/Makefile, main/translate.c, include/asterisk/cdr.h, apps/app_queue.c, channels/iax2-parser.c, funcs/func_realtime.c, utils/Makefile, res/res_config_curl.c, main/tcptls.c, include/asterisk/app.h, funcs/func_curl.c, utils/hashtest2.c, include/asterisk/strings.h, include/asterisk/pbx.h, main/asterisk.c, main/xmldoc.c, apps/app_voicemail.c, utils/check_expr.c: Merge ast_str_opaque branch (discontinue usage of ast_str internals) 2008-12-13 03:03 +0000 [r163951-163952] Sean Bright * doc/tex/asterisk.tex: This shouldn't have gotten commited. We might want to generate this into a separate file instead of the version controlled one. * doc/tex/qos.tex, doc/tex/asterisk.tex: Use actual tables instead of ASCII art ones. 2008-12-13 00:59 +0000 [r163912] Joshua Colp * apps/app_chanspy.c: Only detach and destroy the whisper audiohooks if they are actually in use. 2008-12-12 23:48 +0000 [r163873] Terry Wilson * apps/app_queue.c: When using realtime queues, app_queue wasn't updating the strategy if it was changed in the realtime backend. This patch resolves the issue for almost all situations. It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered. (closes issue #14034) Reported by: cristiandimache Tested by: otherwiseguy, cristiandimache 2008-12-12 23:06 +0000 [r163828] Russell Bryant * res/res_clioriginate.c: Add a note to indicate why this only supports one channel for now. 2008-12-12 22:04 +0000 [r163762] Tilghman Lesher * main/editline/read.c, /, main/asterisk.c: Merged revisions 163761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a pointer inside editline to look back to asterisk.c, so others don't spend as much time as I did looking (in the wrong place) for the appropriate function. Reported by: ZX81, via the #asterisk-users channel Fixed by: me (license 14) ........ 2008-12-12 20:12 +0000 [r163716] Russell Bryant * CHANGES, res/res_clioriginate.c: Add a new CLI command, "channel redirect", which is similar in operation to AMI Redirect. Review: http://reviewboard.digium.com/r/89/ 2008-12-12 19:16 +0000 [r163675] Steve Murphy * channels/chan_dahdi.c: demote always-appearing debug message (for certain boards) to ast_debug lev 3 msg instead 2008-12-12 18:45 +0000 [r163642-163670] Russell Bryant * main/tcptls.c, channels/chan_sip.c: Rename a number of tcptls_session variables. There are no functional changes here. The name "ser" was used in a lot of places. However, it is a relic from when the struct was a server_instance, not a session_instance. It was renamed since it represents both a server or client connection. * channels/chan_sip.c: Fix a small race condition in sip_tcp_locate(). We must increase the reference count on the tcptls_session _before_ unlocking the thread list. * channels/chan_sip.c: Resolve crashes when using SIP TCP/TLS with qualify. The problem was a reference count error on the tcptls_session structure. (closes issue #13989) Reported by: Nugget 2008-12-12 18:17 +0000 [r163629] Joshua Colp * channels/chan_sip.c: When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found. (closes issue #13811) Reported by: pj 2008-12-12 17:22 +0000 [r163582-163612] Michiel van Baak * res/res_monitor.c: Document default Monitor file location. (closes issue #14065) Reported by: kshumard Patches: res_monitor.documentation.patch.txt uploaded by kshumard (license 92) * channels/chan_skinny.c: Fix codec capability setup in chan_skinny Behaviour now is that general codec config flows to default_line and default_device. [devices] stuff amends default_device and similar for [lines]. These are copied to individual device and line as they are created. Added confcapability and confprefs for the configured stuff which doesn't change as device and so on are connected. prefs are based on line prefs if they exist, else the device prefs are used (prefs identifies codec order). (closes issue #13806) Reported by: pj Patches: codecs.diff uploaded by wedhorn (license 30) Tested by: pj and me 2008-12-12 16:55 +0000 [r163579] Joshua Colp * main/channel.c, channels/chan_sip.c: Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven. (closes issue #13525) Reported by: pj 2008-12-12 15:10 +0000 [r163516] Doug Bailey * configs/phoneprov.conf.sample: Add internationalization to sample configuration file 2008-12-12 14:44 +0000 [r163449-163512] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 163511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008) | 5 lines Specify uint32_t for variables storing a CRC32 so that it is actually 32 bits on 64-bit machines, as well. (inspired by issue #13879) ........ * main/channel.c, main/autoservice.c, /, include/asterisk/channel.h: Merged revisions 163448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ 2008-12-11 23:38 +0000 [r163384] Tilghman Lesher * /, main/asterisk.c: Merged revisions 163383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) | 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal is messed up. By intercepting those events with a signal handler in the remote console, we can avoid those issues. (closes issue #13464) Reported by: tzafrir Patches: 20081110__bug13464.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ 2008-12-11 22:49 +0000 [r163317] Matthew Nicholson * /, pbx/pbx_dundi.c: Merged revisions 163316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec 2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes issue #13819) Reported by: adomjan Patches: pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487) dundi_clearecache3.diff uploaded by mnicholson (license 96) Tested by: adomjan ........ 2008-12-11 21:48 +0000 [r163241-163254] Russell Bryant * /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions 163253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008) | 8 lines Fix some observed slowdowns in dialplan processing. The change is to remove autoservice usage from dialplan functions that do not need it because they do not perform operations that potentially block. (closes issue #13940) Reported by: tbelder ........ * res/res_timing_pthread.c: Fix a problem where continuous mode will get inadvertently get turned off if set_rate() is used while continuous mode was already turned on. (closes issue #13738) Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix (license 547) 2008-12-11 20:57 +0000 [r163198-163213] Mark Michelson * configs/voicemail.conf.sample, apps/app_voicemail.c: Add an option to voicemail.conf to allow urgent messages to be forwarded as not urgent. (closes issue #14063) Reported by: jaroth Patches: urgfwd_v2.patch uploaded by jaroth (license 50) * main/features.c: Add an appropriate goto if ast_call fails 2008-12-11 20:07 +0000 [r163171] Russell Bryant * main/channel.c: Fix the "failed" extension for outgoing calls. The conversion to use ast_check_hangup() everywhere instead of checking the softhangup flag directly introduced this problem. The issue is that ast_check_hangup() checked for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances, such as with a dummy channel. The fix is simple. Don't check tech_pvt. It's pointless, because the code path that sets this to NULL is when the channel hangup callback gets called. This happens inside of ast_hangup(), which is the same function responsible for freeing the channel. Any code calling ast_check_hangup() better not be calling it after that point, and if so, we have a bigger problem at hand. (closes issue #14035) Reported by: erogoza 2008-12-11 20:02 +0000 [r163168] Tilghman Lesher * configure, configure.ac: Sometimes even Linux needs -lm to link libtonezone, such as when libtonezone is compiled statically. (closes issue #13887) Reported by: tzafrir 2008-12-11 19:40 +0000 [r163166] Mark Michelson * main/features.c: Reduce indentation level of ast_feature_request_and_dial 2008-12-11 17:06 +0000 [r163094] Russell Bryant * /, main/features.c: Merged revisions 163092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) | 11 lines Fix an issue that made it so you could only have a single caller executing a custom feature at a time. This was especially problematic when custom features ran for any appreciable amount of time. The fix turned out to be quite simple. The dynamic features are now stored in a read/write list instead of a list using a mutex. (closes issue #13478) Reported by: neutrino88 Fix suggested by file ........ 2008-12-11 16:52 +0000 [r163089] Tilghman Lesher * /, res/res_agi.c: Merged revisions 163088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008) | 6 lines Don't wait forever, if there's a specified recording timeout. (closes issue #13885) Reported by: bamby Patches: res_agi.c.patch uploaded by bamby (license 430) ........ 2008-12-11 16:47 +0000 [r163081-163085] Mark Michelson * /, apps/app_queue.c: Merged revisions 163084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec 2008) | 4 lines Revert this cast to long. Using time_t here causes build failures on a FreeBSD 32-bit build. ........ * /, apps/app_queue.c: Merged revisions 163080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec 2008) | 14 lines Fix a potential crash due to unsafe datastore handling. This patch also contains a conversion from using long to time_t for representing times for a queue, as well as some whitespace fixes. (closes issue #14060) Reported by: nivek Patches: datastore_fixup.patch.corrected uploaded by nivek (license 636) with slight modification from me Tested by: nivek ........ 2008-12-11 15:40 +0000 [r163037] Sean Bright * doc/tex/qos.tex: Fix some of the grammar issues in doc/tex/qos.tex. (closes issue #14049) Reported by: kshumard Patches: doc.tex.qos.tex.patch uploaded by kshumard (license 92) (Slight modifications by seanbright) 2008-12-11 15:05 +0000 [r162997] Joshua Colp * channels/chan_sip.c: When a device registers to use it is entirely possible that they may be in use, so tell the core that we don't know the devstate and have it ask us for it. (closes issue #13525) Reported by: pj 2008-12-10 23:01 +0000 [r162930] Tilghman Lesher * main/pbx.c: Previously missing line, now the substitution works correctly 2008-12-10 22:53 +0000 [r162927] Jeff Peeler * /, res/res_musiconhold.c: Merged revisions 162926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10 Dec 2008) | 3 lines Oops, inverted logic for a strcasecmp check. Pointed out by mmichelson, thanks! ........ 2008-12-10 22:48 +0000 [r162923] Joshua Colp * res/res_clialiases.c: Fix reloads of aliased CLI commands. Due to changes done to turn it into a single memory allocation we can't just use the existing CLI alias structure. We have to destroy all existing ones and then create new ones. (closes issue #14054) Reported by: pj 2008-12-10 22:48 +0000 [r162922] Tilghman Lesher * main/pbx.c: Checking global variables here actually overwrote the previous substitution by channel variables, and in any case, was redundant; pbx_substitute_variables_helper ALREADY does substitution for global variables. (closes issue #13327) Reported by: pj 2008-12-10 22:11 +0000 [r162891] Jeff Peeler * /, res/res_musiconhold.c: Merged revisions 162874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 Dec 2008) | 5 lines (closes issue #13229) Reported by: clegall_proformatique Ensure that moh_generate does not return prematurely before local_ast_moh_stop is called. Also, the sleep in mp3_spawn now only occurs for http locations since it seems to have been added originally only for failing media streams. ........ 2008-12-10 19:02 +0000 [r162739-162805] Joshua Colp * /, channels/chan_sip.c: Merged revisions 162804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI. (closes issue #12560) Reported by: vsauer Patches: patch001.diff uploaded by ramonpeek (license 266) ........ * /, channels/chan_sip.c: Merged revisions 162738 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0. (closes issue #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded by hjourdain (license 583) ........ 2008-12-10 17:09 +0000 [r162687] Michiel van Baak * include/asterisk.h, main/asterisk.c, main/cli.c: add tab completion for 'core set debug X filename.c' (closes issue #13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel 2008-12-10 16:39 +0000 [r162664-162667] Mark Michelson * doc/tex/misdn.tex, /: Merged revisions 162659 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec 2008) | 8 lines Add missing documentation to misdn.txt (closes issue #14052) Reported by: festr Patches: misdn.txt.patch uploaded by festr (license 443) ........ * /, channels/chan_sip.c: Merged revisions 162663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec 2008) | 11 lines Revert fix for issue 13570. It has caused more problems than it helped to fix. (closes issue #13783) Reported by: navkumar (closes issue #14025) Reported by: ffs ........ 2008-12-10 16:11 +0000 [r162619-162660] Joshua Colp * res/res_http_post.c: FreeBSD also needs libgen.h (closes issue #14051) Reported by: ys Patches: res_http_post.c.diff uploaded by ys (license 281) * /, main/rtp.c: Merged revisions 162653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change. (closes issue #12983) Reported by: vt Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520) ........ * channels/chan_sip.c: When transmitting a register set the socket port to the local one for the transport being used, not the port for the remote server. (closes issue #13633) Reported by: performer 2008-12-10 11:34 +0000 [r162583] Michiel van Baak * res/snmp/agent.c: Make res_snmp.so compile on OpenBSD. OpenBSD uses an old version of gcc which throws an error if you use a macro that's not #defined 2008-12-10 01:09 +0000 [r162542] Joshua Colp * doc/janitor-projects.txt, channels/iax2-parser.c, apps/app_voicemail.c: Finish conversion to using ARRAY_LEN and remove it as a janitor project. (closes issue #14032) Reported by: bkruse Patches: 14032.patch uploaded by bkruse (license 132) 2008-12-09 23:41 +0000 [r162488] Kevin P. Fleming * include/asterisk/stringfields.h: it does help if the compiler attribute syntax is correct 2008-12-09 23:10 +0000 [r162466] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 162463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........ 2008-12-09 22:38 +0000 [r162414-162418] Russell Bryant * include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen, main/asterisk.c: Add some additional Asterisk project developer documentation. After the nightly update of the documentation on asterisk.org, I'll post an update to asterisk-dev with a pointer to the changes. This covers some release branch and commit policy information. None of this should be a surprise, since it's just documenting what we have already been doing. * include/asterisk/utils.h, /, main/utils.c, main/asterisk.c: Merged revisions 162413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines Remove the test_for_thread_safety() function completely. The test is not valid. Besides, if we actually suspected that recursive mutexes were not working, we would get a ton of LOG_ERROR messages when DEBUG_THREADS is turned on. (inspired by a discussion on the asterisk-dev list) ........ 2008-12-09 21:57 +0000 [r162355] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 162348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) | 4 lines We appear to have documented tz= in the [general] section of voicemail.conf, without actually having implemented it. Oops. (Reported by Olivier on the -users list) ........ 2008-12-09 21:16 +0000 [r162342] Joshua Colp * /, apps/app_directed_pickup.c: Merged revisions 162341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 lines Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing. (closes issue #14005) Reported by: ddl ........ 2008-12-09 20:59 +0000 [r162291] Russell Bryant * /, apps/app_meetme.c: Merged revisions 162286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback. We need to make sure that we don't start writing audio to the trunk channel until we're actually ready to answer it. Otherwise, the channel driver will treat it as inband progress, even though all they are getting is silence. (closes issue #12471) Reported by: mthomasslo ........ 2008-12-09 20:46 +0000 [r162275] Joshua Colp * /, apps/app_festival.c: Merged revisions 162273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 lines Fix double declaration of 'x' on the PPC platform. (closes issue #14038) Reported by: ffloimair ........ 2008-12-09 20:40 +0000 [r162271] Steve Murphy * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 line In discussion with seanbright on #asterisk-dev, I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this. ........ 2008-12-09 20:30 +0000 [r162266] Mark Michelson * main/pbx.c, /: Merged revisions 162265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) | 6 lines If we fail to start a thread for the pbx to run in, we need to be sure to decrease the number of active calls on the system. This fix may relate to ABE-1713, but it is not certain yet. ........ 2008-12-09 19:48 +0000 [r162197-162205] Joshua Colp * /, main/rtp.c: Merged revisions 162204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment. (closes issue #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file (license 11) Tested by: ip-rob, bujones ........ * /, main/rtp.c: Merged revisions 162188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines Take video into account when early bridging RTP. (closes issue #13535) Reported by: davidw ........ 2008-12-09 18:35 +0000 [r162079-162140] Steve Murphy * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1 line Previous fix used ast_malloc and ast_copy_string and messed up the standalone stuff. Fixed. ........ * res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches: 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme, murf This crash was the result of a few small errors that would combine in 64-bit land to result in a crash. 32-bit land might have seen these combine to mysteriously drop the args to an application call, in certain circumstances. Also, in trying to find this bug, I spotted a situation in the flex input, where, in passing back a 'word' to the parser, it would allocate a buffer larger than necessary. I changed the usage in such situations, so that strdup was not used, but rather, an ast_malloc, followed by ast_copy_string. I removed a field from the pval struct, in u2, that was never getting used, and set in one spot in the code. I believe it was an artifact of a previous fix to make switch cases work invisibly with extens. And, for goto's I removed a '!' from before a strcmp, that has been there since the initial merging of AEL2, that might prevent the proper target of a goto from being found. This was pretty harmless on its own, as it would just louse up a consistency check for users. Many thanks to ckjohnsonme for providing a simplified and complete set of information about the bug, that helped considerably in finding and fixing the problem. Now, to get aelparse up and running again in trunk, and out of its "horribly broken" state, so I can run the regression suite! ........ 2008-12-09 16:47 +0000 [r161951-162016] Russell Bryant * /, apps/app_disa.c: Merged revisions 162014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) | 5 lines Allow DISA to handle extensions that start with #. (closes issue #13330) Reported by: jcovert ........ * /, main/app.c: Merged revisions 161948 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) | 15 lines Fix a problem with GROUP() settings on a masquerade. The previous code carried over group settings from the old channel to the new one. However, it did nothing with the group settings that were already on the new channel. This patch removes all group settings that already existed on the new channel. I have a more complicated version of this patch which addresses only the most blatant problem with this, which is that a channel can end up with multiple group settings in the same category. However, I could not think of a use case for keeping any of the group settings from the old channel, so I went this route for now. (closes AST-152) ........ 2008-12-09 14:49 +0000 [r161947] Eliel C. Sardanons * funcs/func_odbc.c: Avoid allocating memory for a thread that don't need it. Also, this memory was not being freed until the main thread ends. (That is never). (closes issue #14040) Reported by: eliel Patches: func_odbc.c.patch uploaded by eliel (license 64) 2008-12-08 23:04 +0000 [r161911] Brandon Kruse * main/pbx.c: Note that the recently changed waittime parameter is in milliseconds. 2008-12-08 21:41 +0000 [r161830-161869] Joshua Colp * formats/format_pcm.c: Add alw as a valid file extension for alaw and ulw as a valid file extension for ulaw. (closes issue #14001) Reported by: henrikw Patches: alw.diff uploaded by henrikw (license 627) * contrib/scripts/autosupport.8, contrib/scripts/autosupport: Update autosupport script with a few changes. 2008-12-08 18:49 +0000 [r161790] Tilghman Lesher * main/manager.c: Allocate enough space initially for the message. (closes issue #14027) Reported by: junky Patches: M14027.diff uploaded by junky (license 177) 2008-12-08 18:47 +0000 [r161726-161787] Joshua Colp * main/pbx.c: Fix a regression introduced when the PBX timeouts were converted to milliseconds. collect_digits now gets milliseconds fed to it, not seconds. (closes issue #14012) Reported by: dveiga Patches: 14012.patch uploaded by bkruse (license 132) * /, channels/chan_sip.c: Merged revisions 161725 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 lines Make the usereqphone option work again. (closes issue #13474) Reported by: mmaguire Patches: 20080912_bug13474.diff uploaded by mmaguire (license 571) ........ 2008-12-08 17:23 +0000 [r161721] Matthew Nicholson * channels/chan_sip.c: Fix a crash that can occur on a transfer in chan_sip when attempting to collect rtp stats. (closes issue #13956) Reported by: chris-mac Tested by: chris-mac 2008-12-08 16:02 +0000 [r161679] Terry Wilson * channels/chan_sip.c, CHANGES: Add the ability to play a courtesy tone to the transfer target in a native SIP attended transfer by setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND. 2008-12-08 04:23 +0000 [r161571-161637] Eliel C. Sardanons * main/xmldoc.c: - Fix a leak while printing an argument description. - Avoid printing the name of an argument in the [Arguments] tag if there is no description for that argument. * apps/app_voicemail.c: Add voicemail related applications and functions XML documentation: applications: - VoiceMail() - VoiceMailMain() - MailboxExists() - VMAuthenticate() functions: - MAILBOX_EXISTS() * apps/app_sms.c: Introduce SMS() application XML documentation. 2008-12-06 21:18 +0000 [r161536] Eliel C. Sardanons * apps/app_speech_utils.c: Move Speech* applications and functions documentation to XML. 2008-12-05 23:24 +0000 [r161493] Mark Michelson * apps/app_stack.c: If the autoloop flag is set on a channel, then we need to add 1 to the priority when checking if the extension exists. Otherwise, gosubs will fail. This was discovered when investigating an asterisk-users mailing list post made by Gary Hawkins. 2008-12-05 21:08 +0000 [r161349-161427] Sean Bright * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions 161426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r161426 | seanbright | 2008-12-05 16:02:20 -0500 (Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned int). (closes issue #14006) Reported by: alphaque Patches: astobj2.h-patch uploaded by alphaque (license 259) (Slightly modified by seanbright) ........ ................ * apps/app_voicemail.c: Use ast_free() instead of free(), pointed out by eliel on IRC. * apps/app_voicemail.c: When using IMAP_STORAGE, it's important to convert bare newlines (\n) in emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed by Mark M. on IRC. 2008-12-05 14:16 +0000 [r161252-161288] Russell Bryant * main/pbx.c, /: Merged revisions 161287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008) | 2 lines Fix a NULL format string warning found by buildbot. ........ * apps/app_minivm.c: Resolve a compiler warning from buildbot about a NULL format string. 2008-12-05 10:31 +0000 [r161218] Eliel C. Sardanons * main/udptl.c, main/frame.c, res/res_musiconhold.c, channels/chan_iax2.c, res/res_jabber.c, res/res_config_sqlite.c, main/config.c, main/cli.c, channels/chan_dahdi.c, main/manager.c, channels/chan_skinny.c, res/res_agi.c, main/features.c, apps/app_minivm.c, pbx/pbx_ael.c, main/logger.c, main/http.c, res/res_realtime.c, channels/chan_alsa.c, res/res_config_ldap.c, apps/app_rpt.c, main/db.c, res/res_config_pgsql.c, main/pbx.c, channels/chan_sip.c, main/translate.c, channels/chan_agent.c, res/res_convert.c, res/res_crypto.c, apps/app_queue.c, channels/chan_oss.c, apps/app_playback.c, channels/chan_usbradio.c, main/file.c, main/astmm.c, pbx/pbx_dundi.c, res/res_indications.c, pbx/pbx_config.c, apps/app_mixmonitor.c, res/res_odbc.c, main/asterisk.c, apps/app_voicemail.c: Janitor, use ARRAY_LEN() when possible. (closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) 2008-12-05 05:41 +0000 [r161181] Tilghman Lesher * main/config.c: The first file should have a blank config filename in the structure, so that when a save occurs to a different filename, everything goes to the alternate filename, instead of appending to the original. This is important for the AMI command UpdateConfig. (closes issue #13301) Reported by: trevo Patches: 20081113__bug13301.diff.txt uploaded by Corydon76 (license 14) 20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage 2008-12-05 02:47 +0000 [r161147] Sean Bright * apps/app_voicemail.c: Check the return value of fread/fwrite so the compiler doesn't complain. Only a problem when IMAP_STORAGE is enabled. 2008-12-04 23:00 +0000 [r161115] Dwayne M. Hubbard * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) after T38 is negotiated. Terry Wilson created the original patch for this functionality, which I slightly modified and added the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not do anything for G711 over SIP fax detection. By default, this option is disabled. Reviewboard: http://reviewboard.digium.com/r/69/ This functionality is for issue AST-140. 2008-12-04 19:31 +0000 [r161077] Eliel C. Sardanons * main/cli.c: Fix minor coding guidelines introduced with CLI permissions. 2008-12-04 18:32 +0000 [r161014] Jeff Peeler * /, main/rtp.c: Merged revisions 161013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines (closes issue #13835) Reported by: matt_b Tested by: jpeeler This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure. Closes AST-142. ........ 2008-12-04 16:45 +0000 [r160945] Mark Michelson * /, main/callerid.c: Merged revisions 160943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec 2008) | 15 lines Fix a callerid parsing issue. If someone formatted callerid like the following: "name " (including the quotation marks), then the parts would be parsed as name: "name number: number This is because the closing quotation mark was not discovered since the number and everything after was parsed out of the string earlier. Now, there is a check to see if the closing quote occurs after the number, so that we can know if we should strip off the opening quote on the name. Closes AST-158 ........ 2008-12-04 16:37 +0000 [r160938] Michiel van Baak * build_tools/cflags-devmode.xml, channels/chan_skinny.c: Add debug flag so skinny debug will show information about packets. We dont want to scare users with this, so we added a devmode compile flag (closes issue #13952) Reported by: wedhorn Patches: packetdebug3.diff uploaded by wedhorn (license 30) Tested by: mvanbaak, wedhorn 2008-12-04 13:45 +0000 [r160896] Eliel C. Sardanons * res/res_agi.c: Added XML documentation for the following AGI commands: - get option - get variable - hangup - noop 2008-12-04 01:36 +0000 [r160854-160856] Richard Mudgett * funcs/func_callerid.c: Jcolp pointed out that num will also match number * funcs/func_callerid.c: * Found a couple more places where num/number needed to be done so 1.4 upgraders will not have problems. * Added curly braces and minor tweaks. 2008-12-03 21:58 +0000 [r160791] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 160770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 Dec 2008) | 2 lines Some compilers warn on null format strings; some don't (caught by buildbot) ........ 2008-12-03 21:09 +0000 [r160760] Steve Murphy * /, funcs/func_callerid.c: Merged revisions 160703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | 11 lines (closes issue #13597) Reported by: john8675309 Patches: patch.13597 uploaded by murf (license 17) Tested by: murf, john8675309 This patch causes the setcid func to update the CDR clid after setting the channel field. I also notice that in trunk, the num/number of 1.4 is left out; I decided to include the option to use either in trunk, so as not to have 1.4 upgraders not to have problems. ........ 2008-12-03 20:35 +0000 [r160699-160700] Jason Parker * main/manager.c: Another place this is missing * main/manager.c: Fix typo when ListCategories returns none. (closes issue #13994) Reported by: mika Patches: ListCategoriesActionPatch.diff uploaded by mika (license 624) 2008-12-03 19:25 +0000 [r160663] Eliel C. Sardanons * channels/iax2-provision.c: - iax2-provision was not freeing iax_templates structure when unloading the chan_iax2.so module. - Move the code to start using the LIST macros. Review: http://reviewboard.digium.com/r/72 (closes issue #13232) Reported by: eliel Patches: iax2-provision.patch.txt uploaded by eliel (license 64) (with minor changes pointed by Mark Michelson on review board) Tested by: eliel 2008-12-03 18:37 +0000 [r160626] Mark Michelson * apps/app_dial.c, apps/app_queue.c, apps/app_stack.c: Add some safety measures when using gosub, especially when using the options for app_dial and app_queue to run a gosub when the call is answered. * Check for the existence of the gosub target in gosub_exec. If it is nonexistent, then this will cause errors when we attempt to actually run the gosub, including a definite memory leak and potential crashes. Return an error in this situation * Check the return value of pbx_exec in app_dial and app_queue before attempting to actually run the gosub routine. If there was an error, we should not attempt to run the gosub. * Change a '|' to a ',' in app_queue. * Add some extra curly braces where they had been missing previously. (closes issue #13548) Reported by: fiddur 2008-12-03 17:48 +0000 [r160562] Eliel C. Sardanons * apps/app_minivm.c: - Add tags when naming a channel variable. - Add tags when naming a filename. - Simplify the xml formatting putting some enters. 2008-12-03 17:38 +0000 [r160559] Tilghman Lesher * pbx/pbx_spool.c, /: Merged revisions 160558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008) | 7 lines If an entry is added to the directory during a scan when another entry expires, then that new entry will not be processed promptly, but must wait for either a future entry to start or a current entry's retry to occur. If no other entries exist in the directory (other than the new entries) when a bunch expire, then the new entries must wait until another new entry is added to be processed. This was a rather weird race condition, really. Fixes AST-147. ........ 2008-12-03 17:07 +0000 [r160555] Mark Michelson * apps/app_queue.c: When investigating issue #13548, I found that gosub handling in app_queue was just completely wrong, mostly because the channel operations being performed were being done on the incorrect channel. With this set of changes, a gosub will correctly run on the answering queue member's channel. There are still crash issues which occur if there are dialplan syntax errors, so I cannot yet close the referenced issue. 2008-12-03 17:01 +0000 [r160481-160552] Tilghman Lesher * pbx/pbx_spool.c, /: Merged revisions 160551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008) | 4 lines Don't start scanning the directory until all modules are loaded, because some required modules (channels, apps, functions) may not yet be in memory yet. Fixes AST-149. ........ * /, channels/chan_sip.c: Merged revisions 160480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I guess that having only ip-phones in mind is not a good approach. Since it is possible to have a sip proxy connected to asterisk we could receive a 407 (unauthorized) or 483 (too many hops) as response and dialog ending would not be a good behavior." So modified. ........ 2008-12-03 11:01 +0000 [r160447] Eliel C. Sardanons * apps/app_stack.c: - Avoid setting .synopsis and .syntax if we are using XML documentation (or the xml documentation wont be loaded). - Use to refer to a dialplan variable. 2008-12-02 18:48 +0000 [r160344-160346] Tilghman Lesher * CHANGES: Info on LOCAL_PEEK function. * apps/app_stack.c: Add LOCAL_PEEK function, as requested by lmadsen. 2008-12-02 18:04 +0000 [r160319-160333] Jeff Peeler * channels/chan_dahdi.c: remove duplicate comment that I accidentally merged * channels/chan_dahdi.c: (closes issue #13786) Reported by: tzafrir Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which fixes not being able to make outgoing calls on some FXO adapters: http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553 2008-12-02 17:56 +0000 [r160208-160308] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 160297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion fails, and the resulting integer is garbage. Thus, we must initialize the integer and check it afterwards for success. (closes issue #14000) Reported by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626) ........ * main/pbx.c, main/frame.c, /, channels/chan_features.c, include/asterisk/stringfields.h, apps/app_voicemail.c, main/cli.c: Merged revisions 160207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ 2008-12-01 23:37 +0000 [r160170-160172] Sean Bright * main/manager.c, /: Merged revisions 159976 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) | 3 lines Get rid of the useless format string and argument in the Bogus/ manager channelname. Noted by kpfleming and name Bogus/manager suggested by eliel ........ * channels/chan_phone.c: Silence a build warning. (chan_phone.c:810: warning: value computed is not used) * utils/smsq.c: Pay attention to the return value of system(), even if we basically ignore it. 2008-12-01 21:23 +0000 [r160097] Tilghman Lesher * configure, configure.ac: Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or bad things happen. 2008-12-01 18:52 +0000 [r160062] Eliel C. Sardanons * configs/cli_permissions.conf.sample (added), configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/cli.h, include/asterisk/_private.h, CHANGES, main/asterisk.c, main/cli.c: Introduce CLI permissions. Based on cli_permissions.conf configuration file, we are able to permit or deny cli commands based on some patterns and the local user and group running rasterisk. (Sorry if I missed some of the testers). Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue #11123) Reported by: eliel Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak 2008-12-01 17:34 +0000 [r159911-160004] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 160003 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they both have the potential to send control frames in the middle of call setup. We have to wait until we have received a message back from the remote end before we try to send any more frames. Otherwise, the remote end will consider it invalid, and we'll get stuck in an INVAL/VNAK storm. ........ * /, .cleancount: Merged revisions 159900 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008) | 2 lines Force a "make clean" to avoid a bizarre build issue ... ........ 2008-12-01 14:09 +0000 [r159898] Michiel van Baak * main/manager.c, /: Merged revisions 159897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008) | 4 lines make manager compile on OpenBSD. The last (10th) argument to ast_channel_alloc here should be a pointer and NULL is not really a pointer. ........ 2008-11-29 18:33 +0000 [r159853] Tilghman Lesher * apps/app_readexten.c: Allow the '#' sign to exist within an extension (inspired by issue #13330) 2008-11-29 17:57 +0000 [r159774-159818] Kevin P. Fleming * channels/chan_vpb.cc, /, main/utils.c, channels/chan_iax2.c, utils/frame.c, include/asterisk/astmm.h, configure, include/asterisk/compat.h, main/features.c, include/asterisk/module.h, main/logger.c, include/asterisk/dlinkedlists.h, main/dns.c, include/asterisk/utils.h, include/asterisk/devicestate.h, channels/chan_sip.c, include/asterisk/dundi.h, include/asterisk/enum.h, configure.ac, channels/chan_agent.c, include/asterisk/config.h, utils/astman.c, include/asterisk/cli.h, include/asterisk/channel.h, include/jitterbuf.h, include/asterisk/manager.h, utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c, include/asterisk/logger.h, Makefile, include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c, include/asterisk/linkedlists.h, main/event.c, include/asterisk/lock.h, include/asterisk/strings.h, utils/extconf.c, makeopts.in, include/asterisk/stringfields.h, main/xmldoc.c, utils/check_expr.c: incorporates r159808 from branches/1.4: ------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings * Makefile, funcs/func_sprintf.c (added), main/Makefile, channels/misdn/ie.c, funcs/func_strings.c, UPGRADE.txt, res/res_config_sqlite.c, channels/misdn_config.c, funcs/Makefile: we can now build with -Wformat=2, which found a couple of real bugs because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking 2008-11-28 14:20 +0000 [r159734] Michiel van Baak * res/Makefile: Make res_config_ldap compile with the official OpenLDAP 2.3.X versions. They removed the LDAP_DEPRECATED define from their source and since we are using a couple of deprecated function calls we should define it with a CFLAG. Tested by me on OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps compiling. It shouldn't break, we only define the LDAP_DEPRECATED with this which is what all 2.2.X and older versions of OpenLDAP did in their own tree. 2008-11-27 20:29 +0000 [r159701] Philippe Sultan * res/res_jabber.c: Removed duplicate code 2008-11-26 22:11 +0000 [r159664-159666] Russell Bryant * main/pbx.c: Make a formatting change to test a new post-commit hook for reviewboard. http://reviewboard.digium.com/r/65/ * main/pbx.c: Make a formatting change to test a new post-commit hook for reviewboard. http://reviewboard.digium.com/r/65/ * main/pbx.c: Make a formatting change to test a new post-commit hook for reviewboard. http://reviewboard.digium.com/r/65/ 2008-11-26 21:20 +0000 [r159629-159631] Kevin P. Fleming * include/asterisk/agi.h, configure, include/asterisk/autoconfig.h.in, contrib/asterisk-ng-doxygen, autoconf/ast_gcc_attribute.m4, configure.ac, res/res_agi.c, apps/app_stack.c, include/asterisk/optional_api.h (added): improve handling of API calls provided by loaded modules through use of some GCC features; this makes app_stack's usage of AGI APIs even cleaner, and will allow it to work 'as expected' either with or without res_agi being loaded reviewed at http://reviewboard.digium.com/r/62 * main/manager.c, CHANGES: add support for event suppression for AMI-over-HTTP 2008-11-26 19:57 +0000 [r159554] Mark Michelson * apps/app_dial.c: Add some necessary hangup commands in the case that forwarding a call fails 1) Hang up the original destination if the local channel cannot be requested. 2) Hang up the local channel (in addition to the original destination) if ast_call fails when calling the newly created local channel. This prevents channels from sticking around forever in the case of a botched call forward (e.g. to an extension which does not exist). (closes issue #13764) Reported by: davidw Patches: 13764_v2.patch uploaded by putnopvut (license 60) Tested by: putnopvut, davidw 2008-11-26 19:08 +0000 [r159534] Kevin P. Fleming * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged revisions 159476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov 2008) | 7 lines simplify (and slightly bug-fix) the recent developer-oriented COMPILE_DOUBLE mode ensure that 'make clean' removes dependency files for .i files that are created in COMPILE_DOUBLE mode ........ 2008-11-26 18:33 +0000 [r159475] Tilghman Lesher * main/udptl.c: If the config file does not exist, then the first use crashes Asterisk. (closes issue #13848) Reported by: klaus3000 Patches: udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage 2008-11-26 14:58 +0000 [r159437] Mark Michelson * channels/chan_agent.c: Don't allow for configuration options to overwrite options set via channel variables on a reload. (closes issue #13921) Reported by: davidw Patches: 13921.patch uploaded by putnopvut (license 60) Tested by: davidw 2008-11-26 03:18 +0000 [r159402] Jeff Peeler * main/features.c: Always parse arguments in park_call_exec so that app_args is valid. This prevents a crash when executing Park from the dialplan with no arguments. 2008-11-25 23:03 +0000 [r159360] Steve Murphy * main/cdr.c, /, channels/chan_iax2.c: Merged revisions 159316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines (closes issue #12694) Reported by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17) Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX fix. the change to cdr.c allows no-answer to percolate up into CDR's, and feels like the right place to locate this fix; if BUSY is done here, no-answer should be, too. ........ 2008-11-25 22:45 +0000 [r159276-159317] Tilghman Lesher * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, include/asterisk/dsp.h, CHANGES, main/dsp.c: Add an option, waitfordialtone, for UK analog lines which do not end a call until the originating line hangs up. (closes issue #12382) Reported by: one47 Patches: zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463) Tested by: fleed * /, channels/chan_iax2.c: Merged revisions 159269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines Don't try to send a response on a NULL pvt. (closes issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch uploaded by eliel (license 64) Tested by: barthpbx ........ 2008-11-25 21:49 +0000 [r159250] Mark Michelson * apps/app_followme.c: Make the options for the general and profiles more consistent for the "pls_hold_prompt" option. This does not affect any released version of Asterisk, so there is no need to update the CHANGES file for this. (closes issue #13893) Reported by: eliel 2008-11-25 21:42 +0000 [r159162-159247] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 159246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) | 7 lines Regression fix for last security fix. Set the iseqno correctly. (closes issue #13918) Reported by: ffloimair Patches: 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) Tested by: ffloimair ........ ................ * pbx/pbx_realtime.c: Don't actually do anything with a negative priority, because we ignore it in the result, anyway. * main/pbx.c: Don't limit the length of the hint at the final step (from ~8100 chars max (or ~500 chars max on LOW_MEMORY) to 80 chars max). This will allow more channels to be used in a single hint. 2008-11-25 16:18 +0000 [r159093] Terry Wilson * apps/app_festival.c: Add missing variable declaration for PPC code 2008-11-25 05:19 +0000 [r159050-159054] Tilghman Lesher * apps/app_readexten.c: Copyright clarification; also, have variable set to "t" or "i" on timeout or invalid extension, respectively. (closes issue #13944) Reported by: chappell * channels/chan_usbradio.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/xpmr/xpmr.c, apps/app_rpt.c: Merged revisions 159025 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) | 3 lines System call ioperm is non-portable, so check for its existence in autoconf. (Closes issue #13863) ........ 2008-11-25 03:49 +0000 [r158992] Terry Wilson * channels/chan_usbradio.c: Make chan_usbradio compile under dev mode 2008-11-25 01:01 +0000 [r158959] Sean Bright * funcs/func_dialgroup.c, channels/chan_sip.c, include/asterisk/astobj2.h, res/res_phoneprov.c, main/taskprocessor.c, channels/chan_console.c, channels/chan_iax2.c, apps/app_queue.c, main/astobj2.c, main/config.c, main/manager.c, res/res_timing_pthread.c, main/features.c, res/res_timing_timerfd.c, utils/hashtest2.c, res/res_clialiases.c: This is basically a complete rollback of r155401, as it was determined that it would be best to maintain API compatibility. Instead, this commit introduces ao2_callback_data() which is functionally identical to ao2_callback() except that it allows you to pass arbitrary data to the callback. Reviewed by Mark Michelson via ReviewBoard: http://reviewboard.digium.com/r/64 2008-11-25 00:19 +0000 [r158876-158925] Matthew Nicholson * main/file.c: Fix compiling in dev mode. * UPGRADE.txt, apps/app_queue.c: Make the Join event from app_queue use CallerIDNum insead of CallerID for indicating the callerid number just like the rest of asterisk. (closes issue #13883) Reported by: davidw * main/manager.c, res/res_agi.c, include/asterisk/manager.h: Added EVENT_FLAG_AGI and used it for manager calls in res_agi.c (closes issue #13873) Reported by: fnordian Patches: ami_agievent.patch uploaded by fnordian (license 110) 2008-11-24 21:52 +0000 [r158857] Tilghman Lesher * main/dsp.c: Add a bit of documentation (thanks, I-MOD) on what the silence threshold constant actually does and what values are valid for it. 2008-11-24 21:27 +0000 [r158851] Matthew Nicholson * main/file.c: Make ast_streamfile() check the result of ast_openstream() before doing anything with it. (closes issue #13955) Reported by: chris-mac 2008-11-24 18:11 +0000 [r158808] Terry Wilson * apps/app_minivm.c: This patch adds a new application for sending MWI to phones via Asterisk's event subsystem. Also, the minivm documentation is all converted to use xmldocs. (closes issue #13946) Reported by: Marquis Patches: minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32) Tested by: otherwiseguy, Marquis 2008-11-23 03:36 +0000 [r158754-158756] Sean Bright * channels/chan_sip.c, configs/sip.conf.sample: If you enabled 'notifycid' one of the limitations is that the calling channel is only found if it dialed the extension that was subscribed to. You can now specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as it's value implies, ignore the current context of the caller when doing the lookup. * channels/chan_sip.c: No need to use a separate structure for this since we can just pass our sip_pvt pointer in directly. 2008-11-22 17:17 +0000 [r158686-158723] Michiel van Baak * funcs/func_realtime.c: last commit worked on OpenBSD but still generated warning on Ubuntu. Initialise a variable so --enable-dev-mode does not complain * channels/chan_skinny.c: dont send reorder tone after a device is hungup if a dialout is abandoned or failed. Without this reorder tone will play after hangup and both wedhorn's and my wife have threatened to use an axe on our asterisk box (closes issue #13948) Reported by: wedhorn Patches: switch.diff uploaded by wedhorn (license 30) * channels/chan_skinny.c: Add Media Source Update to skinny's control2str (issue #13948) * channels/chan_skinny.c: fix a very occasional core dump in chan_skinny found by wedhorn. (issue #13948) * funcs/func_realtime.c: make this compile under devmode 2008-11-21 23:40 +0000 [r158606] Steve Murphy * /, main/features.c: Merged revisions 158603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) | 11 lines In reference to the fix made for 13871, I was merging the fix into 1.6.0 and realized I missed the code in the h-exten block, and didn't catch it because my test case had the h-exten commented out. So, this corrects the code I missed, as a preventative against another crash report. Tested with the h-exten defined, all is well. ........ 2008-11-21 23:33 +0000 [r158602-158605] Tilghman Lesher * main/pbx.c: Allow space within an extension, when the space is within a character class. (requested by lmadsen on -dev, patch by me) * main/pbx.c, /: Merged revisions 158600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines The passed extension may not be the same in the list as the current entry, because we strip spaces when copying the extension into the structure. Therefore, use the copied item to place the item into the list. (found by lmadsen on -dev, fixed by me) ........ 2008-11-21 22:12 +0000 [r158540] Russell Bryant * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ 2008-11-21 21:47 +0000 [r158484] Steve Murphy * /, main/features.c: Merged revisions 158483 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | 11 lines (closes issue #13871) Reported by: mdu113 This one is totally my fault. The code doesn't even create a bridge CDR if the channel CDR has POST_DISABLED. I didn't check for that at the end of the bridge. Fixed with a few small insertions. Tested. Looks good. No cdr generated, no crash, no unnecc. data objects created either. ........ 2008-11-21 21:06 +0000 [r158482] Matthew Fredrickson * channels/chan_dahdi.c: Fix for #13963. Make physical channel mapping unconfigured default 2008-11-21 20:42 +0000 [r158449] Kevin P. Fleming * UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, UPGRADE-1.6.txt, CHANGES: as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files 2008-11-21 19:40 +0000 [r158414] Jason Parker * main/manager.c: Make sure we add the Event header for CoreShowChannels. (closes issue #13334) Reported by: srt Patches: 13334_missing_event_header_in_core_show_channel.diff uploaded by srt (license 378) 2008-11-21 17:08 +0000 [r158374] Terry Wilson * cdr/cdr_csv.c: Reloading the config and having no changes still initialized some settings to 0. Initialize settings after doing all of the cfg checks. (closes issue #13942) Reported by: davidw Patches: cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by: davidw 2008-11-21 15:53 +0000 [r158315] Doug Bailey * channels/chan_sip.c: Add fix to prevent crash during reload if there is an outstanding MWI registration message pending. 2008-11-21 01:22 +0000 [r158230-158266] Mark Michelson * channels/chan_sip.c: Use a more expressive constant for a 64-bit scanned int * channels/chan_sip.c: Use some magic constants to get the right size for this sscanf statement. Thanks Richard! * channels/chan_sip.c: Fix the build for 32-bit systems. %lu is only 32-bits on 32-bit systems, so we need to use %llu instead. Of course %llu is 128-bits on 64-bit systems, so we have to cast to unsigned long long. No harm, but it's sure annoying. * channels/chan_sip.c: Change the remote user agent session version variable from an int to a uint64_t. This prevents potential comparison problems from happening if the version string exceeds INT_MAX. This was an apparent problem for one user who could not properly place a call on hold since the version in the SDP of the re-INVITE to place the call on hold greatly exceeded INT_MAX. This also aligns with RFC 2327 better since it recommends using an NTP timestamp for the version (which is a 64-bit number). (closes issue #13531) Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut (license 60) Tested by: sgofferj 2008-11-20 19:41 +0000 [r158188] Sean Bright * res/ael/pval.c: Fix one case where the application argument was not converted from a pipe to a comma. This was causing problems with switch statements with empty expressions. (closes issue #13901) Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by seanbright (license 71) Tested by: seanbright Reviewed by: murf 2008-11-20 18:20 +0000 [r158082-158133] Mark Michelson * include/asterisk/file.h, main/frame.c, /, channels/chan_sip.c, main/file.c, include/asterisk/frame.h: Merged revisions 158072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ * /, channels/chan_sip.c: Merged revisions 158071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines We don't handle 4XX responses to BYE well. According to section 15 of RFC 3261, we should terminate a dialog if we receive a 481 or 408 in response to our BYE. Since I am aware of at least one phone manufacturer who may sometimes send a 404 as well, I am being liberal and saying that any 4XX response to a BYE should result in a terminated dialog. (closes issue #12994) Reported by: pabelanger Patches: 12994.patch uploaded by putnopvut (license 60) Closes AST-129 ........ 2008-11-20 17:53 +0000 [r158078] Ryan Brindley * main/config.c: more formatting corrections :: one line for loops and if statements still need {} 2008-11-20 17:48 +0000 [r158072] Terry Wilson * cdr/cdr_sqlite3_custom.c, cdr/cdr_sqlite.c, cdr/Makefile, cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c, cdr/cdr_csv.c: Begin on a crusade to end trailing whitespace! 2008-11-20 17:46 +0000 [r158070] Ryan Brindley * main/config.c: formatting changes :: one line for loops and if statements should have {} 2008-11-20 17:39 +0000 [r158066] Mark Michelson * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 158053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ 2008-11-20 17:37 +0000 [r158062] Jeff Peeler * main/file.c: (closes issue #12929) Reported by: snyfer This handles the case for a zero length file to attempt to be streamed. Instead of failing from not playing any data, go ahead and return success as ast_streamfile should consider playing nothing a success when there is nothing to play. 2008-11-20 17:37 +0000 [r158061] Jason Parker * README: Whitespace fix 2008-11-20 00:08 +0000 [r157974] Kevin P. Fleming * main/stdtime, /, main/db1-ast/hash, codecs/gsm/Makefile, Makefile.moddir_rules, main/db1-ast/db, channels/misdn, main/db1-ast/mpool, res/ais, res/Makefile, pbx/Makefile, Makefile.rules, res/snmp, main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree, channels/misdn/Makefile, main/db1-ast/recno, res/ael, pbx/ael, channels, main/db1-ast/Makefile: Merged revisions 157859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems. with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course). while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain ........ 2008-11-20 00:06 +0000 [r157973] Terry Wilson * res/res_timing_timerfd.c: Fix compiling 2008-11-19 23:30 +0000 [r157906-157940] Mark Michelson * apps/app_queue.c: Add a space to the output * apps/app_queue.c: Add a RES_NOT_DYNAMIC case for the CLI command 'queue remove member' * CHANGES: Commit CHANGES change I promised when submitting res_timing_timerfd 2008-11-19 22:01 +0000 [r157893] Tilghman Lesher * CHANGES: Add info about REALTIME_FIELD and REALTIME_HASH 2008-11-19 21:55 +0000 [r157874] Mark Michelson * res/res_timing_timerfd.c: Cast this value since a uint64_t is not the same as an unsigned long long on a 64-bit machine. Reported by kpfleming on IRC 2008-11-19 21:54 +0000 [r157870] Tilghman Lesher * funcs/func_realtime.c: Two new functions, REALTIME_FIELD, and REALTIME_HASH, which should make querying realtime from the dialplan a little more consistent and easy to use. The original REALTIME function is preserved, for those who are already accustomed to that interface. (closes issue #13651) Reported by: Corydon76 Patches: 20081119__bug13651__2.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage, Corydon76 2008-11-19 19:37 +0000 [r157820] Mark Michelson * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, res/res_timing_pthread.c, configure.ac, res/res_timing_dahdi.c, res/res_timing_timerfd.c (added), makeopts.in: Merge the changes from the res_timing_timerfd branch. This provides a new timing interface. In order to use it, you must be running a Linux with a kernel version of 2.6.25 or newer and glibc 2.8 or newer. This timing interface is a good alternative if a timing source is necessary (e.g. for IAX trunking) but DAHDI is otherwise unnecessary for the system. For now, this commit contains the actual work done in the res_timing_timerfd branch. There are no notices in the README or CHANGES files yet, but they will be added in my next commit. The timing API of Asterisk also needs to have a bit of work done with regards to choosing which timing interface to use. This commit makes the choice a build-time decision, by only allowing one of the timer interfaces to be chosen in menuselect. It would be preferable if the choice could be made at run-time, however. The preferred timing interface could be loaded and tested, and if it does not work, choice number two may be used instead. That sort of thing. That is beyond the scope of work in this branch though. 2008-11-19 19:25 +0000 [r157818] Terry Wilson * channels/chan_vpb.cc, cdr/cdr_sqlite3_custom.c, channels/iax2-provision.c, cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, cdr/cdr_tds.c, channels/misdn_config.c, cdr/cdr_csv.c, channels/chan_usbradio.c, channels/chan_skinny.c, main/logger.c, res/ais/evt.c, pbx/pbx_dundi.c, cdr/cdr_odbc.c, cdr/cdr_custom.c, cdr/cdr_manager.c, main/xmldoc.c, res/res_clialiases.c: Fix checking for CONFIG_STATUS_FILEINVALID so that modules don't crash upon trying to parse an invalid config 2008-11-19 18:28 +0000 [r157784] Tilghman Lesher * configure, configure.ac: Add check for t38_terminal_init in spandsp (not found in 0.0.6, so it should fail reasonably) (closes issue #13473) Reported by: genie Patches: 20080916__bug13473.diff.txt uploaded by Corydon76 (license 14) 2008-11-19 13:45 +0000 [r157706-157743] Kevin P. Fleming * res/res_agi.c: correct small bug introduced during API conversion * UPGRADE.txt, UPGRADE-1.6.txt: move relevant entries into UPGRADE.txt and resync UPGRADE-1.6.txt with previous branches * include/asterisk/agi.h, res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added), apps/app_stack.c: make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases 2008-11-19 05:37 +0000 [r157675] Terry Wilson * configs/cdr_adaptive_odbc.conf.sample: Comment out config line that is in a commented out context 2008-11-19 01:02 +0000 [r157639] Tilghman Lesher * include/asterisk/logger.h, main/logger.c, main/utils.c, include/asterisk/strings.h: Starting with a change to ensure that ast_verbose() preserves ABI compatibility in 1.6.1 (as compared to 1.6.0 and versions of 1.4), this change also deprecates the use of Asterisk with FreeBSD 4, given the central use of va_copy in core functions. va_copy() is C99, anyway, and we already require C99 for other purposes, so this isn't really a big change anyway. This change also simplifies some of the core ast_str_* functions. 2008-11-19 00:59 +0000 [r157632] Mark Michelson * main/astmm.c: If malloc returns NULL, we need to return NULL immediately or else Asterisk will crash when attempting to dereference the NULL pointer (closes issue #13858) Reported by: eliel Patches: astmm.c.patch uploaded by eliel (license 64) 2008-11-19 00:27 +0000 [r157600] Sean Bright * Makefile, build_tools/make_version, configure, configure.ac, build_tools/make_buildopts_h, makeopts.in: Fix a few build problems on Solaris (and check for an md5 utility in configure instead of the icky loop I was doing before). (closes issue #13842) Reported by: snuffy Patches: bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff uploaded by seanbright (license 71) Tested by: snuffy 2008-11-18 23:59 +0000 [r157496-157592] Mark Michelson * res/res_musiconhold.c: This change prevents a crash from occurring if res_musiconhold.so is unloaded and then Asterisk is stopped. The problem was that we are not unregistering the ast_moh_destroy function at exit. (closes issue #13761) Reported by: eliel Patches: res_musiconhold.c.patch uploaded by eliel (license 64) * Makefile: Add some missing $(DESTDIR)s to the bininstall target of the Makefile. (closes issue #13875) Reported by: pabelanger Patches: Makefile.155928 uploaded by pabelanger (license 224) * apps/app_voicemail.c: Fix the logic for when delete=yes when IMAP storage is in use so that the message is deleted from both local and IMAP storage. (closes issue #13642) Reported by: jaroth Patches: deleteyes.patch uploaded by jaroth (license 50) * channels/chan_sip.c: Merged revisions 157503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines Add some missing invite state changes necessary in the sip_write function. Not setting the invite state correctly on the call was resulting in the Record application leaving empty files. I also have updated the doxygen comment next to the declaration of the INV_EARLY_MEDIA constant to reflect that we also use this state when we *send* a 18X response to an INVITE. (closes issue #13878) Reported by: nahuelgreco Patches: sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162) Tested by: putnopvut ........ * channels/chan_sip.c: Based on Russell's advice on the asterisk-dev list, I have changed from using a global lock in update_call_counter to using the locks within the sip_pvt and sip_peer structures instead. 2008-11-18 21:15 +0000 [r157460-157463] Jason Parker * Makefile: Remove echo line that is unnecessary (Thanks seanbright). * contrib/init.d/rc.archlinux.asterisk: Make this executable * Makefile, contrib/init.d/rc.archlinux.asterisk (added): Add init script for ArchLinux (closes issue #13667) Reported by: sherif Patches: archlinux_rc_makefile.patch uploaded by sherif (license 591) archlinux_rc_makefile-2.patch uploaded by mvanbaak (license 7) 2008-11-18 20:23 +0000 [r157427] Mark Michelson * channels/chan_sip.c: * Add a lock to be used in the update_call_counter function. * Revert logic to mirror 1.4's in the sense that it will not allow the call counter to dip below 0. These two measures prevent potential races that could cause a SIP peer to appear to be busy forever. (closes issue #13668) Reported by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586) 2008-11-18 19:16 +0000 [r157366] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 157365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines (closes issue #13899) Reported by: akkornel This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer. ........ 2008-11-18 18:31 +0000 [r157306] Mark Michelson * apps/app_dial.c, channels/chan_local.c, /, main/features.c, include/asterisk/channel.h, apps/app_followme.c: Merged revisions 157305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ 2008-11-18 18:07 +0000 [r157302] Steve Murphy * main/config.c: (closes issue #13420) Reported by: alex70 Patches: 13420.13539.patch uploaded by murf (license 17) Tested by: murf, awk This fixes two problems: a spurious linefeed insertion probably left over from pre-precomment times. Only generated when category had no previous comments. The other problem: Insertions could get the line-numbering out of whack and generate negative line numbers, causing chunks of line numbers to be emitted, on the scale of the number of lines up to that point in the file. In such cases, abort the looping, and all is well. 2008-11-17 22:25 +0000 [r157253] Tilghman Lesher * apps/app_dial.c: Can't use items duplicated off the stack frame in an element returned from a function: in these cases, we have to use the heap, or garbage will result. (closes issue #13898) Reported by: alecdavis Patches: 20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis 2008-11-15 19:51 +0000 [r157105-157167] Kevin P. Fleming * Makefile.rules: ensure that if a .i file (preprocessed source) is present, the .o file is made from it, not from the .c file (this only works because GNU makes respects the order the rules are defined) * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged revisions 157162-157163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov 2008) | 1 line dist-clean should remove dependency information files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03 +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory dist-clean is run, run clean in that directory first, and when running top-level dist-clean, do not run subdirectory clean operations twice ........ * /, contrib/asterisk-ng-doxygen: Merged revisions 157104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157104 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov 2008) | 13 lines major update to doxygen configuration file: 1) update to doxygen 1.5.x style file, as used in trunk 2) tell doxygen where are header files are, so include-file processing can be done 3) make all macros that are used to define variables/functions be expanded, so that doxygen will properly document the resulting variable/function 4) make all macros that are used to provide the contents of a variable (structure) be expanded, so that doxygen will be able to document the resulting fields 5) suppress compiler attributes (__attribute__(xxx)) from being seen by doxygen, so it will properly match up function definition and usage (for an example of th effect of this, look at the doxygen docs for ast_log() from before and afte this commit) ........ 2008-11-15 15:37 +0000 [r157073] Eliel C. Sardanons * main/xmldoc.c: Avoid a not needed cast, making code more readable. 2008-11-15 04:25 +0000 [r157039-157041] Russell Bryant * channels/chan_sip.c, main/features.c, main/taskprocessor.c: Fix a few more places where the case insensitive hash should be used since the comparison is case insensitive. * channels/chan_console.c: Use the new case insensitive hash function for console interfaces. The comparison function is case insensitive. 2008-11-14 22:36 +0000 [r157006] Tilghman Lesher * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample: Allow setting static values in CDRs 2008-11-14 21:19 +0000 [r156962] Mark Michelson * channels/chan_sip.c: Revision 155513 of chan_sip.c in trunk inadvertently removed a very important line to set the "len" field for incoming SIP requests. The result was that all incoming SIP messages appeared to be 0-length, meaning Asterisk could do no meaningful processing of anything SIP-related 2008-11-14 17:35 +0000 [r156916-156918] Terry Wilson * res/res_phoneprov.c: Cleanup whitespace issues * res/res_phoneprov.c: Use Mark's new ast_str_case_hash function instead of jumping through hoops to do insensitive case lookups 2008-11-14 17:02 +0000 [r156911] Tilghman Lesher * main/manager.c: Ping is missing the standard double-newline after the event. (closes issue #13903) Reported by: kebl0155 2008-11-14 16:53 +0000 [r156883] Mark Michelson * UPGRADE.txt, include/asterisk/strings.h, apps/app_queue.c: Fix some refcounting in app_queue.c and change the hashing used by app_queue.c to be case-insensitive. This is accomplished by adding a new case-insensitive hashing function. This was necessary to prevent bad refcount errors (and potential crashes) which would occur due to the fact that queues were initially read from the config file in a case-sensitive manner. Then, when a user issued a CLI command or manager action, we allowed for case-insensitive input and used that input to directly try to find the queue in the hash table. The result was either that we could not find a queue that was input or worse, we would end up hashing to a completely bogus value based on the input. This commit resolves the problem presented in issue #13703. However, that issue was reported against 1.6.0. Since this fix introduces a behavior change, I am electing to not place this same fix in to the 1.6.0 or 1.6.1 branches, and instead will opt for a change which does not change behavior. 2008-11-14 16:34 +0000 [r156874] Matthew Fredrickson * channels/chan_dahdi.c: Remove some useless locking and make sure we hangup channels on a link when we get a GRS. 2008-11-14 15:20 +0000 [r156817] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 156816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines If the prompt to reenter a voicemail password timed out, it resulted in the password not being saved, even if the input matched what you gave when first prompted to enter a new password. This is because the return value of ast_readstring was checked, but not checked properly. This bug was discovered by Jared Smith during an Asterisk training course. Thanks for reporting it! ........ 2008-11-14 00:43 +0000 [r156690-156756] Tilghman Lesher * /, apps/app_while.c: Merged revisions 156755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines ast_waitfordigit() requires that the channel be up, for no good logical reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me ........ * main/manager.c, /: Merged revisions 156688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines Provide more space for all the data which can appear in an originating channel name. (closes issue #13398) Reported by: bamby Patches: manager.c.diff uploaded by bamby (license 430) ........ 2008-11-13 19:17 +0000 [r156649] Jeff Peeler * main/pbx.c: (closes issue #13891) Reported by: smurfix This reverts a change I made in 116297. At the time it seemed the change was required to solve an issue with attempting a transfer but then letting it timeout without dialing any digits. However, I didn't realize that having an empty extension was possible. I'm removing the immediate return that was added in pbx_find_extension if the extension is null. 2008-11-13 19:10 +0000 [r156647] Tilghman Lesher * channels/chan_dahdi.c: Command offsets were not changed correctly when the command syntax for 'pri set debug' was changed from 'pri debug'. 2008-11-13 17:07 +0000 [r156612] Mark Michelson * configure, autoconf/ast_c_compile_check.m4: Kevin sent a note indicating that this change is not necessary, so I am reverting it 2008-11-13 15:46 +0000 [r156535-156575] Eliel C. Sardanons * apps/app_meetme.c, doc/appdocsxml.dtd, main/xmldoc.c: Introduce XML documentation for: - MeetMe() - MeetMeCount() - MeetMeChannelAdmin() - MeetMeAdmin() - SLAStation() - SLATrunk() - Add an attribute to optionlist 'hasparams' with the same functionality as the one we have in and (the DTD was updated) - Fix a leak when getting an attribute while parsing an . * main/xmldoc.c: Fix a typo introduced when changing xmldoc_has_arguments() to xmldoc_has_inside() we need to pass the name of the node that we are looking for. * include/asterisk/xml.h, include/asterisk/xmldoc.h, main/xmldoc.c: Remove trailing whitespaces using ':%s/\s\+$//' pointed by seanbright on #asterisk-dev 2008-11-12 23:13 +0000 [r156443] Sean Bright * /: Use the reviewboard:url SVN property so post-review will work without modification. 2008-11-12 21:34 +0000 [r156388] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 156386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) | 5 lines When using call limits under 1 second, infinite call lengths are allowed, instead. (closes issue #13851) Reported by: ruddy ........ 2008-11-12 20:27 +0000 [r156355] Eliel C. Sardanons * res/res_clialiases.c: - Make alias->real_cmd point to the allocated space outside alias->alias. - Register the aliased cli command (or we will not alias anything). - Use ARRAY_LEN() when possible. 2008-11-12 19:47 +0000 [r156299] Steve Murphy * main/pbx.c, /: Merged revisions 156297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines It turns out that the 0x0XX00 codes being returned for N, X, and Z are off by one, as per conversation with jsmith on #asterisk-dev; he was teaching a class and disconcerted that this published rule was not being followed, with patterns _NXX, _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should have been. This change, tested on these 3 patterns now picks the proper one. However, this change may surprise users who set up dialplans based on previous behavior, which has been there for what, 2 and half years or so now. ........ 2008-11-12 19:38 +0000 [r156298] Russell Bryant * res/res_clialiases.c: Fix a bug caused by using sizeof(pointer) instead of sizeof(the struct) 2008-11-12 19:28 +0000 [r156295] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 156294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines If the SLA thread is not started, then reload causes a memory leak. (closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) ........ 2008-11-12 19:11 +0000 [r156290] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 156289 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) | 3 lines For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1. ........ 2008-11-12 18:55 +0000 [r156243] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 156229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) | 11 lines Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not to be sent, and instead, schedule a task to destroy the iax2 pvt structure 10 seconds later. This allows the IAX2 HANGUP packet to be queued, transmitted, and ACKed before the pvt is destroyed. (closes issue #13645) Reported by: dzajro Patches: 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14) Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/ ........ 2008-11-12 18:32 +0000 [r156228] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 156178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines (closes issue #13173) Reported by: pep This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference. Reviewed by Russell on Review Board: http://reviewboard.digium.com/r/25/ ........ 2008-11-12 17:41 +0000 [r156169] Mark Michelson * apps/app_dial.c, /: Merged revisions 156167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines When doing some tests, I was having a crash at the end of every call if an attended transfer occurred during the call. I traced the cause to the CDR on one of the channels being NULL. murf suggested a check in the end bridge callback to be sure the CDR is non-NULL before proceeding, so that's what I'm adding. ........ 2008-11-12 17:38 +0000 [r156166] Russell Bryant * /, main/asterisk.c: Merged revisions 156164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines Move the sanity check that makes sure "always fork" is not set along with the console option to be after the code that reads options from asterisk.conf. This resolves a situation where Asterisk can start taking up 100% when misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to figure out what was causing the 100% CPU problem.) ........ 2008-11-12 17:28 +0000 [r156162] Eliel C. Sardanons * main/xmldoc.c: - The paramname is a pointer allocated with strdup() or malloc(), so, we need to free it with ast_free(). 2008-11-12 15:33 +0000 [r156127] Mark Michelson * configure, autoconf/ast_c_compile_check.m4: Add a couple of AC_SUBST calls to the AST_C_COMPILE_CHECK macro. These missing calls were discovered when working on timerfd support in a separate branch. 2008-11-12 13:43 +0000 [r156125] Eliel C. Sardanons * res/res_agi.c: Add XML documentation for AGI commands: - database deltree - database get - exec - get data - get full variable 2008-11-12 06:46 +0000 [r156120] Michiel van Baak * main/udptl.c, main/pbx.c, channels/chan_sip.c, configs/cli_aliases.conf.sample (added), include/asterisk/cli.h, CHANGES, res/res_jabber.c, main/rtp.c, main/cli.c, main/cdr.c, channels/chan_skinny.c, res/res_agi.c, pbx/pbx_ael.c, pbx/pbx_dundi.c, funcs/func_devstate.c, main/asterisk.c, channels/chan_mgcp.c, res/res_clialiases.c (added): This commit does two things: - Add CLI aliases module to asterisk. - Remove all deprecated CLI commands from the code Initial work done by file. Junk-Y and lmadsen did a lot of work and testing to get the list of deprecated commands into the configuration file. Deprecated CLI commands are now handled by this new module, see cli_aliases.conf for more info about that. ok russellb@ via reviewboard (closes issue #13735) Reported by: mvanbaak 2008-11-12 02:20 +0000 [r156051-156087] Eliel C. Sardanons * res/res_agi.c, doc/appdocsxml.dtd: - Add 'database del', 'database put' and 'set music' AGI commands XML documentation. - Add to the DTD the possibility to put a parameter inside an . * include/asterisk/agi.h, res/res_agi.c, doc/appdocsxml.dtd, main/xmldoc.c: Implement AGI XML documentation parsing functions. A new element is used to describe the XML documentation. We have the usual synopsis,syntax,description and seealso for AGI commands. The CLI 'agi show commands' command was changed to show all the documentation se ctions. 2008-11-11 23:32 +0000 [r156017-156018] Pari Nannapaneni * main/manager.c: changing comment style to conform coding guidelines * main/manager.c: Patch by Ryan Brindley -- Make sure that manager refuses any duplicate 'new category' requests in updateconfig 2008-11-11 17:57 +0000 [r155967] Kevin P. Fleming * include/asterisk/strings.h: use some fancy compiler magic (thanks to Matthew Woehlke on the gcc-help mailing list) to restore type-safety to S_OR by going back to a macro, but preserve the side-effect-safe usage of the macro arguments 2008-11-11 16:46 +0000 [r155934] Doug Bailey * res/res_phoneprov.c, phoneprov/polycom_line.xml: Add LINEKEYS variable to allow for a user to set the number of keys assigned to a line on a polycom phone 2008-11-11 16:07 +0000 [r155929] Russell Bryant * channels/chan_sip.c: Remove commentary from the issues list for SIP TCP/TLS 2008-11-10 21:14 +0000 [r155863] Mark Michelson * /, channels/chan_agent.c: Merged revisions 155861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines Channel drivers assume that when their indicate callback is invoked, that the channel on which the callback was called is locked. This patch corrects an instance in chan_agent where a channel's indicate callback is called directly without first locking the channel. This was leading to some observed locking issues in chan_local, but considering that all channel drivers operate under the same expectations, the generic fix in chan_agent is the right way to go. AST-126 ........ 2008-11-10 21:12 +0000 [r155763-155862] Tilghman Lesher * res/res_realtime.c: Make documentation of update method match documentation and update update2 method to match. Reported by: atis, via -dev mailing list. Fixed by: me * /, doc/valgrind.txt: Merged revisions 155803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line I got tired of saying this in every single bugnote referring to this file. ........ * main/editline/readline.c: Fix memory leak when MALLOC_DEBUG is enabled. (closes issue #13864) Reported by: eliel Patches: readline.c.patch uploaded by eliel (license 64) 2008-11-10 13:53 +0000 [r155711] Eliel C. Sardanons * main/pbx.c, main/Makefile, include/asterisk/xmldoc.h (added), include/asterisk/term.h, include/asterisk/_private.h, main/asterisk.c, main/xmldoc.c (added): Move all the XML documentation API from pbx.c to xmldoc.c. Export the XML documentation API: ast_xmldoc_build_synopsis() ast_xmldoc_build_syntax() ast_xmldoc_build_description() ast_xmldoc_build_seealso() ast_xmldoc_build_arguments() ast_xmldoc_printable() ast_xmldoc_load_documentation() 2008-11-09 16:30 +0000 [r155554-155671] Sean Bright * configs/chan_dahdi.conf.sample: Fix this as well. Pointed out by tzafrir. * configs/chan_dahdi.conf.sample: Fix some spelling errors, and convert tabs to spaces. * main/channel.c, channels/chan_sip.c, apps/app_directed_pickup.c, main/features.c, include/asterisk/channel.h: In order to move away from nested function use, some changes to the recently introduced ast_channel_search_locked need to be made. Specifically, the caller needs to be able to pass arbitrary data which in turn is passed to the callback. This patch addresses all of the nested functions currently in asterisk trunk. * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h, apps/app_followme.c, apps/app_queue.c: Merged revisions 155553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ 2008-11-08 21:46 +0000 [r155513-155516] Russell Bryant * channels/chan_sip.c, include/asterisk/strings.h: - Check for failure when putting the packet in the ast_str - fix a spelling error in a header file * channels/chan_sip.c: Remove some code that is basically a no-op. Code above this already ensures that the buffer is terminated. 2008-11-07 23:41 +0000 [r155467] Mark Michelson * channels/chan_sip.c: Set the invite state to INV_CANCELLED in a place that makes more sense. Where it was set before, it was impossible to actually delay sending a CANCEL if we had not yet received a provisional response to an INVITE. (closes issue #13626) Reported by: atis Patches: 13626.patch uploaded by putnopvut (license 60) Tested by: atis 2008-11-07 22:39 +0000 [r155401] Sean Bright * main/manager.c, channels/chan_sip.c, funcs/func_dialgroup.c, res/res_timing_pthread.c, include/asterisk/astobj2.h, main/features.c, res/res_phoneprov.c, utils/hashtest2.c, channels/chan_console.c, main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c, main/astobj2.c, main/config.c: Add ability to pass arbitrary data to the ao2_callback_fn (called from ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either of these mandates that the passed 'arg' is a hashable object, making searching for an ao2 object based on outside criteria difficult. Reviewed by Russell and Mark M. via ReviewBoard: http://reviewboard.digium.com/r/36/ 2008-11-07 22:28 +0000 [r155395-155399] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 155398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines Clarify error message. (closes issue #13809) Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded by Corydon76 (license 14) Tested by: denke ........ * funcs/func_odbc.c: Two bugs relating to colnames found by Marquis42 on #asterisk-dev 2008-11-07 21:14 +0000 [r155360] Mark Michelson * configs/voicemail.conf.sample: Remove one more instance of the sample configuration lying about what's possible. The tz cannot be set in a context like this. It can only be set in the general section or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing this out 2008-11-07 20:13 +0000 [r155324] Tilghman Lesher * channels/chan_dahdi.c: Send call release with unallocated cause instead of normal call clearing, when invalid extension is called. (closes issue #13408) Reported by: adomjan Patches: chan_dahdi.c-ss7-unallocated-2 uploaded by adomjan (license 487) 2008-11-07 16:18 +0000 [r155284] Sean Bright * include/asterisk/indications.h, res/res_indications.c, main/indications.c: Convert open-coded linked list in indications to the AST_LIST_* macros. This cleans the code up some and should make it more maintainable as time goes on. Reviewed by Russell, Kevin, Mark M., and Tilghman via ReviewBoard: http://reviewboard.digium.com/r/34/ 2008-11-07 15:52 +0000 [r155282] Kevin P. Fleming * channels/chan_sip.c: stringfields conversion for struct sip_peer, as requested :-) 2008-11-07 15:42 +0000 [r155241-155264] Russell Bryant * channels/chan_sip.c: Remove a bogus ast_free() that Kevin noticed. This was probably just left over from pre-astobj2ified chan_sip. * include/asterisk/astobj2.h: Clarify which part of OBJ_MULTIPLE is not implemented, and under what case it is perfectly fine to use. (Inspired by a question I received about my last commit.) * main/pbx.c, channels/chan_sip.c: Fix some code in chan_sip that was intended to unlink multiple objects from a container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would only remove a single object. 2008-11-07 03:09 +0000 [r155206] Kevin P. Fleming * pbx/pbx_config.c: correct logic error noticed by rmudgett (thanks!) 2008-11-07 03:02 +0000 [r155175-155204] Eliel C. Sardanons * main/pbx.c: If 'asterisk.conf' is not found, instead of giving up, load documentation for the 'en_US' language (fix my last commit). * main/pbx.c: Fix an asterisk crash if no asterisk.conf configuration file is present. 2008-11-06 22:49 +0000 [r155066-155121] Kevin P. Fleming * res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: don't blindly assume that Darwin and Cygwin need GLOB_ABORTED defined; only define it if it is not already defined * pbx/pbx_config.c: coding style/guidelines cleanup, plus use new side-effect safe S_OR * include/asterisk/strings.h: make S_OR and S_COR safe to use even if the parameters are function calls or have side effects. it still bothers me that these are called S_OR and not something like ast_string_or, but that's water over the bridge * channels/chan_dahdi.c: put ifdef protection around the rest of the libpri function calls that were added at the same time as progress_with_cause move parsing of the qsig channel mapping configuration option outside ifdef HAVE_PRI_INBANDDISCONNECT and into a properly ifdef'd block 2008-11-06 19:46 +0000 [r155012] Mark Michelson * /, configs/voicemail.conf.sample: Merged revisions 155011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov 2008) | 8 lines The documentation listed the ability to set 'maxmsg' per context. The truth is that you can only set this in the general section or per mailbox. Thus I am updating the sample config file to be more accurate. Thanks to sasargen on IRC for bringing up this issue. ........ 2008-11-06 18:19 +0000 [r154967] Eliel C. Sardanons * main/pbx.c: Simplify the output of [See Also]. Functions are printed without parenthesis like: FUNTION Applications are printed with parenthesis like: AppName() Cli commands are printed like: 'core show application' The other type of references are printed as they are inside the tag. 2008-11-05 22:22 +0000 [r154923-154926] Sean Bright * apps/app_directed_pickup.c: Fix some whitespace. * apps/app_directed_pickup.c, main/features.c: Update a couple places to use the new ast_channel_search_locked API call. 2008-11-05 22:19 +0000 [r154922] Tilghman Lesher * main/asterisk.c: Don't read history on -rx commands. (Closes issue #13571) Reported by: tzafrir Patch '0001-no-need-for-history-on-asterisk-rx.patch' uploaded by tzafrir. 2008-11-05 22:01 +0000 [r154919] Sean Bright * include/asterisk.h: Fix a problem found while building res_snmp. 2008-11-05 21:58 +0000 [r154915] Tilghman Lesher * include/asterisk/app.h, funcs/func_strings.c, main/app.c, CHANGES: Add LISTFILTER dialplan function, along with supporting documentation. See documentation for more information on how to use it. 2008-11-05 20:45 +0000 [r154875] Matthew Fredrickson * channels/chan_dahdi.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Make compilation of chan_dahdi so that it does not require the new pri_progress_with_cause function to have libpri support work. 2008-11-05 20:33 +0000 [r154839] Michiel van Baak * res/res_http_post.c: make this compile on OpenBSD again. 2008-11-05 20:17 +0000 [r154796-154837] Eliel C. Sardanons * channels/chan_agent.c: Add AgentLogin(), AgentMonitorOutgoing() applications and AGENT() function XML documentation. * apps/app_test.c: Add TestClient() and TestServer() applications XML documentation. * apps/app_mixmonitor.c: Add more [see also] references based on TFOT. * apps/app_macro.c: Add Macro(), MacroExit(), MacroExclusive() and MacroIf() applications XML documentation. (closes issue #13699) Reported by: snuffy Patches: bug13699_20081016.diff uploaded by snuffy (license 35) 2008-11-05 16:11 +0000 [r154687] Steve Murphy * main/channel.c, /: Merged revisions 154685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1 line This fix was prompted by communication from user, who was seeing thousands of error logs... looks like EAGAIN. Made such uninteresting. ........ 2008-11-05 14:37 +0000 [r154467-154647] Eliel C. Sardanons * main/pbx.c, apps/app_privacy.c, apps/app_sayunixtime.c, main/features.c, apps/app_morsecode.c, apps/app_alarmreceiver.c, apps/app_amd.c: Add more SeeAlso references based on TFOT. * doc/appdocsxml.dtd: We now can have a reference to a filename inside a tag. * apps/app_parkandannounce.c: - Add ParkAndAnnounce() application XML documentation. * main/pbx.c, apps/app_page.c, apps/app_authenticate.c, apps/app_dumpchan.c, apps/app_disa.c, apps/app_image.c, apps/app_chanspy.c, apps/app_stack.c, apps/app_adsiprog.c: - Add more based on TFOT. - Add the 'filename' type to the see-also ref. To be able to reference a filename. * apps/app_readfile.c, funcs/func_db.c, apps/app_sendtext.c, funcs/func_blacklist.c, apps/app_url.c, apps/app_queue.c, apps/app_senddtmf.c, apps/app_db.c: - Add some see-also references based on TFOT. * apps/app_read.c: - Add Read() application XML documentation. * apps/app_followme.c: - Add FollowMe() application XML documentation. * apps/app_forkcdr.c, res/res_indications.c: - Add PlayTones() and StopPlayTones() applications XML documentation. - Fix a dot that was outside of the in the ForkCDR() XML documentation. 2008-11-04 23:23 +0000 [r154429] Sean Bright * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h: Introduce a new API call ast_channel_search_locked, which iterates through the channel list calling a caller-defined callback. The callback returns non-zero if a match is found. This should speed up some of the code that I committed earlier today in chan_sip (which is also updated by this commit). Reviewed by russellb and kpfleming via ReviewBoard: http://reviewboard.digium.com/r/28/ 2008-11-04 23:03 +0000 [r154366-154428] Tilghman Lesher * channels/chan_iax2.c: Switch to using a thread condition to signal that a child thread is ready for work, rather than a busy wait. (closes issue #13011) Reported by: jpgrayson Patches: chan_iax2_find_idle.patch uploaded by jpgrayson (license 492) * /, channels/chan_iax2.c: Merged revisions 154365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines On busy systems, it's possible for the values checked within a single line of code to change, unless the structure is locked to ensure a consistent state. (closes issue #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ 2008-11-04 20:12 +0000 [r154329] Eliel C. Sardanons * Makefile: We need to pass the DTD to xmlstarlet to validate against it the XML. (I thought it was being read within the DOCTYPE definition inside the XML). 2008-11-04 19:07 +0000 [r154268] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 154266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state when it receives the indication AST_CONTROL_RINGING. ........ 2008-11-04 18:59 +0000 [r154260-154264] Tilghman Lesher * /, channels/chan_skinny.c, channels/chan_h323.c: Recorded merge of revisions 154263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines Make the monitor thread non-detached, so it can be joined (suggested by Russell on -dev list). ........ * include/asterisk/devicestate.h, main/manager.c, apps/app_page.c, include/asterisk/config.h, main/features.c, main/devicestate.c, apps/app_queue.c, main/config.c, apps/app_voicemail.c: Slightly optimize ast_devstate_str and rename global functions devstate2str and config_text_file_save to have an ast_ prefix 2008-11-04 18:06 +0000 [r154225] Eliel C. Sardanons * apps/app_forkcdr.c: Add XML documentation for the ForkCDR() application. 2008-11-04 17:23 +0000 [r154186-154191] Sean Bright * main/pbx.c: GLOB_BRACE is already added to MY_GLOB_FLAGS if it is supported on the platform. This should resolve some build errors on Solaris. (issue #13704) Reported by: dougm * channels/chan_sip.c, configs/sip.conf.sample: Allow devices that accept dialog-info+xml (like snoms) to get the Caller ID of the calling party when subscribed to the state of an extension that is ringing. This has some limitations which are documented in sip.conf.sample. (closes issue #13827) Reported by: seanbright Patches: issue13827.patch uploaded by seanbright (license 71) Reviewed by: russellb * main/Makefile: Fix build errors. 2008-11-04 15:07 +0000 [r154151] Kevin P. Fleming * channels/chan_vpb.cc, res/res_crypto.c, configure.ac, cdr/cdr_adaptive_odbc.c, channels/chan_oss.c, channels/chan_usbradio.c, res/res_config_odbc.c, apps/app_osplookup.c, funcs/func_odbc.c, configure, build_tools/menuselect-deps.in, channels/chan_alsa.c, makeopts.in, cdr/cdr_odbc.c, res/res_odbc.c, apps/app_voicemail.c: improve configure script to remember the previous value of each dependency in build_tools/menuselect-deps, so that (once it has been written) menuselect can use this information to warn the user when a previously met dependency is no longer met along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named 2008-11-04 14:38 +0000 [r154149] Eliel C. Sardanons * channels/chan_dahdi.c: Add XML documentation for: Applications - DAHDISendKeypadFacility() - DAHDISendCallreroutingFacility() 2008-11-03 22:28 +0000 [r154023-154072] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 154066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008) | 5 lines Attempting to expunge a mailbox when the mailstream is NULL will crash Asterisk. (Closes issue #13829) Reported by: jaroth Patch by: me (modified jaroth's patch) ........ * /, main/rtp.c: Merged revisions 154060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines Remove the potential for a division by zero error. (Closes issue #13810) ........ * funcs/func_odbc.c: Should have passed the string pointer, not the ast_str structure. (closes issue #13830) Reported by: Marquis 2008-11-03 18:02 +0000 [r153983] Olle Johansson * configs/sip.conf.sample: Updating docs 2008-11-03 17:11 +0000 [r153947] Eliel C. Sardanons * apps/app_stack.c: Add LOCAL() function XML documentation. 2008-11-03 15:25 +0000 [r153904-153905] Olle Johansson * configs/sip.conf.sample: Spaces to replace tabs... * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding a separation of remote authentication and our authentication. remotesecret => our password for a remote service secret => our authentication when someone calls us Secret => still has both functions if remotesecret is not used. 2008-11-03 13:33 +0000 [r153803-153852] Eliel C. Sardanons * channels/chan_iax2.c: Add XML documentation for: Functions - IAXPEER() - IAXVAR() * channels/chan_sip.c: Add XML documentation for: Applications - SIPDtmfMode() - SIPAddHeader() Functions - SIP_HEADER() - SIPPEER() - SIPCHANINFO() - CHECKSIPDOMAIN() 2008-11-03 12:26 +0000 [r153787] Kevin P. Fleming * configure, autoconf/ast_ext_lib.m4: when --without- is passed to the configure script, explicitly inform menuselect that the package was disabled by the user 2008-11-03 01:01 +0000 [r153747] Eliel C. Sardanons * apps/app_waitforring.c, apps/app_waitforsilence.c, apps/app_db.c, apps/app_ivrdemo.c: Add XML documentation for: - WaitForSilence() - WaitForNoise() - WaitForRing() - IVRDemo() - DBDel() - DBDeltree() (issue #13699) Reported by: snuffy Patches: bug13699_20081016.diff uploaded by snuffy (license 35) (With minor changes) 2008-11-02 23:34 +0000 [r153709] Kevin P. Fleming * include/asterisk/agi.h, configure, include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4, configure.ac, include/asterisk/compiler.h, apps/app_stack.c: instead of trying to forcibly load res_agi when app_stack is loaded (even if the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it. 2008-11-02 20:06 +0000 [r153652] Russell Bryant * /, include/asterisk/features.h: Merged revisions 153651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008) | 2 lines features.h depends on linkedlists.h, so include it ........ 2008-11-02 19:39 +0000 [r153616-153650] Kevin P. Fleming * channels/chan_dahdi.c: fix one more warning missed because i did not have new enough libpri installed * res/res_musiconhold.c: fix small bug introduced while cleaning up compiler warnings * /: mark this revision as merged manually * utils/muted.c, apps/app_authenticate.c, res/res_phoneprov.c, main/utils.c, formats/format_wav_gsm.c, res/res_http_post.c, res/res_musiconhold.c, channels/chan_iax2.c, res/res_jabber.c, res/res_config_sqlite.c, utils/frame.c, utils/stereorize.c, main/channel.c, channels/chan_dahdi.c, main/manager.c, res/ael/ael.tab.c, funcs/func_odbc.c, main/ast_expr2f.c, res/res_agi.c, main/http.c, main/logger.c, formats/format_gsm.c, apps/app_adsiprog.c, apps/app_dial.c, channels/chan_sip.c, apps/app_festival.c, formats/format_wav.c, res/ael/ael.y, main/db1-ast/hash/hash_page.c, agi/eagi-test.c, res/res_crypto.c, utils/astman.c, pbx/pbx_lua.c, formats/format_ogg_vorbis.c, utils/astcanary.c, apps/app_queue.c, channels/chan_oss.c, agi/eagi-sphinx-test.c, res/ael/ael_lex.c, channels/chan_h323.c, main/file.c, apps/app_sms.c, pbx/pbx_dundi.c, res/ael/ael.flex, pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c, utils/streamplayer.c, main/asterisk.c, apps/app_voicemail.c: bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 branch, and add the ones needed for all the new code here too 2008-11-02 06:24 +0000 [r153582] Eliel C. Sardanons * channels/chan_iax2.c: Add IAX2Provision() application XML documentation. 2008-11-02 05:56 +0000 [r153577-153580] Russell Bryant * Makefile: validate-docs is a PHONY target * Makefile, configure, configure.ac, makeopts.in: Add a handy makefile target so that you can validate the documentation against the DTD by running "make validate-docs" * Makefile: Modify the Makefile logic for extracting documentation. - Build the documentation when you run "make", as opposed to "make install" - Only rebuild the documentation when source code has been changed 2008-11-02 05:10 +0000 [r153541-153543] Eliel C. Sardanons * apps/app_flash.c: Add Flash() application XML documentation. * apps/app_talkdetect.c: Fix a typo in the name of the application. 2008-11-02 04:14 +0000 [r153472-153507] Sean Bright * channels/Makefile: There is a troublesome assert() in the alsa/control.h header that causes GCC 4.3.2 to complain that the passed argument will always evaluate to true. So to get things to compile, disable assert when building chan_usbradio.so. * apps/app_record.c: Another little one. 2008-11-02 02:55 +0000 [r153362-153470] Russell Bryant * apps/app_page.c: fix a typo (thanks sean) * apps/app_dial.c, funcs/func_speex.c, apps/app_page.c, apps/app_record.c, funcs/func_env.c, apps/app_dahdiras.c, funcs/func_math.c, funcs/func_strings.c, apps/app_userevent.c, apps/app_exec.c, apps/app_chanspy.c, apps/app_playback.c: Fix various spelling and grammatical issues in documentation * apps/app_voicemail.c: - Use a for loop instead of a while loop - Get rid of an unnecessary variable * apps/app_directed_pickup.c: Instead of doing a couple of strlen() calls each iteration of the loop, only do it once at the beginning of the function * channels/chan_sip.c: Don't ignore the result of find_peer() when looking for a peer by IP in check_peer_ok(). * funcs/func_speex.c, apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, funcs/func_module.c, funcs/func_dialgroup.c, include/asterisk/autoconfig.h.in, funcs/func_env.c, apps/app_dahdiscan.c, apps/app_record.c, funcs/func_strings.c, apps/app_sayunixtime.c, include/asterisk/extconf.h, apps/app_alarmreceiver.c, apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c, main/config.c, main/term.c, include/asterisk/compat.h, configure, funcs/func_shell.c, apps/app_skel.c, apps/app_dumpchan.c, include/asterisk/module.h, main/features.c, apps/app_amd.c, apps/app_url.c, apps/app_milliwatt.c, apps/app_dial.c, main/pbx.c, include/asterisk/xml.h (added), apps/app_page.c, funcs/func_timeout.c, main/Makefile, apps/app_privacy.c, apps/app_echo.c, apps/app_softhangup.c, apps/app_fax.c, funcs/func_math.c, apps/app_dahdiras.c, configure.ac, apps/app_disa.c, apps/app_morsecode.c, funcs/func_cut.c, apps/app_talkdetect.c, apps/app_transfer.c, apps/app_playback.c, doc/tex/asterisk-conf.tex, Makefile, apps/app_sendtext.c, funcs/func_channel.c, funcs/func_cdr.c, apps/app_zapateller.c, build_tools/get_documentation (added), funcs/func_iconv.c, apps/app_mixmonitor.c, apps/app_chanspy.c, main/asterisk.c, apps/app_cdr.c, funcs/func_base64.c, funcs/func_md5.c, apps/app_dictate.c, apps/app_authenticate.c, apps/app_readexten.c, apps/app_userevent.c, funcs/func_vmcount.c, main/xml.c (added), funcs/func_sha1.c, funcs/func_logic.c, funcs/func_uri.c, apps/app_controlplayback.c, funcs/func_enum.c, apps/app_setcallerid.c, funcs/func_groupcount.c, funcs/func_config.c, funcs/func_volume.c, funcs/func_odbc.c, apps/app_mp3.c, apps/app_directory.c, apps/app_jack.c, apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c, funcs/func_dialplan.c, funcs/func_db.c, funcs/func_version.c, apps/app_festival.c, funcs/func_lock.c, apps/app_waituntil.c, doc, include/asterisk/term.h, include/asterisk/_private.h, apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c, funcs/func_global.c, funcs/func_extstate.c, funcs/func_realtime.c, apps/app_channelredirect.c, funcs/func_blacklist.c, apps/app_directed_pickup.c, include/asterisk/pbx.h, include/asterisk/strings.h, makeopts.in, apps/app_senddtmf.c, funcs/func_devstate.c, funcs/func_callerid.c, doc/appdocsxml.dtd (added), apps/app_verbose.c, apps/app_stack.c: Merge changes from team/group/appdocsxml This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html * channels/chan_sip.c: Ensure that the sip_pvt properly has its refcount incremented when the scheduler holds a reference to it for session timer processing. 2008-11-01 01:55 +0000 [r153296] Sean Bright * configs/sip.conf.sample: The default in chan_sip for notifyringing is yes, so update the sample conf to reflect that. 2008-10-31 20:05 +0000 [r153223] Mark Michelson * main/dial.c, apps/app_page.c, include/asterisk/dial.h, CHANGES: * Fixed timeout logic in the dialing API as setting timeouts had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. 2008-10-31 18:55 +0000 [r153181] Terry Wilson * apps/app_dial.c, main/features.c, include/asterisk/channel.h, apps/app_followme.c, apps/app_queue.c: Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech 2008-10-31 17:18 +0000 [r153122-153124] Tilghman Lesher * configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Failover for func_odbc, allowing an INSERT query to be performed when the UPDATE query initially affects 0 rows. (closes issue #13083) Reported by: Corydon76 Patches: 20081031__bug13083.diff.txt uploaded by Corydon76 (license 14) * /, channels/chan_sip.c: Merged revisions 153114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) | 3 lines Turn off qualify on uncached realtime peers. (Closes issue #13383) ........ 2008-10-31 09:31 +0000 [r153057] Russell Bryant * main/channel.c: Use the ast_str API call to reset the string instead of manually editing its internals (closes issue #13816) Reported by: eliel Patches: channel.c.patch uploaded by eliel (license 64) 2008-10-30 20:59 +0000 [r152993] Sean Bright * /, bootstrap.sh: Merged revisions 152992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct 2008) | 2 lines The -I argument to aclocal needs a space before the include directory name. ........ 2008-10-30 20:46 +0000 [r152990] Russell Bryant * include/asterisk/timing.h: Add a todo for a new timing API implementation that would work for Linux systems as of kernel 2.6.25 and glibc 2.8 2008-10-30 20:35 +0000 [r152923-152969] Tilghman Lesher * /, channels/chan_h323.c: Merged revisions 152958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) | 3 lines Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html (Closes issue #13400) ........ * channels/chan_local.c, /: Merged revisions 152922 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152922 | tilghman | 2008-10-30 14:43:38 -0500 (Thu, 30 Oct 2008) | 6 lines Unlock before returning, when extension doesn't exist. (closes issue #13807) Reported by: eliel Patches: chan_local.c.patch uploaded by eliel (license 64) ........ 2008-10-30 19:40 +0000 [r152887-152920] Russell Bryant * channels/chan_sip.c: Fix the sip_peer reference count with respect to scheduler entries for scheduling peer pokes, and scheduling peer poke expirations. * channels/chan_sip.c: Fix the sip_peer reference count with respect to scheduler entries for registration expirations. * include/asterisk/sched.h: Fix a bug in AST_SCHED_REPLACE_UNREF(). The reference count of the object _must_ be increased before creating the scheduler entry. Otherwise, you create a race condition where the reference count may hit zero and the object can disappear out from under you. This could also would have incorrectly decreased the reference count in the case that the scheduler add failed. 2008-10-30 19:23 +0000 [r152879] Mark Michelson * channels/chan_sip.c: I just noticed this construct and thought it was silly to have a bunch of case statements with duplicated code in each case. Instead, just use the built-in fallthrough capability of case statements and reduce the code to a single instance 2008-10-30 19:21 +0000 [r152875-152877] Russell Bryant * channels/chan_sip.c: Modify the documentation of the sip_registry struct - Remove a comment that says that the monitor thread is the only one that ever touches these objects. This is no longer the case with TCP. Also, I would eventually like to get the scheduler in its own thread, so this is just a poor assumption to make. - Note that reference counting of these objects with respect to scheduler entries is not complete. There are some leaked references when deleting scheduler entries. * funcs/func_db.c: - spaces to tabs - add some braces - remove unnecessary cast 2008-10-30 16:54 +0000 [r152809-152812] Kevin P. Fleming * main/cdr.c, /: Merged revisions 152811 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct 2008) | 3 lines instead of comparing the string pointer to 0, let's compare the value that was actually parsed out of the string (found by sparse) ........ * include/asterisk/buildinfo.h (added): try to get this committed before the buildbot complains about a broken tree * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, main/dial.c, main/dnsmgr.c, main/buildinfo.c, codecs/lpc10/chanwr.c, utils/astcanary.c, channels/misdn/isdn_lib.c, main/asterisk.c, apps/app_adsiprog.c: fix a few small things found by using sparse 2008-10-30 16:38 +0000 [r152807] Mark Michelson * main/features.c, CHANGES, configs/features.conf.sample: After seeing another problem in #asterisk stemming from the low default value of featuredigittimeout, I decided it was high time to change it. I have changed the default to 2000 ms based on a suggestion from Leif Madsen. 2008-10-30 04:26 +0000 [r152689-152765] Tilghman Lesher * configs/extensions.conf.sample: Set up an example stdexten that preserves the original context and extension in the CDR. (Related to issue #13799) Reported by: davidw * CHANGES, apps/app_directory.c: Pay attention to the searchcontexts entry in voicemail.conf (related to AST-125) * main/pbx.c: Track down and fix annoying lock errors 2008-10-29 20:53 +0000 [r152646] Mark Michelson * apps/app_directory.c: If there was no named defined in a voicemail.conf mailbox entry, then app_directory would crash when attempting to read that entry from the file. We now check for the NULL or empty string properly so that there will be no crash. (closes issue #13804) Reported by: bluecrow76 2008-10-29 05:47 +0000 [r152605] Steve Murphy * apps/app_dial.c, /, apps/app_queue.c, configs/features.conf.sample: Merged revisions 152538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines A little documentation cross-ref between features and dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... ........ 2008-10-29 05:34 +0000 [r152569] Russell Bryant * /, channels/chan_sip.c: Merged revisions 152539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) | 7 lines Fix an incorrect usage of sizeof() (closes issue #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch uploaded by andrew53 (license 519) ........ 2008-10-29 05:01 +0000 [r152536] Steve Murphy * apps/app_dial.c, /, main/features.c, include/asterisk/pbx.h, apps/app_queue.c, include/asterisk/features.h: Merged revisions 152535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. Why? because CDR's aren't generated via parking, so nothing is needed, but if a transfer occurred, there are critical things I need. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. ........ 2008-10-28 22:33 +0000 [r152467] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 152463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008) | 3 lines Quoting in the wrong direction (Fixes AST-107) ........ 2008-10-28 22:26 +0000 [r152448] Doug Bailey * configs/phoneprov.conf.sample: Add more polycom firmware files to static mapping 2008-10-28 21:38 +0000 [r152369-152442] Tilghman Lesher * channels/chan_mgcp.c: Only re-add the io port if it was closed, otherwise reload causes a memory leak. (closes issue #13785) Reported by: eliel Patches: chan_mgcp.c.patch uploaded by eliel (license 64) * apps/app_dial.c, /: Merged revisions 152368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines Reset all DIAL variables back to blank, in case Dial is called multiple times per call (which could otherwise lead to inconsistent status reports). (closes issue #13216) Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 (license 14) Tested by: ruddy ........ 2008-10-27 23:31 +0000 [r152287] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 152286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008) | 2 lines Buffer policy setting for half is not needed. ........ 2008-10-27 21:34 +0000 [r152134-152216] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 152215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008) | 6 lines Inherit ALL elements of CallerID across a local channel. (closes issue #13368) Reported by: Peter Schlaile Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 (license 14) ........ * apps/app_stack.c: Set ARGC in subroutines with the number of arguments passed. * apps/app_stack.c: Oops, only delete the ARG variables once upon release. The following section would have removed them again (removing variables from 2 stack frames, instead of just one). 2008-10-27 16:03 +0000 [r152132] Jason Parker * apps/app_transfer.c: Remove options argument parsing/syntax (it isn't used any longer) (closes issue #13789) Reported by: IgorG Patches: app_transfer.c.diff uploaded by IgorG (license 20) 2008-10-26 20:25 +0000 [r152060] Sean Bright * /, funcs/func_strings.c: Merged revisions 152059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct 2008) | 7 lines Since passing \0 as the second argument to strchr is valid (and will match the trailing \0 of a string) we need to check that first, otherwise we end up with incorrect results. Fix suggested by reporter. (closes issue #13787) Reported by: meitinger ........ 2008-10-26 10:23 +0000 [r151980-152020] Olle Johansson * channels/chan_sip.c: Trying to fix the user/peer matching correctly. This will need some testing before getting merged into 1.6.1 * channels/chan_sip.c: Moving more variables to the sip_cfg structure, as I have some future ideas for the usage of that structure. * channels/chan_sip.c: Doxygen changes and some formatting. 2008-10-25 11:02 +0000 [r151906] Russell Bryant * /, main/asterisk.c: Merged revisions 151905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008) | 8 lines Move AMI initialization to occur after loading modules. This prevents a deadlock when someone tries to initiate a module reload from the AMI just as Asterisk is starting. (closes issue #13778) Reported by: hotsblanc Fix suggested by hotsblanc ........ 2008-10-23 21:27 +0000 [r151830] Terry Wilson * funcs/func_odbc.c: allow to compile under --enable-dev-mode (gcc didn't actually complain when I was using ccache...) 2008-10-23 15:54 +0000 [r151762] Tilghman Lesher * contrib/scripts/vmdb.sql: Clarify documentation, following merge of realtime_update2 branch 2008-10-23 15:38 +0000 [r151739-151761] Olle Johansson * CHANGES: Thanks russellb for reminding an old man.... * channels/chan_sip.c, doc/tex/channelvariables.tex: Adding a small new feature. Setting _SIPFROMDOMAIN in a channel will set the domain we use for the URI in the outbound call leg. 2008-10-23 15:28 +0000 [r151732] Tilghman Lesher * funcs/func_odbc.c: Simplify some nested functions, as suggested by Russell on -dev 2008-10-23 15:09 +0000 [r151722] Doug Bailey * res/res_http_post.c: Add patch to handle how IE7 issues POST requests using Window path spec including backslash delimiters 2008-10-22 22:11 +0000 [r151682] Tilghman Lesher * funcs/func_odbc.c, CHANGES: Added debugging CLI functions 2008-10-22 20:45 +0000 [r151642] BJ Weschke * channels/chan_sip.c: revert the changes in issue #13705 - it's being re-opened as while the results fixed the complaint in the issue, it introduced other more undesirable issues than what was already reported 2008-10-22 20:05 +0000 [r151601] Tilghman Lesher * contrib/scripts/live_ast (added): Add a contributed script for running Asterisk without installing it, first. (closes issue #11680) Reported by: tzafrir Patches: live_ast_6 uploaded by tzafrir (license 46) 2008-10-22 20:05 +0000 [r151600] Mark Michelson * channels/chan_dahdi.c: Change some logical ands to bitwise ands and add messages alerting that a channel is being ignored if the PROC_DAHDI_NOCHAN option is set in process_dahdi. (closes issue #13759) Reported by: smurfix Patches: dahdi.patch uploaded by smurfix (license 547) 2008-10-22 17:45 +0000 [r151554-151555] Russell Bryant * channels/chan_sip.c: Print out the right var in the log message * channels/chan_sip.c: Fix this check to use the proper variable (the result from get_in_brackets) 2008-10-22 15:08 +0000 [r151420-151512] Mark Michelson * channels/chan_sip.c: The logic of a strncasecmp call was reversed. (closes issue #13706) Reported by: andrew53 Patches: sip_notify_from_rfc3265.patch uploaded by andrew53 (license 519) * channels/chan_sip.c: Make the sip_standard_port function more granular by allowing separate type and port arguments. This is necessary because when building our From and Contact headers, we need to be absolutely sure that we are placing our source port there and not the peer's source port. (closes issue #12761) Reported by: asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455) * channels/chan_sip.c: Get this compiling in dev-mode * channels/chan_sip.c: If a peer uses any transport other than UDP, then MWI will fail for that peer since sip_alloc will allocate a sip_pvt with a default transport of UDP. This change resets the socket type immediately after allocating the sip_pvt in sip_send_mwi_from_peer, so that the proceeding call to create_addr_from_peer does not fail right away. The socket data from the peer is properly copied to the sip_pvt in create_addr_from_peer. (closes issue #13710) Reported by: andrew53 Patches: sip_notify_use_tcp.patch uploaded by andrew53 (license 519) * channels/chan_sip.c: When attempting to resolve hostnames, we need to be sure to remove any parameters from the string so that name resolution succeeds. (closes issue #13727) Reported by: fnordian Patches: resolvewithouturiparameter.patch uploaded by fnordian (license 110) 2008-10-21 15:20 +0000 [r151371] Tilghman Lesher * apps/app_mixmonitor.c: Default file modes should always be full read and write, to allow the system administrator to make the decision of what permissions will actually be given, through the use of the process umask. (Closes issue# 13751) 2008-10-21 11:02 +0000 [r151327] BJ Weschke * channels/chan_sip.c: Fix configuration parsing so type=friend still identifies "friend" as a peer even though it is now a legacy configuration verb. (closes issue #13705) reported by: blitzrage patched by: bweschke 2008-10-20 05:07 +0000 [r151246] BJ Weschke * pbx/pbx_config.c, main/config.c: Do NOT attempt to do anything with the ast_config struct when it's been returned as INVALID by the config file interpreter. (closes issue #13741) 2008-10-20 05:00 +0000 [r151242-151243] Kevin P. Fleming * autoconf/ast_check_pwlib.m4, /, autoconf/ast_check_openh323.m4, configure.ac: Merged revisions 151241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct 2008) | 2 lines rename this macro to properly reflect what it does ........ * autoconf/ast_prog_egrep.m4, autoconf/ast_c_define_check.m4, autoconf/ast_ext_tool_check.m4 (added), autoconf/ast_check_mandatory.m4 (added), /, autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4, autoconf/ast_prog_sed.m4, acinclude.m4 (removed), autoconf/ast_check_pwlib.m4, autoconf (added), autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure, autoconf/ast_gcc_attribute.m4, bootstrap.sh, autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4, autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4: Merged revisions 151240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct 2008) | 3 lines break up acinclude.m4 into individual files, which will make it easier to maintain, easier to add new macros (less patching) and will ease maintenance of these macros across Asterisk branches ........ 2008-10-19 20:30 +0000 [r151188-151190] BJ Weschke * /: Block 151167 from coming forward into the /trunk this is a 1.4 fix only. * /: Block 151100 from coming forward into the /trunk this is a 1.4 fix only. 2008-10-19 19:11 +0000 [r151101] Kevin P. Fleming * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c, apps/app_externalivr.c, include/asterisk/tcptls.h: cleaup of the TCP/TLS socket API: 1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines 2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines) 3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines) 4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied 5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address 2008-10-19 13:10 +0000 [r151060] Michiel van Baak * channels/chan_skinny.c: dont segfault when placing a call to a line that has no registered device. 2008-10-19 07:20 +0000 [r151019] Olle Johansson * channels/chan_sip.c: Adding changes from train and flight back home from SIPit23 in Lannion, France. - Additional comments on TCP/TLS implementation - Some additions for new drafts/rfcs (no new functionality really, mostly documentation) - Other random small fixes 2008-10-18 10:27 +0000 [r150930-150971] Michiel van Baak * Makefile: Make sure we support nested functions and generation of trampolines under OpenBSD. (closes issue #13724) Reported by: mvanbaak * contrib/init.d/rc.mandriva.asterisk, contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.redhat.asterisk, contrib/init.d/rc.suse.asterisk: dont use deprecated commands in the init scripts. (closes issue #13720) Reported by: decryptus_proformatique Patches: contrib_initd_module_reload.patch uploaded by decryptus (license 555) With mods by me to fix stop commands as well 2008-10-18 03:35 +0000 [r150773-150887] BJ Weschke * apps/app_authenticate.c, CHANGES: Give app_authenticate the ability to select a prompt other than the default. (closes issue #13734) reported and patched by: jvandal * main/manager.c, /: Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766. We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing. (closes issue #13715) reported by: makoto patch by: bweschke * doc/manager_1_1.txt, CHANGES, apps/app_queue.c: The QueueEntry event now has the uniqueid of the channel included. (closes issue #13731) reported and patched by: caio1982 2008-10-17 21:48 +0000 [r150731] Matthew Fredrickson * configure, configure.ac: Update configure check to check for new function in libpri (pri_progress_with_cause) 2008-10-17 21:35 +0000 [r150729] Jason Parker * codecs/codec_adpcm.c, codecs/ex_g722.h (added), codecs/codec_gsm.c, codecs/ex_adpcm.h (added), codecs/ex_alaw.h (added), codecs/ex_g726.h (added), codecs/ex_gsm.h (added), codecs/slin_ulaw_ex.h (removed), codecs/slin_lpc10_ex.h (removed), codecs/codec_resample.c, codecs/slin_g722_ex.h (removed), codecs/g722_slin_ex.h (removed), codecs/ex_ulaw.h (added), codecs/adpcm_slin_ex.h (removed), codecs/ex_ilbc.h (added), codecs/slin_adpcm_ex.h (removed), codecs/g726_slin_ex.h (removed), codecs/slin_g726_ex.h (removed), codecs/codec_lpc10.c, codecs/gsm_slin_ex.h (removed), codecs/slin_gsm_ex.h (removed), codecs/codec_a_mu.c, codecs/codec_g722.c, codecs/ex_lpc10.h (added), codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c, include/asterisk/slin.h (added), codecs/ex_speex.h (added), codecs/slin_resample_ex.h (removed), codecs/ulaw_slin_ex.h (removed), codecs/slin_ilbc_ex.h (removed), codecs/ilbc_slin_ex.h (removed), codecs/lpc10_slin_ex.h (removed), codecs/codec_ulaw.c, codecs/codec_ilbc.c, codecs/speex_slin_ex.h (removed), codecs/slin_speex_ex.h (removed): Merge codec_consistency branch. This should make sample usage much happier. 2008-10-17 17:31 +0000 [r150664] Michiel van Baak * main/cli.c: Fix CLI command 'channel request hangup' Prodded on IRC by Russell and fixed by eliel (closes issue #13730) Reported by: eliel Patches: main_cli.patch uploaded by eliel (license 64) 2008-10-17 17:25 +0000 [r150640] Matthew Fredrickson * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Merge in patch for #13454. Includes CallRereouting dialplan application, option for discard of remote hold messages, and using the alternate logical channel mapping in Q.SIG instead of the default physical channel mapping. 2008-10-17 17:09 +0000 [r150580-150635] Tilghman Lesher * channels/chan_iax2.c: Make helper call a little safer (suggested by Russell on IRC) * include/asterisk/sched.h, channels/chan_iax2.c: Fix the FRACK! warnings in chan_iax2 when POKE/LAGRQ packets are not answered. 2008-10-17 08:42 +0000 [r150469-150510] Olle Johansson * channels/chan_sip.c: Adding some additional thoughts on configuration changes to TCP/TLS * Makefile: Make sure we support nested functions with GCC 4.01 OS/X. This might not be OS/X only, but I'll leave it to kpfleming to add this to the configure script for testing. 2008-10-17 06:00 +0000 [r150426] Michiel van Baak * channels/chan_skinny.c, UPGRADE.txt, configs/skinny.conf.sample, CHANGES: Break up skinny.conf into seperate sections for devices and lines. (closes issue #13412) Reported by: wedhorn Patches: config-restruct-v4.diff uploaded by wedhorn (license 30) 2008-10-17 04:28 +0000 [r150384] Tilghman Lesher * apps/app_meetme.c: Fix option handling code. (closes issue #11040) Reported by: DEA Patches: rt-meetme-flag-fixes-v2.txt uploaded by DEA (license 3) with additional fixes by me 2008-10-17 00:18 +0000 [r150311] Mark Michelson * doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Add an IAXregistry manager command. See doc/manager_1_1.txt for more details of this command. (closes issue #13326) Reported by: ib2 Patches: bug13326_trunk_20080822.diff uploaded by snuffy (license 35) 2008-10-17 00:14 +0000 [r150309] Jeff Peeler * apps/app_meetme.c: Initialize character arrays as they are not guaranteed to be set. 2008-10-17 00:13 +0000 [r150207-150307] Mark Michelson * channels/chan_sip.c: After a long discussion on #asterisk-bugs, it seems kind of odd that a channel would be named after the originating port. For endpoints that always include ":5060" as part of the From: header, it will mean that you have a ton of channels with names like "SIP/5060-3ea38a8b." I am boldly moving forward with this change in trunk, but I'm not touching other branches with this one since this definitely would qualify as a behavior change. If there is a problem with this commit, and I haven't seen the obvious reason why you'd want to name the channel after the port from which the call originated, then please feel free to revert this * main/manager.c, /: Merged revisions 150304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct 2008) | 6 lines Reverting changes from commits 150298 and 150301 since I was mistakenly under the assumption that dialplan functions *always* required that a channel be present. I need to go home earlier, I think :) ........ * main/manager.c: Merged revisions 150298,150301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct 2008) | 10 lines Don't try to call a dialplan function's read callback from the manager's GetVar handler if an invalid channel has been specified. Several dialplan functions, including CHANNEL and SIP_HEADER, do not check for NULL-ness of the channel being passed in. (closes issue #13715) Reported by: makoto ........ r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct 2008) | 3 lines And don't forget to return on the error condition ........ * apps/app_sms.c: Answer the channel prior to checking for the 'a' option in app_sms. (closes issue #13675) Reported by: alecdavis Patches: app_sms.bug13675.148985.diff.txt uploaded by alecdavis (license 585) * apps/app_skel.c: Updating app_skel.c to follow coding guidelines with regards to braces used on if statements. (closes issue #13696) Reported by: alecdavis Patches: app_skel.bug13696B.115850.diff.txt uploaded by alecdavis (license 585) * channels/chan_iax2.c: Remove an odd redundant comparison * configure, configure.ac: Change configure script to search for openais in both /usr/lib and /usr/lib64 since some distros place 64-bit libraries only in the /usr/lib64 directory. (closes issue #13721) Reported by: jcollie Patches: 0007-Look-in-64bit-dirs-for-openais.patch uploaded by jcollie (license 412) * channels/chan_sip.c: INVITES with proxy auth were sent with a different branch than what was in the invite_branch of a sip_pvt, meaning that if a CANCEL were sent later, the branch in the CANCEL would not match the branch in the latest INVITE sent out, leading to some endpoints responding to the CANCEL with a 481. (closes issue #13714) Reported by: fnordian Patches: invite_branch.patch uploaded by fnordian (license 110) 2008-10-16 16:04 +0000 [r150125] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 150124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16 Oct 2008) | 1 line Fix memory leak found by customer ........ 2008-10-16 15:48 +0000 [r150118-150121] Terry Wilson * configs/modules.conf.sample: This is nolonger needed * res/res_phoneprov.c: func_strings isn't a dependency of this module anymore 2008-10-16 15:02 +0000 [r150052] Kevin P. Fleming * channels/chan_sip.c: ensure that type=peer entries are only matched on IP/port, not on name (after oej audits all the calls to find_peer() to make sure that forcenamematch is set correctly in each case) 2008-10-16 15:00 +0000 [r150008-150051] Olle Johansson * channels/chan_sip.c: Doxygen addition * channels/chan_sip.c: Add some notes on problems with the TCP/TLS implementation 2008-10-16 13:28 +0000 [r149917-149981] Kevin P. Fleming * channels/chan_sip.c: return this logic to where it used to be, *after* the dialog->needdestroy flag has been determined to be set; otherwise, we generate these debug messages every time we inspect every active dialog * channels/chan_sip.c: some additional debugging tools added at SIPit23: - move all setting of 'needdestroy' on dialog structures into the history - report all tags involved when a pedantic check fails on a REFER * res/res_phoneprov.c: inter-module dependencies should be included in the source code, not just in sample config files * res/res_phoneprov.c: correct file name in message * configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES: support relative paths in musiconhold.conf, which makes moh work by default when Asterisk was configured using --prefix and 'make samples' is run 2008-10-15 21:36 +0000 [r149848] BJ Weschke * /: Blocking 149840 from coming forward. 2008-10-15 20:55 +0000 [r149802] Mark Michelson * channels/chan_sip.c: Make the sip_proxy struct reference counted. This is necessary to allow for a sip_pvt to maintain a reference to a sip_peer's outboundproxy even after the peer has been freed. (closes issue #13700) Reported by: fnordian Patches: 13700.patch uploaded by putnopvut (license 60) Tested by: fnordian 2008-10-15 20:14 +0000 [r149756] BJ Weschke * configs/agents.conf.sample, /: Merged revisions 149683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008) | 4 lines An update to the documentation/example of agents.conf.sample with the correct parameter for this feature as defined in chan_agent.c (closes issue #13709) ........ 2008-10-15 19:07 +0000 [r149588-149687] Tilghman Lesher * funcs/func_odbc.c: Permit data fields to contain more than 255 characters. (closes issue #13631) Reported by: seanbright Patches: 20081015__bug13631.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage * funcs/func_odbc.c: Only set buf to blank before the goto. * codecs/lpc10/lpcini.c: When using MALLOC_DEBUG, codec_lpc10 leaks memory, because it matches a library malloc() with an ast_free (which, of course, doesn't match up with known allocated memory, so the free fails). (closes issue #13702) Reported by: eliel Patches: codec_lpc10_lpcini.c uploaded by eliel (license 64) * apps/app_echo.c: Minor spacing change (closes issue #13697) Reported by: alecdavis Patches: app_echo.bug13697.103249.diff.txt uploaded by alecdavis (license 585) 2008-10-15 13:52 +0000 [r149542] Olle Johansson * channels/chan_sip.c: Adding a note about a missing part of "kill-the-user" - I got lost in the Ao2 world... We're going to try to get time to fix this and kpfleming believes that there's code in ao2 so that we can solve it... 2008-10-15 11:26 +0000 [r149384-149487] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 149452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct 2008) | 3 lines fix some problems when parsing SIP messages that have the maximum number of headers or body lines that we support ........ * configure, configure.ac: reverting this change... had not read the commit list yet, didn't realize the code had been upgraded * configure, configure.ac: do complete version check for SpanDSP, since the app_fax code is not compatible with 0.0.6 yet * apps/app_stack.c: building this module depends on res_agi being built as well 2008-10-15 07:45 +0000 [r149342] Olle Johansson * channels/chan_sip.c: Fixing sytax errors ;-) 2008-10-14 23:57 +0000 [r149201-149279] Mark Michelson * apps/app_dial.c, CHANGES: When specifying an invalid timeout to Dial, take it to mean that no timeout is desired. (closes issue #13625) Reported by: atis * /, channels/chan_sip.c: Merged revisions 149266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149266 | mmichelson | 2008-10-14 18:43:58 -0500 (Tue, 14 Oct 2008) | 4 lines Change this warning to an error message. Suggestion comes from Sean Bright. Thanks Sean! ........ * /, channels/chan_sip.c: Merged revisions 149207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines Call register_peer_exten even in the case that the peer's IP/port does not change. (closes issue #13309) Reported by: dimas Patches: v2-13309.patch uploaded by dimas (license 88) ........ * /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 149204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ * /, apps/app_queue.c: Merged revisions 149200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines Update the queue with the correct number of calls and whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) ........ 2008-10-14 22:38 +0000 [r149199] Tilghman Lesher * main/hashtab.c, pbx/pbx_spool.c, channels/chan_sip.c, include/asterisk/chanvars.h, include/asterisk/config.h, include/asterisk/strings.h, res/res_indications.c, include/asterisk/hashtab.h, main/chanvars.c, main/config.c: Add additional memory debugging to several core APIs, and fix several memory leaks found with these changes. (Closes issue #13505, closes issue #13543) Reported by: mav3rick, triccyx Patches: 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) Tested by: mav3rick, triccyx 2008-10-14 21:08 +0000 [r149131] Mark Michelson * /, channels/chan_sip.c: Merged revisions 149130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines Don't allow reserved characters to be used in register lines in sip.conf. (closes issue #13570) Reported by: putnopvut ........ 2008-10-14 20:16 +0000 [r149062] Tilghman Lesher * /, apps/app_waitforsilence.c: Merged revisions 149061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) | 6 lines Check correct values in the return of ast_waitfor(); also, get rid of a possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me ........ 2008-10-14 19:35 +0000 [r149040] Leif Madsen * doc/manager_1_1.txt: Add missing documentation for SipShowRegistry action and RegistryEntry event. (closes issue #13342) Reported and patch by: Laureano 2008-10-14 19:03 +0000 [r148917-148988] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 148987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) | 2 lines Some compilers warn, some don't. Fixing. ........ * apps/app_sms.c: App is ignoring 'p' parameter -- initial pause. (closes issue #13617) Reported by: alecdavis Patches: app_sms.13oct.diff.txt uploaded by alecdavis (license 585) * /, apps/app_voicemail.c: Merged revisions 148916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) | 4 lines Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used in headers like 'Subject' and 'To'. Closes AST-107. ........ 2008-10-14 17:38 +0000 [r148913] Mark Michelson * channels/chan_local.c, /: Merged revisions 148912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, 14 Oct 2008) | 9 lines Deadlock prevention in chan_local. (closes issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded by putnopvut (license 60) Tested by: tacvbo ........ 2008-10-14 15:15 +0000 [r148868] Tilghman Lesher * apps/app_fax.c: API differences in spandsp 0.0.6pre1 and higher (closes issue #13688) Reported by: irroot Patches: app_fax-span6.patch uploaded by irroot (license 52) with minor modifications by me 2008-10-14 15:00 +0000 [r148867] Joshua Colp * channels/chan_sip.c: Fix reference count issue that Russell brought up in SIP MWI NOTIFY support. Bump the reference count up before we add it to the scheduler, duh. 2008-10-14 14:18 +0000 [r148825] Doug Bailey * phoneprov/polycom.xml: Allow MWI registration for all configured lines. 2008-10-14 11:31 +0000 [r148695-148754] Kevin P. Fleming * channels/chan_sip.c: fix some references to the owner of a private structure that may not be present * Makefile, /: Merged revisions 148736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct 2008) | 3 lines on Ubuntu (at least), recent versions of ld in binutils delete all debugging symbols when -x is supplied; since the reasons why -x is being passed are lost in the mists of time, remove it so debugging will work properly ........ * channels/chan_sip.c: this structure should be static * channels/chan_sip.c: ensure that *all* fields in the req structure are cleared out before reusing it; has_to_tag was not cleared, which caused the second incoming call over a TCP socket to fail if pedantic checking was enabled 2008-10-14 09:16 +0000 [r148679] Olle Johansson * channels/chan_sip.c: Adding some clarifications 2008-10-14 08:06 +0000 [r148612] Kevin P. Fleming * /, main/translate.c: Merged revisions 148611 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct 2008) | 3 lines it would be nice if this message printing code had actually been tested before it was committed... ........ 2008-10-14 00:08 +0000 [r148570] Tilghman Lesher * res/res_config_curl.c, res/res_config_pgsql.c, res/res_config_odbc.c, include/asterisk/config.h, res/res_realtime.c, include/asterisk/strings.h, res/res_config_ldap.c, res/res_config_sqlite.c, main/config.c, apps/app_voicemail.c: Merge realtime_update2 branch, which adds a new realtime API call named 'update2', which permits updates which match across multiple columns, instead of requiring all tables to have a single unique identifier. All of the other API calls with the exception of 'update' already had the ability to match on multiple fields, so it was a missing and very desireable feature that an API call implementing an update should have this, too. This does not change any outward performance of Asterisk, but it should make life easier for application developers who use the RealTime framework. 2008-10-13 17:14 +0000 [r148519] Steve Murphy * main/pbx.c: Hmmm. Nobody (but me) is interested in seeing the trie info when they do 'dialplan show ...' (even with debug set to non-zero); so I set up a 'dialplan debug [context]' cli command instead, to explicitly show just the trie info. I even added an extension_exists() call to make sure the trie info is built. I moved the explanatory header to above the extension loop to ensure it only prints once. And it will do this now, whether debug is set or not. I removed the trie printing from the 'dialplan show' command entirely. 2008-10-13 15:56 +0000 [r148471-148474] Olle Johansson * channels/chan_sip.c: - Doxygen formatting. (tss tss) - Fixing language * main/tcptls.c, channels/chan_sip.c: Highlightning even more bugs in the current tcp/tls implementation. * channels/chan_sip.c: Sending a 403 after a 200 is considered very bad. (found at SIPit) 2008-10-12 09:19 +0000 [r148425] Michiel van Baak * res/res_agi.c: fix the 'agi show commands' CLI function. (closes issue #13666) Reported by: eliel Patches: res_agi.c.patch uploaded by eliel (license 64) 2008-10-10 21:21 +0000 [r148373-148376] Mark Michelson * channels/chan_sip.c: The logic used when checking a peer got changed subtly in the "kill the user" commit and caused calls relying on the insecure setting to not work properly. I changed for finding a peer back to how it was prior to that commit. (closes issue #13644) Reported by: pj Patches: 13644_trunkv2.patch uploaded by putnopvut (license 60) Tested by: pj * channels/chan_sip.c: Make sure that the inUse and inRinging fields for a sip peer cannot go below zero. This is a regression from 1.4 and so it will be applied to 1.6.0 as well. (closes issue #13668) Reported by: mjc 2008-10-10 18:59 +0000 [r148268-148329] Tilghman Lesher * pbx/pbx_config.c: Reset continuation items at the beginning of each context (suggested by kpfleming). * CHANGES, pbx/pbx_config.c: Add keyword "same", which allows you to create multiple steps in a dialplan, without needing to respecify an extension pattern multiple times. (closes issue #13632) Reported by: blitzrage Patches: 20081006__bug13632.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage, Corydon76 * /, apps/app_voicemail.c: Merged revisions 148257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) | 7 lines User not notified of temporary greeting, if ODBC storage is in use. (closes issue #13659) Reported by: moliveras Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 (license 14) Tested by: moliveras ........ 2008-10-10 00:42 +0000 [r148200] Sean Bright * include/asterisk.h, main/tdd.c, main/cryptostub.c, res/res_config_sqlite.c, apps/app_voicemail.c: Don't include logger.h in asterisk.h by default as it is causing problems building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson 2008-10-09 23:54 +0000 [r148144-148160] Mark Michelson * main/manager.c: The priority was unnecessary for the manager atxfer, so it has been removed. Furthermore, now we actually use the Context argument passed to set the transfer context and don't error out if no context is specified. This addresses the actual problems outlined in issue 12158. Regarding the other points brought up, regarding the inability to not transfer to extensions which cannot be represented by DTMF, it is not enough of a constraint that it is worth attempting to rework the feature. (closes issue #12158) Reported by: davidw * apps/app_voicemail.c: Read the callerid in the correct order and make sure to read the Urgent flag value from the IMAP headers. (closes issue #13652) Reported by: jaroth Patches: imapheaders.patch uploaded by jaroth (license 50) 2008-10-09 23:25 +0000 [r148120] Tilghman Lesher * configs/res_ldap.conf.sample: Fix example schema (closes issue #12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded by flyn (license 503) 2008-10-09 23:15 +0000 [r148112] Mark Michelson * /, main/features.c: Merged revisions 146026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ 2008-10-09 21:47 +0000 [r148000-148071] Tilghman Lesher * formats/format_wav.c, apps/app_minivm.c, channels/chan_agent.c, main/file.c, res/res_monitor.c, apps/app_voicemail.c: Reverting format addition for now * apps/app_minivm.c, channels/chan_agent.c, main/file.c, res/res_monitor.c, apps/app_voicemail.c: Fudges for wav16, just like wav49 * formats/format_wav.c: Add native 16kHz format for wav file format. (Closes issue #13657) * sounds/sounds.xml, sounds/Makefile: Publish MOH files in sln16 format * /, apps/app_voicemail.c: Merged revisions 147997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) | 4 lines When blank, callerid name and number should display "unknown caller" in voicemail emails. (Closes issue #13643) ........ 2008-10-09 19:27 +0000 [r147952] Jeff Peeler * main/features.c: (closes issue #13139) Reported by: krisk84 Tested by: krisk84 This change prevents a call that is placed in the parkinglot to be picked up before the PBX is finished. If another extension dials the parking extension before the PBX thread has completed at minimum warnings will occur about the PBX not properly being terminated. At worst, a crash could occur. 2008-10-09 17:48 +0000 [r147899] Michiel van Baak * include/asterisk/endian.h: only include this for OpenBSD. At least FreeBSD is borked when including it (closes issue #13649) Reported by: ys 2008-10-09 17:46 +0000 [r147896] Tilghman Lesher * configs/extensions.conf.sample: Remove "second form" of extensions, as it no longer applies. Also, cleanup the grammar, formatting, and introduce several clarifications to the text. (Closes issue #13654) 2008-10-09 17:04 +0000 [r147854] Terry Wilson * phoneprov/000000000000.cfg, res/res_phoneprov.c, configs/phoneprov.conf.sample: Make phoneprov case-insensitive to remove the func_strings dependency of the default config 2008-10-09 17:01 +0000 [r147853] Michiel van Baak * channels/chan_dahdi.c, channels/chan_misdn.c, channels/chan_h323.c: fix some CLI commands we borked during devcon2008 Thanks rmudget for letting me know and providing hints on how to fix it best. 2008-10-09 14:17 +0000 [r147807] Steve Murphy * main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES, include/asterisk/autoconfig.h.in, channels/vcodecs.c, main/translate.c, configure.ac, channels/console_video.c, channels/chan_iax2.c, main/astobj2.c, channels/chan_oss.c, main/rtp.c, main/config.c, main/cli.c, channels/chan_usbradio.c, configure, channels/console_gui.c, utils/extconf.c: (closes issue #13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in 2008-10-09 01:43 +0000 [r147760-147761] Joshua Colp * configs/sip.conf.sample: *whistle* * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add support for subscribing to a voice mailbox on a remote SIP server and making the new/old message count available to local devices. (issue #AST-77) 2008-10-08 22:32 +0000 [r147714] Mark Michelson * apps/app_meetme.c: Some small tweaks regarding realtime conference announcements. (closes issue #13522) Reported by: DEA Patches: meetme-rt-fixes.txt uploaded by DEA (license 3) 2008-10-08 22:26 +0000 [r147689] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 147681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected) ........ 2008-10-08 20:07 +0000 [r147635] Sean Bright * configs/voicemail.conf.sample: Add some examples of IMAP accounts. 2008-10-08 19:08 +0000 [r147592] Tilghman Lesher * apps/app_sms.c: Correct a typo in the help; also, ensure that the date and time are correctly set, if not specified in the message. (Closes issue #13594, closes issue #13595) Reported by: alecdavis Patches: 20081001__bug13595.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis 2008-10-08 14:53 +0000 [r147518] Joshua Colp * /, apps/app_speech_utils.c: Merged revisions 147517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) ........ 2008-10-08 12:28 +0000 [r147476] Bradley Latus * configs/iax.conf.sample: Adjust commented default trunkmtu value to match documentation above it 2008-10-08 12:15 +0000 [r147388-147457] Sean Bright * funcs/func_curl.c, apps/app_meetme.c, cdr/cdr_adaptive_odbc.c, res/res_odbc.c: Keep up with shadow warnings. One day I'll actually enable this in the Makefile. * utils/Makefile: When echoing our copies, strip off ASTTOPDIR from the front of the source file. * apps/app_dial.c, channels/chan_dahdi.c, channels/chan_iax2.c: Move the DAHDI-to-DAHDI operator mode check from app_dial into chan_dahdi so we don't have to hardcode anything. (closes issue #13636) Reported by: seanbright Patches: 13636.diff uploaded by seanbright (license 71) Reviewed by: russellb, putnopvut 2008-10-07 20:15 +0000 [r147266-147347] Michiel van Baak * configure, configure.ac: Make format_vorbis_ogg compile on OpenBSD (closes issue #13639) Reported by: mvanbaak Patches: 2008100700_oggsupportOBSD.diff.txt uploaded by mvanbaak (license 7) 2008100700_oggsupportOBSD-configurescript.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak * channels/Makefile: make this work on OpenBSD * configure, configure.ac: Make sure the configs on OpenBSD are in /etc/asterisk by default (closes issue #13641) Reported by: jtodd * contrib/scripts/safe_asterisk_restart, contrib/scripts/safe_asterisk: use pkill instead of killall to be more portable 2008-10-07 18:00 +0000 [r147265] Sean Bright * apps/app_voicemail.c: This was flawed. The issue that I was trying to address was addressed by adding the imapsecret alias for imappassword. Will rethink this one and give it another shot on a rainy day TBD. 2008-10-07 17:49 +0000 [r147264] Michiel van Baak * CHANGES: fix wording as pointed out by Corydon 2008-10-07 17:44 +0000 [r147262] Tilghman Lesher * UPGRADE.txt, include/asterisk/options.h, main/asterisk.c, main/term.c: Allow people to select the old console behavior of white text on a black background, by using the startup flag '-B'. 2008-10-07 16:52 +0000 [r147191-147194] Sean Bright * /, apps/app_voicemail.c: Merged revisions 147193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147193 | seanbright | 2008-10-07 12:48:30 -0400 (Tue, 07 Oct 2008) | 2 lines Make 'imapsecret' an alias to 'imappassword' in voicemail.conf. ........ * apps/app_voicemail.c: Or not. * apps/app_voicemail.c: There was a boo-boo in TFOT that is causing some confusion on the mailing lists so include 'imapsecret' as an alias to 'imappassword' (and print a little notice nudging users toward the right option name). 2008-10-07 16:04 +0000 [r147146] Jeff Peeler * main/features.c: Explicitly setting these fields to NULL was done because I wasn't sure if they would be NULL otherwise. Since they will be set automatically, removing. 2008-10-07 14:59 +0000 [r147050-147099] Sean Bright * apps/app_voicemail.c: If we encounter something in mailbox options that we don't grok, then spit out a warning instead of just silently ignoring it. * apps/app_dial.c: Make sure to compare the correct number of characters when special-casing our DAHDI operator mode stuff. Technically, it would work fine, as 'DAH' is currently unique amongst our channel technologies, but as Jared points out: <@jsmith> Sure... as long as the technology starts whith DAH.... but it could be DAHDOO! 2008-10-07 02:02 +0000 [r147011] Richard Mudgett * funcs/func_callerid.c: Independent change from branch issue8824 that is not part of COLP. (-r142574 rmudgett) 2008-10-07 00:02 +0000 [r146970] Terry Wilson * channels/chan_sip.c: A blind transfer to the parking thread would cause a segfault because copy_request accesses dst->data w/o being able to tell whether it is proerly initialized 2008-10-06 23:21 +0000 [r146928] Tilghman Lesher * include/asterisk/threadstorage.h: Update documentation; AST_THREADSTORAGE() in trunk only takes a single argument. 2008-10-06 23:14 +0000 [r146925] Michiel van Baak * res/res_config_odbc.c, build_tools/menuselect-deps.in, configure, funcs/func_odbc.c, include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, makeopts.in, res/res_odbc.c, apps/app_voicemail.c: All ODBC parts can now use either unixodbc or iodbc. This allows for the ODBC parts to work on OpenBSD as well. 99.99% of the work is done by seanbright (bow, bow) and I actually did nothing but test and yell at him that it still didn't work :) Thanks for helping out ! 2008-10-06 23:08 +0000 [r146875-146923] Jeff Peeler * main/features.c, res/res_agi.c, include/asterisk/features.h: Similar to r143204, masquerade the channel in the case of Park being called from AGI. * include/asterisk/endian.h: Mvanbaak said this was needed to compile on OpenBSD, so put it in the OpenBSD section. * main/features.c: This commit squashes together three commits because the wrong approach was originally used. (One of the commits was only one line.) 1) r143204: The main change here was to masquerade the channel if the channel that was to be parked was running a PBX on it. The PBX thread can then maintain full control of the channel (the zombie) as it expects to while allowing the parking thread full control of the real (parked) channel. 2) r143270: Changed park_call_full to hold the parkinglot lock a little longer, which protects the parkeduser struct from being freed out from underneath. Made sure that the parking extension is added to the parking context while holding the lock thereby ensuring that there are no spurious warnings from removal attempts when a hangup occurs while the parking lot is being announced. 3) r143475: (the one liner) compare peer and chan instead of looking at the parked user (pu), which could have possibly already have been freed by the parking thread * main/features.c: fix some comment placement * main/features.c: Explicitly set args in park_call_exec NULL so in the case of no options being passed in, there is no garbage attempted to be used. Also, do not set args to unknown value again if there are no options passed in. 2008-10-06 21:18 +0000 [r146807] Michiel van Baak * include/asterisk/endian.h: make aescrypt.c compile on OpenBSD again 2008-10-06 21:09 +0000 [r146802] Tilghman Lesher * funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c, /, channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c, funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c, funcs/func_callerid.c, apps/app_speech_utils.c: Merged revisions 146799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines Dialplan functions should not actually return 0, unless they have modified the workspace. To signal an error (and no change to the workspace), -1 should be returned instead. (closes issue #13340) Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14) ........ 2008-10-06 17:32 +0000 [r146738] Sean Bright * configure, configure.ac: Pretty-print a couple configure options 2008-10-06 16:52 +0000 [r146713] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 146711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146711 | tilghman | 2008-10-06 11:51:21 -0500 (Mon, 06 Oct 2008) | 9 lines Check whether an extension exists in the _call method, rather than the _alloc method, because we need to evaluate the callerid (since that data affects whether an extension exists). (closes issue #13343) Reported by: efutch Patches: 20080915__bug13343.diff.txt uploaded by Corydon76 (license 14) Tested by: efutch ........ 2008-10-06 16:03 +0000 [r146644] Kevin P. Fleming * /: Merged revisions 146643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146643 | kpfleming | 2008-10-06 10:57:49 -0500 (Mon, 06 Oct 2008) | 8 lines ensure that the private structure for pseudo channels is created without 'leaking' configuration data from other configured channels (closes issue #13555) Reported by: jeffg Patches: issue_13555.patch uploaded by kpfleming (license 421) Tested by: jeffg ........ 2008-10-06 15:29 +0000 [r146640] Mark Michelson * configs/queues.conf.sample, CHANGES, apps/app_queue.c: This commit introduces a change to how the "joinempty" and "leavewhenempty" options are configured in queues.conf. Instead of using vague terms like "yes," "no," "loose," and "strict," we now accept a comma-separated list of values to determine when to consider a member available. Extended details can be found in the queues.conf.sample file. Note also that the above four referenced values are still accepted for backwards-compatibility, but are mapped internally to the new method of representing the option. AST-105 2008-10-06 00:36 +0000 [r146555-146597] Sean Bright * utils/Makefile: Make NOISY_BUILD work for the calls to cp in utils/Makefile * utils/Makefile: Quote arguments to cp so we can handle spaces in our paths. 2008-10-05 22:11 +0000 [r146514] Russell Bryant * utils/muted.c: Make this build on my mac. 2008-10-05 21:21 +0000 [r146449] Jason Parker * /, channels/chan_sip.c: Recorded merge of revisions 146448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146448 | qwell | 2008-10-05 16:17:44 -0500 (Sun, 05 Oct 2008) | 1 line Fix silly formatting. ........ 2008-10-05 01:59 +0000 [r146312-146407] Sean Bright * build_tools/make_buildopts_h: This is far from optimal, but I just found a FreeBSD system without md5 installed on it. So look around for all of the different binaries that we could possibly use. I'd wager this gets completely replaced by someone else in less than 24 hours... :) * main/asterisk.c: Fix a bug with the last item in CLI history getting duplicated when read from the .asterisk_history file (and subsequently being duplicated when written). We weren't checking the result of fgets() which meant that we read the same line twice before feof() actually returned non- zero. Also, stop writing out an extra blank line between each item in the history file, fix a minor off-by-one error, and use symbolic constants rather than a hardcoded integer. * configs/sip_notify.conf.sample: Add ability to remotely reboot snom phones. Also cleaned up and reorganized sip_notify.conf.sample a bit as well. Tested snom reboot on snom 360 and verified snom-check-cfg worked as well. (closes issue #13601) Reported by: mjc Tested by: seanbright 2008-10-03 22:40 +0000 [r146242] Jeff Peeler * main/features.c: remove superfluous reference counting operations in manage_parkinglot since ao2_interator_next increments the ref count automatically 2008-10-03 22:10 +0000 [r146198] Sean Bright * main/cli.c: Resolve a subtle bug where we would never successfully be able to get the first item in the CLI entry list. This was preventing '!' from showing up in either 'help' or in tab completion. (closes issue #13578) Reported by: mvanbaak 2008-10-03 18:30 +0000 [r146081] Tilghman Lesher * CHANGES: document meetme schedule changes (related to issue #11040) 2008-10-03 17:36 +0000 [r146053] Michiel van Baak * CHANGES: put a note in CHANGES about the cli_cleanup done during AstriDevCon 2008-10-03 17:35 +0000 [r146052] Terry Wilson * main/dial.c: The dialing API should inherit datastores as well as variables 2008-10-02 19:30 +0000 [r145959-145962] Russell Bryant * CHANGES: The 'P' command for ExternalIVR was also added in 1.6.0 * CHANGES: TCP support for ExternalIVR went in to 1.6.1, not 1.6.0 2008-10-02 18:02 +0000 [r145915] Michiel van Baak * apps/app_meetme.c: fix the 'meetme list', 'meetme list concise', 'meetme list $confno' and 'meetme list $confno concise' CLI commands (closes issue #13586) Reported by: john8675309 Help and feedback from eliel, thanks! 2008-10-02 17:16 +0000 [r145846] Tilghman Lesher * configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Permit the syntax and synopsis fields to be set (for func_odbc). 2008-10-02 16:42 +0000 [r145842] Michiel van Baak * apps/app_meetme.c: make this compile under devmode again 2008-10-02 15:28 +0000 [r145771] Sean Bright * configure, configure.ac: This is much cleaner, methinks. 2008-10-02 15:17 +0000 [r145752] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 145751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r145751 | tilghman | 2008-10-02 10:13:21 -0500 (Thu, 02 Oct 2008) | 3 lines Some sanity checks that may have led to prior crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup of incorrectly-used constants. ........ 2008-10-01 23:48 +0000 [r145692] Sean Bright * configure, configure.ac: Try a test compile using the GMime library. Some distros install gmime-config in the base package instead of the -devel package. Now we print a notice and disable GMime support instead of bombing during the main compilation. (closes issue #13583) Reported by: arkadia 2008-10-01 23:02 +0000 [r145649] Tilghman Lesher * apps/app_meetme.c, funcs/func_strings.c, include/asterisk/localtime.h, main/stdtime/localtime.c: Add schedule extensions to app_meetme. In addition, the reporter found a problem within strptime(3), which we are correcting here with ast_strptime(). (closes issue #11040) Reported by: DEA Patches: 20080910__bug11040.diff.txt uploaded by Corydon76 (license 14) Tested by: DEA 2008-10-01 22:23 +0000 [r145553-145606] Mark Michelson * main/features.c: Okay, this should really do it now. While I did manage to fix blind transfers with my last commit here, I also caused an unwanted side-effect. That is, only the first priority of the 'h' extension would be executed when a blind transfer occurred instead of all priorities. Essentially, my last commit corrected the return value of ast_bridge_call. However, the implementation still was not 100% correct. Now it is. * main/features.c: if (!(x) == 0) is the same as if (x). * main/features.c: The logic surrounding the return value of ast_spawn_extension within ast_bridge_call was reversed. This problem was observed when a blind transfer placed from the callee channel of a test call failed. While the problem I am solving here is exactly the same as what was reported in issue #13584, the difference is that this fix I am applying is trunk-only. Issue #13584 was reported against the 1.4 branch, and my tests of 1.4's blind transfers appear to work fine. 2008-10-01 17:26 +0000 [r145487] Leif Madsen * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 145479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r145479 | lmadsen | 2008-10-01 13:18:30 -0400 (Wed, 01 Oct 2008) | 6 lines Update the realtime_pgsql.sql script to create the setinterfacevar column. (closes issue #13549) Reported by: fiddur ........ 2008-10-01 15:44 +0000 [r145428] Tilghman Lesher * apps/app_sms.c: Initializing buffer prevents a segfault when arguments are incomplete. (closes issue #13471) Reported by: alecdavis Patches: 20080916__bug13471.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis 2008-10-01 14:44 +0000 [r145381] Mark Michelson * Makefile: Too many times have I mistyped the word 'install' as 'isntall' Now this typo shall no longer be a problem since 'make isntall' just builds the 'install' target. 2008-10-01 12:29 +0000 [r145329] Russell Bryant * CHANGES: tabs to spaces 2008-09-30 22:21 +0000 [r145249] Jeff Peeler * channels/chan_sip.c: (closes issue #13337) Reported by: pj Tested by: pj Set transport to SIP_TRANSPORT_UDP mode if not specified which fixes calls to get_transport returning UNKNOWN. 2008-09-30 21:32 +0000 [r145226] Russell Bryant * channels/chan_sip.c, CHANGES: Add support for call pickup on Snom phones. Asterisk now includes a magic call-id in the dialog-info event package used with extension state subscriptions on Snom phones. Then, when the phone sends an INVITE with Replaces for the special callid, Asterisk will perform a pickup on the extension that was subscribed to. The original code on this issue was submitted by xylome. However, contributions have been made by (at least) mgernoth and pkempgen. The final patch was written by seanbright, and includes the necessary logic to allow this work in a technology independent way. (closes issue #5014) Reported by: xylome Patches: issue5014-trunk.diff uploaded by seanbright (license 71) 2008-09-30 21:00 +0000 [r145200] Richard Mudgett * channels/misdn/isdn_lib.h, doc/tex/misdn.tex, channels/chan_misdn.c, channels/misdn/isdn_lib.c: * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP 2008-09-28 23:32 +0000 [r145121] Michiel van Baak * channels/chan_unistim.c, res/res_config_pgsql.c, apps/app_meetme.c, res/ais/clm.c, res/res_limit.c, main/taskprocessor.c, channels/chan_console.c, apps/app_queue.c, channels/chan_oss.c, main/astobj2.c, main/cli.c, channels/chan_dahdi.c, main/manager.c, channels/chan_misdn.c, channels/chan_features.c, res/res_agi.c, channels/chan_h323.c, res/ais/evt.c, res/res_config_ldap.c, apps/app_mixmonitor.c, res/res_clioriginate.c: Merge the cli_cleanup branch. This work is done by lmadsen, junky and mvanbaak during AstriDevCon. This is the second audit the CLI got, and this time lmadsen made sure he had _ALL_ modules loaded that have CLI commands in them. 2008-09-28 21:39 +0000 [r145076] Tilghman Lesher * res/res_config_pgsql.c: Change several improper "sizeof" to "strlen", as sizeof in that context would incorrectly use the size of a pointer, rather than the length of a string. (Closes issue #13574) 2008-09-28 17:08 +0000 [r145027] Kevin P. Fleming * channels/chan_dahdi.c: rename chandup() and clarify its usage 2008-09-27 16:17 +0000 [r144949-144951] Kevin P. Fleming * utils/Makefile: remove incorrect comment * agi/Makefile, utils/Makefile, include/asterisk/astmm.h: fix bugs caused by r144949 when MALLOC_DEBUG is defined * include/asterisk.h, /, main/Makefile, main/ast_expr2.y, Makefile.moddir_rules, utils/astman.c, main/ast_expr2.c, Makefile, utils/Makefile, main/ast_expr2f.c, pbx/pbx_ael.c, main/astmm.c, utils/ael_main.c, main/stdtime/localtime.c, utils/extconf.c, main/ast_expr2.fl: Merged revisions 144924-144925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) | 6 lines improve header inclusion process in a few small ways: - it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose - astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled - simplify the usage of some of these headers in the AEL-related stuff in the utils directory ........ r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines fix some minor issues with rev 144924 ........ 2008-09-27 00:49 +0000 [r144879] Michiel van Baak * channels/chan_dahdi.c, apps/app_queue.c: fix a couple of CLI commands that did not have a help description. 2008-09-26 23:12 +0000 [r144829] Joshua Colp * configs/rtp.conf.sample: Update documentation to include default setting. This is for you jtodd! 2008-09-26 18:02 +0000 [r144482-144681] Steve Murphy * pbx/pbx_lua.c: (closes issue #13564) Reported by: mnicholson Patches: pbx_lua9.diff uploaded by mnicholson (license 96) Many thanks to Matt for his upgrade to the lua dialplan option! the Description from the bug: This patch adds a stack trace to errors encountered while executing lua extensions. The patch also handles out of memory errors reported by lua. * main/pbx.c, /: Merged revisions 144677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) | 12 lines (closes issue #13563) Reported by: mnicholson Patches: found1.diff uploaded by mnicholson (license 96) This patch was mainly meant to apply to trunk and 1.6.x, but I'm applying it to 1.4 also, which should be a perfectly harmless fix to the vast majority of users who are not using external switches, but the few who might be affected will not have to go to the pain of filing a bug report. ........ * utils/build-extensions-conf.lua (removed): Matt suggests we remove utils/build-extensions-conf.lua, as per bug 12961, it is no longer necessary. * main/pbx.c, funcs/func_cut.c, channels/chan_oss.c, apps/app_playback.c: (closes issue #13557) Reported by: nickpeirson The user attached a patch, but the license is not yet recorded. I took the liberty of finding and replacing ALL index() calls with strchr() calls, and that involves more than just main/pbx.c; chan_oss, app_playback, func_cut also had calls to index(), and I changed them out. 1.4 had no references to index() at all. * pbx/pbx_lua.c: (closes issue #13559) Reported by: mnicholson Patches: pbx_lua8.diff uploaded by mnicholson (license 96) * pbx/pbx_lua.c, configs/extensions.lua.sample, include/asterisk/hashtab.h: I added a little verbage to hashtab for the hashtab_destroy func. It was pretty sparsely documented. This update fleshes out the pbx_lua module, to add the switch statements to the extensions in the extensions.lua file, as well as removing them when the module is unloaded. Many thanks to Matt Nicholson for his fine contribution! * pbx/pbx_lua.c: (closes issue #13558) Reported by: mnicholson Considering that the example extensions.lua used nothing but ["12345"] notation, and that the resulting error message: [Sep 24 17:01:16] ERROR[12393]: pbx_lua.c:1204 exec: Error executing lua extension: attempt to call a nil value is not very informative as to the nature of the problem, I think this bug fix is a big win! 2008-09-25 01:46 +0000 [r144357] Tilghman Lesher * /: Recorded merge of revisions 144356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144356 | tilghman | 2008-09-24 20:44:47 -0500 (Wed, 24 Sep 2008) | 6 lines Backport Hebrew language to voicemail. (closes issue #13155) Reported by: greenfieldtech Patches: voicemail-hebrew-patch-1.4-SVN.c.patch uploaded by greenfieldtech (license 369) ........ 2008-09-24 22:05 +0000 [r144314] Doug Bailey * res/res_phoneprov.c: Blanch the 404 error message for those with no sense of humor 2008-09-24 08:42 +0000 [r144257] Christian Richter * channels/chan_misdn.c, /: Merged revisions 144238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24 Sep 2008) | 1 line improved helptext of misdn_set_opt. ........ 2008-09-24 06:43 +0000 [r144199] Tilghman Lesher * funcs/func_curl.c: Create a 'hashcompat' option that permits the results of a CURL() able to be passed directly into the HASH() function. Requested via the -users list, and committed at Astricon in the Code Zone. 2008-09-23 23:33 +0000 [r144149] Mark Michelson * channels/chan_sip.c: Fix a conflict in flag values 2008-09-23 16:52 +0000 [r144067] Steve Murphy * /, main/features.c: Merged revisions 144066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) | 29 lines (closes issue #13489) Reported by: DougUDI Tested by: murf (closes issue #13490) Reported by: seanbright Tested by: murf (closes issue #13467) Reported by: edantie Tested by: murf, edantie, DougUDI This crash happens because we are unsafely handling old pointers. The channel whose cdr is being handled, has been hung up and destroyed already. I reorganized the code a bit, and tried not to lose the fork-cdr-chain concepts of the previous code. I now verify that the 'previous' channel (the channel we had when the bridge was started), still exists, by looking it up by name in the channel list. I also do not try to reset the CDR's of channels involved in bridges. Testing shows it solves the crash problem, and should not negatively impact previous fixes involving CDR's generated during/after blind transfers. (The reason we need to reset the CDR's on the "beginning" channels in the first place). ........ 2008-09-23 15:37 +0000 [r144025] Mark Michelson * channels/chan_sip.c: When a promiscuous redirect contained both a user and host portion in the Contact URI and specifies a transport, the parsing done in parse_moved_contact resulted in a malformed URI. This commit fixes the parsing so that a proper Dial string may be formed when the forwarded call is placed. (closes issue #13523) Reported by: mattdarnell Patches: 13523v2.patch uploaded by putnopvut (license 60) Tested by: mattdarnell 2008-09-22 22:50 +0000 [r143904] Sean Bright * /, formats/format_pcm.c: Merged revisions 143903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143903 | seanbright | 2008-09-22 18:49:00 -0400 (Mon, 22 Sep 2008) | 8 lines Use the advertised header size in .au files instead of just assuming they are 24 bytes (the minimum). (closes issue #13450) Reported by: jamessan Patches: pcm-header.diff uploaded by jamessan (license 246) ........ 2008-09-21 09:53 +0000 [r143799-143843] Michiel van Baak * doc/tex/privacy.tex: fix privacymanager example so it shows how to use the PRIVACYMRGSTATUS variable * doc/tex/privacy.tex: document the new context argument for privacymanager so people can do pattern matching on the input * doc/tex/privacy.tex: fix privacy documentation. We no longer do priority jumping +101 * channels/chan_skinny.c: make 'module unload chan_skinny.so' actually work. (closes issue #13524) Reported by: wedhorn Patches: unload.diff uploaded by wedhorn (license 30) With small tweak by me to prevent a crash 2008-09-20 00:52 +0000 [r143737] Sean Bright * /, contrib/scripts/vmail.cgi: Merged revisions 143736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143736 | seanbright | 2008-09-19 20:50:10 -0400 (Fri, 19 Sep 2008) | 9 lines Make vmail.cgi work with mailboxes defined in users.conf, too. (closes issue #13187) Reported by: netvoice Patches: 20080911__bug13187.diff.txt uploaded by Corydon76 (license 14) (Slightly modified to take alchamist's comments on mantis into account) Tested by: msales, alchamist, seanbright ........ 2008-09-19 21:41 +0000 [r143697] Steve Murphy * /: This blocks 143674 from trunk; it appears to already done in trunk, since ast_odbc_direct_execute creates a new stmt for each attempt. 2008-09-19 15:43 +0000 [r143609] Mark Michelson * channels/chan_agent.c: We should only unsubscribe to the device state event subscription if we have previously subscribed. Otherwise a segfault will occur. (closes issue #13476) Reported by: jonnt Patches: 13476.patch uploaded by putnopvut (license 60) Tested by: jonnt 2008-09-18 23:41 +0000 [r143559] Steve Murphy * /, channels/chan_sip.c: Merged revisions 143534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143534 | murf | 2008-09-18 16:11:51 -0600 (Thu, 18 Sep 2008) | 1 line A micro-fix, in sip_park_thread, where d is freed before the func is done using it. ........ 2008-09-17 20:57 +0000 [r143405] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 143404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143404 | tilghman | 2008-09-17 15:55:47 -0500 (Wed, 17 Sep 2008) | 6 lines When callerid is blank, we want to use "unknown caller" in those cases, too. (closes issue #13486) Reported by: tomo1657 Patches: 20080917__bug13486.diff.txt uploaded by Corydon76 (license 14) ........ 2008-09-17 20:25 +0000 [r143340-143400] Mark Michelson * main/astmm.c: If attempting to free a NULL pointer when MALLOC_DEBUG is set, don't bother searching for a region to free, just immediately exit. This has the dual benefit of suppressing a warning message about freeing memory at (nil) and of optimizing the free() replacement by not having to do any futile searching for the proper region to free. (closes issue #13498) Reported by: pj Patches: 13498.patch uploaded by putnopvut (license 60) Tested by: pj * /, main/rtp.c: Merged revisions 143337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep 2008) | 6 lines Allow for "G.729" if offered in an SDP even though it is not RFC 3551 compliant. Some Cisco switches will send this in an SDP, and it doesn't hurt to be able to accept this. ........ 2008-09-15 21:31 +0000 [r143141] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 143140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143140 | tilghman | 2008-09-15 16:29:32 -0500 (Mon, 15 Sep 2008) | 6 lines Set the raw formats at the same time as the other formats. (closes issue #13240) Reported by: jvandal Patches: 20080813__bug13240.diff.txt uploaded by Corydon76 (license 14) ........ 2008-09-14 22:16 +0000 [r143082] Michiel van Baak * channels/chan_skinny.c: plug a couple of memleaks in chan_skinny. (closes issue #13452) Reported by: pj Patches: memleak5.diff uploaded by wedhorn (license 30) Tested by: wedhorn, pj, mvanbaak (closes issue #13294) Reported by: pj 2008-09-13 14:15 +0000 [r143034] Sean Bright * apps/app_osplookup.c: Everytime a compile fails, a puppy dies. 2008-09-13 13:54 +0000 [r142992-143031] Tilghman Lesher * apps/app_dial.c, channels/chan_iax2.c, channels/iax2-parser.c: Repair IAXVAR implementation so that it works again (regression?) (closes issue #13354) Reported by: adomjan Patches: 20080828__bug13354.diff.txt uploaded by Corydon76 (license 14) 20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, adomjan * channels/chan_unistim.c, main/udptl.c, apps/app_meetme.c, res/res_snmp.c, codecs/codec_adpcm.c, res/res_phoneprov.c, codecs/codec_gsm.c, apps/app_alarmreceiver.c, channels/chan_gtalk.c, res/res_http_post.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c, res/res_jabber.c, main/enum.c, res/res_config_sqlite.c, main/config.c, main/loader.c, main/cdr.c, channels/chan_dahdi.c, channels/chan_phone.c, res/res_smdi.c, main/manager.c, funcs/func_config.c, apps/app_osplookup.c, channels/chan_skinny.c, funcs/func_odbc.c, main/features.c, apps/app_minivm.c, main/http.c, channels/chan_alsa.c, apps/app_amd.c, apps/app_directory.c, res/res_config_ldap.c, apps/app_rpt.c, channels/chan_mgcp.c, codecs/codec_lpc10.c, res/res_config_pgsql.c, main/dnsmgr.c, codecs/codec_g722.c, channels/chan_sip.c, apps/app_festival.c, codecs/codec_speex.c, codecs/codec_alaw.c, res/res_adsi.c, include/asterisk/config.h, channels/chan_agent.c, codecs/codec_g726.c, channels/chan_console.c, apps/app_queue.c, channels/chan_oss.c, main/rtp.c, apps/app_playback.c, channels/chan_jingle.c, channels/chan_h323.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c, res/res_indications.c, main/asterisk.c, res/res_odbc.c, main/dsp.c, apps/app_voicemail.c: Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) 2008-09-12 22:24 +0000 [r142929] Jeff Peeler * channels/chan_local.c, /: Merged revisions 142927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142927 | jpeeler | 2008-09-12 17:22:28 -0500 (Fri, 12 Sep 2008) | 6 lines (closes issue #12965) Reported by: rlsutton2 Prevents local channels from playing MOH at each other which was causing ast_generic_bridge to loop much faster. ........ 2008-09-12 20:49 +0000 [r142866] Tilghman Lesher * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 142865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines Create rules for disallowing contacts at certain addresses, which may improve the security of various installations. As this does not change any default behavior, it is not classified as a direct security fix for anything within Asterisk, but may help PBX admins better secure their SIP servers. (closes issue #11776) Reported by: ibc Patches: 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage ........ 2008-09-12 18:22 +0000 [r142808] Michiel van Baak * /: Recorded merge of revisions 142807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142807 | mvanbaak | 2008-09-12 19:59:25 +0200 (Fri, 12 Sep 2008) | 2 lines fix copyright year range ........ 2008-09-12 16:54 +0000 [r142741-142748] Tilghman Lesher * main/app.c: When checking for an encoded character, make sure the string isn't blank, first. (Closes issue #13470) * /, apps/app_voicemail.c: Merged revisions 142744 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142744 | tilghman | 2008-09-12 11:38:02 -0500 (Fri, 12 Sep 2008) | 4 lines Missing merge from 1.2 fixes errant exit on DTMF, only when language is Italian (cf commit 34242) (Closes issue #7353) ........ * /, main/file.c: Merged revisions 142740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142740 | tilghman | 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008) | 4 lines Don't return a free'd pointer, when a file cannot be opened. (closes issue #13462) Reported by: wackysalut ........ 2008-09-12 04:50 +0000 [r142676] Steve Murphy * apps/app_dial.c, main/pbx.c, /, main/features.c, include/asterisk/channel.h, apps/app_queue.c: Merged revisions 142675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ 2008-09-12 00:49 +0000 [r142635] Sean Bright * cdr/cdr_adaptive_odbc.c: Build under dev-mode 2008-09-11 23:12 +0000 [r142576] Steve Murphy * /, main/features.c: Merged revisions 142575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) | 20 lines (closes issue #13364) Reported by: mdu113 Well, fundamentally, the problems revealed in 13364 are because of the ForkCDR call that is done before the dial. When the bridge is in place, it's dealing with the first (and wrong) cdr in the list. So, I wrote a little func to zip down to the first non-locked cdr in the chain, and thru-out the ast_bridge_call, these results are used instead of raw chan->cdr and peer->cdr pointers. This shouldn't affect anyone who isn't forking cdrs before a dial, and should correct the cdr's of those that do. So, this change ends up correcting the dstchannel and userfield; the disposition was fixed by a previous patch, it was OK coming into this problem. ........ 2008-09-11 21:45 +0000 [r142536] Tilghman Lesher * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample: Add usegmtime, as per the recent -users list discussion, and also add my explanation to the file, since that additional text helps people understand the concept. 2008-09-10 22:11 +0000 [r142475] Steve Murphy * /, main/features.c: Merged revisions 142474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) | 30 lines (closes issue #12318) Reported by: krtorio I made a small change to the code that handles local channel situations. In that code, I copy the answer time from the peer cdr, to the bridge_cdr, but I wasn't also copying the disposition from the peer cdr. So, Now I copy the disposition, and I've tested against these cases: 1. phone 1 never answers the phone; no cdr is generated at all. this should show up as a manager command failure or something. 2. phone 2 never answers. CDR is generated, says NO ANSWER 3. phone 2 is busy. CDR is generated, says BUSY 4. phone 2 answers: CDR is generated, times are correct; disposition is ANSWERED, which is correct. The start time is the time that the manager dialed the first phone. The answer time is the time the second phone picks up. I purposely left the cid and src fields blank; since this call really originates from the manager, there is no 'easy' data to put in these fields. If you feel strongly that these fields should be filled in, re-open this bug and I'll dig further. ........ 2008-09-10 19:09 +0000 [r142417] Sean Bright * /, configure, acinclude.m4: Merged revisions 142416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142416 | seanbright | 2008-09-10 15:05:46 -0400 (Wed, 10 Sep 2008) | 9 lines Fix detection of PWLIB and OpenH323 version when spacing in the headers isn't consistent. (closes issue #13426) Reported by: bamby Patches: detect_openh323.diff uploaded by bamby (license 430) (Modified by me to use sed instead of tr) ........ 2008-09-10 16:55 +0000 [r142359] Tilghman Lesher * /, sounds/Makefile: Merged revisions 142358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142358 | tilghman | 2008-09-10 11:54:29 -0500 (Wed, 10 Sep 2008) | 2 lines Publish new extra sounds version. ........ 2008-09-10 16:41 +0000 [r142318-142355] Russell Bryant * /, main/sched.c: Merged revisions 142354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008) | 7 lines It is a normal situation that a task gets put in the scheduler that should run as soon as possible. Accept "0" as an acceptable time to run, and also treat negative as "run now", and don't print a debug message about it. (inspired by a message asking about the "request to schedule in the past" debug message on the -dev list) ........ * CHANGES: Move last change to CHANGES up to the 1.6.2 section 2008-09-09 22:08 +0000 [r142280] Philippe Sultan * configs/jabber.conf.sample, CHANGES, res/res_jabber.c: Disable autoprune by default. (closes issue #13411) Reported by: caio1982 Patches: res_jabber_autoprune1.diff uploaded by caio1982 (license 22) Tested by: caio1982 2008-09-09 19:16 +0000 [r142219] Mark Michelson * /, channels/chan_sip.c: Merged revisions 142218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142218 | mmichelson | 2008-09-09 14:15:28 -0500 (Tue, 09 Sep 2008) | 14 lines Make sure that the branch sent in CANCEL requests matches the branch of the INVITE it is cancelling. (closes issue #13381) Reported by: atca_pres Patches: 13381v2.patch uploaded by putnopvut (license 60) Tested by: atca_pres (closes issue #13198) Reported by: rickead2000 Tested by: rickead2000 ........ 2008-09-09 17:30 +0000 [r142181] Richard Mudgett * main/callerid.c: Cleaned up comment 2008-09-09 17:15 +0000 [r142080-142146] Mark Michelson * apps/app_queue.c: This is the trunk version of the patch to close issue 12979. The difference between this and the 1.6.0 and 1.6.1 versions is that this is a much more invasive change. With this, we completely get rid of the interfaces list, along with all its helper functions. Let me take a moment to say that this change personally excites me since it may mean huge steps forward regarding proper lock order in app_queue without having to strew seemingly unnecessary locks all over the place. It also results in a huge reduction in lines of code and complexity. Way to go Brett! (closes issue #12979) Reported by: sigxcpu Patches: 20080710_issue12979_queue_custom_state_interface_trunk_2.diff uploaded by bbryant (license 36) Tested by: sigxcpu, putnopvut * /, channels/chan_sip.c: Merged revisions 142079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep 2008) | 21 lines When determining if codecs used by SIP peers allow the media to be natively bridged, use the jointcapability instead of the peercapability. It seems that the intent of using the peercapability was to expand the choice of codecs for the call to increase the chances of being able to native bridge the channels. The problem is that if a codec were settled on for the native bridge and that wasn't a codec that was configured to be used by Asterisk for that peer, then Asterisk would send a REINVITE with no codecs in the SDP which is a bug no matter how you slice it. (closes issue #13076) Reported by: ramonpeek Patches: 13076.patch uploaded by putnopvut (license 60) Tested by: tbelder ........ 2008-09-09 15:44 +0000 [r142064] Russell Bryant * /, main/features.c: Merged revisions 142063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) | 5 lines Ensure that the stored CDR reference is still valid after the bridge before poking at it. Also, keep the channel locked while messing with this CDR. (fixes crashes reported in issue #13409) ........ 2008-09-09 12:34 +0000 [r142000] Bradley Latus * include/asterisk/astobj2.h: Minor fix to doco 2008-09-09 12:32 +0000 [r141995-141998] Mark Michelson * apps/app_queue.c: Use ast_debug for debug messages. I was wondering why debug messages weren't showing up when I had set the debug level high for just app_queue.c. It's because we were only checking the global option_debug variable instead of using the awesome macro which checks both the global and file-specific value * channels/chan_oss.c: Fix a memory leak in chan_oss (closes issue #13311) Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel (license 64) 2008-09-09 01:47 +0000 [r141949] Russell Bryant * main/channel.c: Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) 2008-09-08 23:00 +0000 [r141810-141906] Mark Michelson * apps/app_queue.c: Optimization: The only reason we should check member status is if the queue has a joinempty or a leavewhenempty setting which could cause the caller to not join the queue or exit the queue. Prior to this patch, we could potentially traverse the entire queue's member list for no reason since even if the members are currently not available in some way we're going to let the caller join the queue anyway. * channels/chan_sip.c: Um, apparently I didn't actually finish merging before committing. Bad bad bad * /, channels/chan_sip.c: Merged revisions 141809 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep 2008) | 14 lines Fix pedantic mode of chan_sip to only check the remote tag of an endpoint once a dialog has been confirmed. Up until that point, it is possible and legal for the far-end to send provisional responses with a different To: tag each time. With this patch applied, these provisional messages will not cause a matching problem. (closes issue #11536) Reported by: ibc Patches: 11536v2.patch uploaded by putnopvut (license 60) ........ 2008-09-08 21:05 +0000 [r141807] Russell Bryant * main/pbx.c, /: Merged revisions 141806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) | 7 lines When doing an async goto, detect if the channel is already in the middle of a masquerade. This can happen when chan_local is trying to optimize itself out. If this happens, fail the async goto instead of bursting into flames. (closes issue #13435) Reported by: geoff2010 ........ 2008-09-08 20:18 +0000 [r141745] Jason Parker * Makefile, /, redhat (removed): Merged revisions 141741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) | 8 lines Remove RPM package targets from Makefile (and all associated parts). This has never worked in 1.4, and we decided that it makes no sense to be done here. There are many distros out there that already have "proper" spec files that can be (re)used. Closes issue #13113 Closes issue #10950 Closes issue #10952 ........ 2008-09-08 17:13 +0000 [r141682] Sean Bright * build_tools/make_buildopts_h: Quote the arguments to grep so that sh on various platforms doesn't choke on the special characters (like ^). (closes issue #13417) Reported by: dougm Patches: 13417.make_buildopts_h.patch uploaded by seanbright (license 71) Tested by: dougm 2008-09-07 00:04 +0000 [r141626] Michiel van Baak * funcs/func_curl.c: make func_curl.c compile under devmode. 2008-09-06 20:19 +0000 [r141566] Steve Murphy * /, channels/chan_sip.c: Merged revisions 141565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs. ........ 2008-09-06 15:40 +0000 [r141504-141507] Tilghman Lesher * funcs/func_curl.c: Get rid of the casts that cause warnings on OpenBSD. The compiler is errantly detecting warnings when we redefine a structure each time it is used, even though the structure is identical. Reported by: mvanbaak, via #asterisk-dev * /, res/res_agi.c: Merged revisions 141503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008) | 4 lines Reverting behavior change (AGI should not exit non-zero on SUCCESS) (closes issue #13434) Reported by: francesco_r ........ 2008-09-06 12:03 +0000 [r141464] Michiel van Baak * channels/chan_sip.c, channels/chan_iax2.c, main/cli.c: Some fixes to autocompletion in some commands. Changes applied by this patch: - Fix autocompletion in 'sip prune realtime', sip peers where never auto completed. Now we complete this command with: 'sip prune realtime peer' -> all | like | sip peers Also I have modified the syntax in the usage, was wrong... - Pass ast_cli_args->argv and ast_cli_args->argc while running autocompletion on CLI commands (CLI_GENERATE). With this we avoid comparisons on ast_cli_args->line like this: strcasestr(a->line, " description") strcasestr(a->line, "descriptions ") strcasestr(a->line, "realtime peer"), and so on.. Making the code more confusing (check the spaces in description!). The only thing we must be sure is to first check a->pos or a->argc. - Fix 'iax2 prune realtime' autocompletion, now we autocomplete this command with 'all' & 'iax2 peers', check a look that iax2 peers where all the peers, now only the ones in the cache.. (closes issue #13133) Reported by: eliel Patches: clichanges.patch uploaded by eliel (license 64) 2008-09-05 22:03 +0000 [r141367-141425] Mark Michelson * funcs/func_curl.c: Fix func_curl compilation * /, channels/chan_agent.c: Merged revisions 141366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep 2008) | 7 lines Agent's should not try to call a channel's indicate callback if the channel has been hung up. It will likely crash otherwise ABE-1159 ........ 2008-09-05 19:12 +0000 [r141328] Tilghman Lesher * funcs/func_curl.c, CHANGES: Add the CURLOPT dialplan function, which permits setting various options for use with the CURL dialplan function. (closes issue #12920) Reported by: davevg Patches: 20080904__bug12920.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, davevg 2008-09-05 14:18 +0000 [r141115-141157] Steve Murphy * main/channel.c, /: Merged revisions 141156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 line A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention ........ * pbx/ael/ael-test/ref.ael-vtest25 (added), /, pbx/ael/ael-test/ael-vtest25/extensions.ael, pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged revisions 141094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | 70 lines (closes issue #13357) Reported by: pj Tested by: murf (closes issue #13416) Reported by: yarns Tested by: murf If you find this message overly verbose, relax, it's probably not meant for you. This message is meant for probably only two people in the whole world: me, or the poor schnook that has to maintain this code because I'm either dead or unavailable at the moment. This fix solves two reports, both having to do with embedding a function call in a ${} construct. It was tricky because the funccall syntax has parenthesis () in it. And up till now, the 'word' token in the flex stuff didn't allow that, because it would tend to steal the LP and RP tokens. To be truthful, the "word" token was the trickiest, most unstable thing in the whole lexer. I was lucky it made this long without complaints. I had to choose every character in the pattern with extreme care, and I knew that someday I'd have to revisit it. Well, the day has come. So, my brilliant idea (and I'm being modest), was to use the surrounding ${} construct to make a state machine and capture everything in it, no matter what it contains. But, I have to now treat the word token like I did with comments, in that I turn the whole thing into a state-machine sort of spec, with new contexts "curlystate", "wordstate", and "brackstate". Wait a minute, "brackstate"? Yes, well, it didn't take very many regression tests to point out if I do this for ${} constructs, I also have to do it with the $[] constructs, too. I had to create a separate pcbstack2 and pcbstack3 because these constructs can occur inside macro argument lists, and when we have two state machines operating on the same structures we'd get problems otherwise. I guess I could have stopped at pcbstack2 and had the brackstate stuff share it, but it doesn't hurt to be safe. So, the pcbpush and pcbpop routines also now have versions for "2" and "3". I had to add the {KEYWORD} construct to the initial pattern for "word", because previously word would match stuff like "default7", because it was a longer match than the keyword "default". But, not any more, because the word pattern only matches only one or two characters now, and it will always lose. So, I made it the winner again by making an optional match on any of the keywords before it's normal pattern. I added another regression test to make sure we don't lose this in future edits, and had to fix just one regression, where it no longer reports a 'cascaded' error, which I guess is a plus. I've given some thought as to whether to apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I decided to put it in 1.4 because one of the bug reports was against 1.4; and it is unexpected that AEL cannot handle this situation. It actually reduced the amount of useless "cascade" error messages that appeared in the regressions (by one line, ehhem). There is a possible side-effect in that it does now do more careful checking of what's in those ${} constructs, as far as matching parens, and brackets are concerned. Some users may find a an insidious problem and correct it this way. This should be exceedingly rare, I hope. ........ 2008-09-04 17:27 +0000 [r141039] Jeff Peeler * /, main/features.c, res/res_agi.c: Merged revisions 141028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) | 7 lines (closes issue #11979) Fixes multiple parking problems: Crash when executing a park on an extension dialed by AGI due to not returning the proper return code. Crash when using a builtin feature that was a subset of a enabled dynamic feature. Crash due to always hanging up the peer despite the fact that the peer was supposed to be parked. ........ 2008-09-03 20:16 +0000 [r140975] Mark Michelson * apps/app_queue.c: Fix some locking order issues in app_queue. This was brought up by atis on IRC a while ago. 2008-09-03 18:06 +0000 [r140938] Michiel van Baak * channels/chan_skinny.c, CHANGES: Added 'skinny show lines verbose' This will print the subs and their status for every line (if any). wedhorn did most of the work with his patch which introduced 'skinny show debug' but a discussion on IRC stated that it should be added to 'skinny show lines' Input on the output format by Qwell on IRC. (closes issue #13344) Reported by: wedhorn 2008-09-03 14:41 +0000 [r140860-140887] Mark Michelson * apps/app_voicemail.c: Fix compilation * /, apps/app_voicemail.c: Merged revisions 140850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140850 | mmichelson | 2008-09-03 09:29:15 -0500 (Wed, 03 Sep 2008) | 9 lines Fix voicemail forwarding when using ODBC storage. (closes issue #13387) Reported by: moliveras Patches: 13387.patch uploaded by putnopvut (license 60) Tested by: putnopvut, moliveras ........ 2008-09-03 14:01 +0000 [r140824] Steve Murphy * res/ael/pval.c, main/pbx.c, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h: In these changes, I have added some explanation of changes to the Set and MSet apps, so people aren't so shocked and surprised when they upgrade from 1.4 to 1.6. Also, for the sake of those upgrading from 1.4 to 1.6 with AEL, I provide automatic support for the "old" way of using Set(), that still does the exact same old thing with quotes and backslashes and so on as 1.4 did, by having AEL compile in the use of MSet() instead of Set(), everywhere it inserts this code. But, if the app_set var is set to 1.6 or higher, it uses the "new", non-evaluative Set(). This only usually happens if the user manually inserts this into the asterisk.conf file, or runs the "make samples" command. 2008-09-03 13:48 +0000 [r140821] Sean Bright * cdr/cdr_sqlite.c: Move some duplicated code into a separate function. Also try to do some wacky stuff in the commit message, like: a newline \n a bell \a a tab \t a format specification %p That is all. 2008-09-03 13:41 +0000 [r140817-140820] Russell Bryant * main/pbx.c: Formatting change to test something on the svn server * /, main/poll.c: Merged revisions 140816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008) | 4 lines Don't freak out if the poll emulation receives NULL for the pollfds array (closes issue #13307) Reported by: jcovert ........ 2008-09-02 23:48 +0000 [r140752] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 140751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140751 | mmichelson | 2008-09-02 18:47:49 -0500 (Tue, 02 Sep 2008) | 6 lines After adding the context checking to app_voicemail for IMAP storage, I left out a crucial place to copy the context to the vm_state structure. This is the correction. ........ 2008-09-02 23:44 +0000 [r140691-140749] Steve Murphy * main/cdr.c, /: Merged revisions 140747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 line I am turning the warnings generated in ast_cdr_free and post_cdr into verbose level 2 messages. Really, they matter little to end users. You either get the CDR's you wanted, or you don't, and it is a bug. For trunk, I am going one step further. These messages were pretty worthless even for debug, so I'm completely removing them. ........ * main/channel.c, /: Merged revisions 140690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints. Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations, where you'd want to post single-channel cdrs. ........ * main/channel.c, main/pbx.c, /: Merged revisions 140670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines (closes issue #13409) Reported by: tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564) I basically spent the day, verifying that this patch solves the problem, and doesn't hurt in non-problem cases. Why valgrind did not plainly reveal this leak absolutely mystifies and stuns me. Many, many thanks to tomaso for finding and providing the fix. ........ 2008-09-02 18:15 +0000 [r140606] Sean Bright * /, channels/chan_iax2.c: Merged revisions 140605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep 2008) | 8 lines Make sure to use the correct length of the mohinterpret and mohsuggest buffers when copying configuration values. (closes issue #13336) Reported by: decryptus_proformatique Patches: chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded by decryptus (license 555) ........ 2008-09-02 15:11 +0000 [r140563-140566] Russell Bryant * codecs/codec_resample.c, apps/app_jack.c: Update instructions for getting libresample * res/ais/lck.c (removed), res/ais/ckpt.c (removed), res/ais/amf.c (removed): I'm not sure how these files got to trunk (probably my fault), but they should not be here 2008-09-02 14:41 +0000 [r140559] Sean Bright * channels/chan_sip.c: When a call is rejected because of call-limit, the channel driver is behaving as expected, so we shouldn't report it as an error. Change to LOG_NOTICE instead. 2008-08-29 17:53 +0000 [r140491] Jeff Peeler * main/features.c, CHANGES: Added the option s to the Park application which will silence the announcement of the parking space number. Also, fixes the bug of just clearing the flags instead of actually parsing the arguments to Park. 2008-08-29 17:47 +0000 [r140418-140489] Mark Michelson * main/manager.c, res/ais/lck.c, /, channels/chan_sip.c, funcs/func_dialgroup.c, res/res_timing_pthread.c, main/features.c, res/res_phoneprov.c, utils/hashtest2.c, channels/chan_console.c, main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c, main/config.c: Merged revisions 140488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug 2008) | 22 lines After working on the ao2_containers branch, I noticed something a bit strange. In all cases where we provide a callback function to ao2_container_alloc, the callback function would only return 0 or CMP_MATCH. After inspecting the ao2_callback() code carefully, I found that if you're only looking for one specific item, then you should return CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue traversing the current bucket until the end searching for more matches. In cases like chan_iax2 where in 1.4, all the peers are shoved into a single bucket, this makes for potentially terrible performance since the entire bucket will be traversed even if the peer is one of the first ones come across in the bucket. All the changes I have made were for cases where the callback function defined was passed to ao2_container_alloc so that calls to ao2_find could find a unique instance of whatever object was being stored in the container. ........ * main/file.c: Allow for video files to be opened as well as audio files. (closes issue #13372) Reported by: epicac Patches: 13372.patch uploaded by putnopvut (license 60) Tested by: epicac * /, apps/app_voicemail.c: Merged revisions 140421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140421 | mmichelson | 2008-08-29 11:01:07 -0500 (Fri, 29 Aug 2008) | 12 lines Add context checking when retrieving a vm_state. This was causing a problem for people who had identically named mailboxes in separate voicemail contexts. This commit affects IMAP storage only. (closes issue #13194) Reported by: moliveras Patches: 13194.patch uploaded by putnopvut (license 60) Tested by: putnopvut, moliveras ........ * channels/chan_sip.c: Merged revisions 140417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140417 | mmichelson | 2008-08-29 10:26:52 -0500 (Fri, 29 Aug 2008) | 10 lines Fix SIP's parsing so that if a port is specified in a string to Dial(), it is not ignored. (closes issue #13355) Reported by: acunningham Patches: 13355v2.patch uploaded by putnopvut (license 60) Tested by: acunningham ........ 2008-08-27 23:23 +0000 [r140355] Tilghman Lesher * cdr/cdr_pgsql.c: Oops 2008-08-27 20:11 +0000 [r140301] Mark Michelson * channels/chan_sip.c: Merged revisions 140299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when in pedantic mode. The problem was that the wrong tags would be compared depending on the direction of the call. (closes issue #13353) Reported by: flefoll Patches: chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244) ........ 2008-08-26 21:59 +0000 [r140246] Doug Bailey * channels/chan_dahdi.c: Move the mwi send thread functionality back into the do_monitor thread so that it is easier to manage CID spill resources when do_monitor needs to be killed. (closes issue #13213) Reported by: bbryant 2008-08-26 18:48 +0000 [r140205] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 140056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26 Aug 2008) | 9 lines (closes issue #12071) Reported by: tzafrir Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested by: tzafrir, jpeeler This patch fixes closing open file descriptors in the case of an error. ........ 2008-08-26 18:46 +0000 [r140201] Tilghman Lesher * apps/app_followme.c: OpenBSD compat fix (reminded by mvanbaak on #asterisk-dev) 2008-08-26 18:11 +0000 [r140169] Russell Bryant * Makefile: Fix building menuselect-tree with PRINT_DIR set. We _must_ use the --quiet flag here, or else some arbitrary text will end up in the resulting menuselect-tree file and things will explode. 2008-08-26 18:05 +0000 [r140167] Tilghman Lesher * configs/followme.conf.sample, apps/app_followme.c: Standardize the option names for consistency (but continue to work with the existing names for backwards compatibility). (closes issue #13370) Reported by: jsturtevant 2008-08-26 16:10 +0000 [r140061] Russell Bryant * /, channels/chan_sip.c: Merged revisions 140060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140060 | russell | 2008-08-26 11:07:58 -0500 (Tue, 26 Aug 2008) | 6 lines Fix some bogus scheduler usage in chan_sip. This code used the return value of a completely unrelated function to determine whether the scheduler should be run or not. This would have caused the scheduler to not run in cases where it should have. Also, leave a note about another scheduler issue that needs to be addressed at some point. ........ 2008-08-26 15:57 +0000 [r140057] Steve Murphy * main/cdr.c, configs/cdr.conf.sample, CHANGES, include/asterisk/options.h: (closes issue #13366) Reported by: erousseau This was a reasonable enhancement request, which was easy to implement. Since it's an enhancement, it could only be applied to trunk. Basically, for accounting where "initiated" seconds are billed for, if the microseconds field on the end time is greater than the microseconds field for the answer time, add one second to the billsec field. The implementation was requested by erousseau, and I've implemented it as requested. I've updated the CHANGES, the cdr.conf.sample, and the .h files accordingly, to accept and set a flag for the corresponding new option. cdr.c adds in the extra second based on the usec fields if the option is set. Tested, seems to be working fine. 2008-08-26 15:29 +0000 [r140053] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 140051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140051 | russell | 2008-08-26 10:27:23 -0500 (Tue, 26 Aug 2008) | 15 lines Fix a race condition with the IAX scheduler thread. A lock and condition are used here to allow newly scheduled tasks to wake up the scheduler just in case the new task needs to run sooner than the current wakeup time when the thread is sleeping. However, there was a race condition such that a newly scheduled task would not properly wake up the scheduler or affect the wake up period. The order of execution would have been: 1) Scheduler thread determines wake up time of N ms. 2) Another thread schedules a task and signals the condition, with an execution time of < N ms. 3) Scheduler thread locks and goes to sleep for N ms. By moving the sleep time determination to inside the critical section, this possibility is avoided. ........ 2008-08-25 23:13 +0000 [r139981] Tilghman Lesher * Makefile, doc/asterisk.8, include/asterisk/options.h, main/asterisk.c, main/term.c: Optional light colored background, for those who use black on white terminals. (closes issue #13306) Reported by: Corydon76 Patches: 20080814__bug13306__3.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, pkempgen 2008-08-25 21:48 +0000 [r139928] Jeff Peeler * main/manager.c, /: Merged revisions 139927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139927 | jpeeler | 2008-08-25 16:47:33 -0500 (Mon, 25 Aug 2008) | 3 lines Fix a typo I made. Lesson learned, apply the patch if one exists. ........ 2008-08-25 21:32 +0000 [r139915] Sean Bright * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged revisions 139909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug 2008) | 9 lines Some versions of awk (nawk, for example) don't like empty regular expressions so be slightly more verbose. (closes issue #13374) Reported by: dougm Patches: 13374.diff uploaded by seanbright (license 71) Tested by: dougm ........ 2008-08-25 20:59 +0000 [r139870] Terry Wilson * /, channels/chan_sip.c: Merged revisions 139869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008) | 2 lines Make SIPADDHEADER() propagate indefinitely ........ 2008-08-25 17:24 +0000 [r139832] Mark Michelson * apps/app_queue.c: Add output of variables to AgentRingNoAnswer manager event if eventwhencalled is set to "vars" in queues.conf. Yay for consistency. (closes issue #13369) Reported by: srt Patches: 13369_agentringnoanswer_variables.diff uploaded by srt (license 378) 2008-08-25 16:02 +0000 [r139775] Tilghman Lesher * doc/followme.txt (added), apps/app_followme.c: Realtime capabilities for the Find-Me-Follow-Me application. (closes issue #13295) Reported by: Corydon76 Patches: 20080813__followme_realtime_enabled.diff.txt uploaded by Corydon76 (license 14) Tested by: dferrer 2008-08-25 15:54 +0000 [r139770] Steve Murphy * main/pbx.c, /, main/features.c: Merged revisions 139764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 lines This patch reverts the changes made via 139347, and 139635, as users are seeing adverse difference. I will un-close 13251. Back to the drawing board/ concept/ beginning/ whatever! ........ 2008-08-24 16:26 +0000 [r139704-139707] Tilghman Lesher * cdr/cdr_pgsql.c: Memory leak * cdr/cdr_pgsql.c: Eliminate open coding of ast_str 2008-08-22 22:32 +0000 [r139627-139662] Steve Murphy * /, main/features.c: Merged revisions 139635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 lines I found some problems with the code I committed earlier, when I merged them into trunk, so I'm coming back to clean up. And, in the process, I found an error in the code I added to trunk and 1.6.x, that I'll fix using this patch also. ........ * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions 139347 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines (closes issue #13251) Reported by: sergee Tested by: murf THis is a bold move for a static release fix, but I wouldn't have made it if I didn't feel confident (at least a *bit* confident) that it wouldn't mess everyone up. The reasoning goes something like this: 1. We simply cannot do anything with CDR's at the current point (in pbx.c, after the __ast_pbx_run loop). It's way too late to have any affect on the CDRs. The CDR is already posted and gone, and the remnants have been cleared. 2. I was very much afraid that moving the running of the 'h' extension down into the bridge code (where it would be now practical to do it), would result in a lot more calls to the 'h' exten, so I implemented it as another exten under another name, but found, to my pleasant surprise, that there was a 1:1 correspondence to the running of the 'h' exten in the pbx_run loop, and the new spot at the end of the bridge. So, I ifdef'd out the current 'h' loop, and moved it into the bridge code. The only difference I can see is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this is still an important decision point, I can replicate it if there are complaints. To be perfectly honest, the KEEPALIVE situation is not totally clear to me, and how it relates to a post-bridge situation is less clear. I suspect the users will point out everything in total clarity if this steps on anyone's toes! 3. I temporarily swap the bridge_cdr into the channel before running the 'h' exten, which makes it possible for users to edit the cdr before it goes out the door. And, of course, with the endbeforehexten config var set, the users can also get at the billsec/duration vals. After the h exten finishes, the cdr is swapped back and processing continues as normal. Please, all who deal with CDR's, please test this version of Asterisk, and file bug reports as appropriate! ........ I also made a little fix to the app_dial's 'e' option, that is related to my updates. 2008-08-22 21:57 +0000 [r139622-139624] Jeff Peeler * main/manager.c, /: Merged revisions 139621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008) | 5 lines (closes issue #13359) Reported by: Laureano Patches: originate_channel_check.patch uploaded by Laureano (license 265) ........ * main/features.c: remove extra comma typo 2008-08-22 20:20 +0000 [r139457-139563] Mark Michelson * channels/chan_sip.c: The -1 return value from incomplete or improper headers for the SipNotify manager command was causing the current manager session to become disconnected. Change the return value to 0 for these cases. Also change a test for a NULL pointer to be ast_strlen_zero instead. (closes issue #13351) Reported by: Laureano Patches: sipnotify_action_fix.patch uploaded by Laureano (license 265) * main/features.c: Add missing unique id to ParkedCallGiveUp and ParkedCallTimeOut manager events (closes issue #13358) Reported by: srt Patches: 13358_parking_events.diff uploaded by srt (license 378) * /, include/asterisk/threadstorage.h: Merged revisions 139553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy (license 35) ........ * /, channels/chan_iax2.c: Merged revisions 139466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........ * /, channels/chan_iax2.c: Merged revisions 139456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from incorrect locking order between iax2_pvt and ast_channel structures. AST-13 ........ 2008-08-21 23:41 +0000 [r139391] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 139387 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008) | 3 lines Fixes loop that could possibly never exit in the event of a channel never being able to be opened or specify after a restart. (closes issue #11017) ........ 2008-08-21 23:00 +0000 [r139345-139346] Dwayne M. Hubbard * apps/app_receivefax.c (removed), apps/app_sendfax.c (removed): oops * apps/app_receivefax.c (added), apps/app_sendfax.c (added): initiate T38 negotiation in FaxSend; use channel variables; other stuff too 2008-08-21 09:55 +0000 [r139281] Philippe Sultan * channels/chan_gtalk.c: Fix two memory leaks in chan_gtalk, thanks Eliel! (closes issue #13310) Reported by: eliel Patches: chan_gtalk.c.patch uploaded by eliel (license 64) 2008-08-20 22:16 +0000 [r139215] Russell Bryant * /, apps/app_chanspy.c: Merged revisions 139213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) | 11 lines Fix a crash in the ChanSpy application. The issue here is that if you call ChanSpy and specify a spy group, and sit in the application long enough looping through the channel list, you will eventually run out of stack space and the application with exit with a seg fault. The backtrace was always inside of a harmless snprintf() call, so it was tricky to track down. However, it turned out that the call to snprintf() was just the biggest stack consumer in this code path, so it would always be the first one to hit the boundary. (closes issue #13338) Reported by: ruddy ........ 2008-08-20 22:06 +0000 [r139210] Jason Parker * channels/chan_sip.c: Fix output of sipshowpeer manager response. (closes issue #13346) Reported by: srt Patches: 13346_malformed_sip_show_peer_response.diff uploaded by srt (license 378) 2008-08-20 20:03 +0000 [r139153-139154] Shaun Ruffell * codecs/codec_dahdi.c: Remove extraneous debugging messages. * codecs/codec_dahdi.c: Fix bug where the samples were not accurate when in G723 mode, which would cause the timestamp field of the RTP header to be invalid. 2008-08-20 17:25 +0000 [r139083] Steve Murphy * main/cdr.c, /: Merged revisions 139074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | 12 lines (closes issue #13263) Reported by: brainy Tested by: murf The specialized reset routine is tromping on the flags field of the CDR. I made a change to not reset the DISABLED bit. This should get rid of this problem. ........ 2008-08-20 16:16 +0000 [r139020] Michiel van Baak * channels/chan_skinny.c: fix unholding phones after hangup on older cisco phones. Patch by wedhorn. 2008-08-20 15:38 +0000 [r138887-139016] Mark Michelson * /, channels/chan_sip.c: Merged revisions 139015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines sip_read should properly handle a NULL return from sip_rtp_read. (closes issue #13257) Reported by: travishein ........ * /, channels/chan_agent.c: Merged revisions 138942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138942 | mmichelson | 2008-08-19 18:17:17 -0500 (Tue, 19 Aug 2008) | 11 lines Reset agent_pvt variables back to the values in agents.conf (from what the corresponding channel variables were set to) when the agent logs out. (closes issue #13098) Reported by: davidw Patches: 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by bbryant (license 36) Tested by: davidw ........ * /, apps/app_chanspy.c: Merged revisions 138886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug 2008) | 23 lines Add a lock and unlock prior to the destruction of the chanspy_ds lock to ensure that no other threads still have it locked. While this should not happen under normal circumstances, it appears that if the spyer and spyee hang up at nearly the same time, the following may occur. 1. ast_channel_free is called on the spyee's channel. 2. The chanspy datastore is removed from the spyee's channel in ast_channel_free. 3. In the spyer's thread, the spyer attempts to remove and destroy the datastore from the spyee channel, but the datastore has already been removed in step 2, so the spyer continues in the code. 4. The spyee's thread continues and calls the datastore's destroy callback, chanspy_ds_destroy. This involves locking the chanspy_ds. 5. Now the spyer attempts to destroy the chanspy_ds lock. The problem is that in step 4, the spyee has locked this lock, meaning that the spyer is attempting to destroy a lock which is currently locked by another thread. The backtrace provided in issue #12969 supports the idea that this is possible (and has even occurred). This commit does not close the issue, but should help in preventing one type of crash associated with the use of app_chanspy. ........ 2008-08-19 16:56 +0000 [r138851] Michiel van Baak * channels/chan_skinny.c: chan_skinny now respects callwaiting=no (closes issue #12691) Reported by: sbisker Patches: callwaitingv1.diff uploaded by wedhorn (license 30) Tested by: wedhorn on old skinny phones, mvanbaak on 7960 and 7905 with latest firmware 2008-08-19 16:31 +0000 [r138815-138845] Steve Murphy * res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h, utils/ael_main.c, utils/conf2ael.c: Oops. put a decl in a generated file. My bad, but fixed now. * main/pbx.c, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h: These changes are in regards to bug 13249, where users are being surprised by the changes made to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x installation where a "make samples" was executed, or where they hand-edited the asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher). (this commit does not totally solve 13249, at least not yet) The change involves issueing a single warning while the AEL file is loading, if: 1. app_set is present in the config file, and set to 1.6 or higher. 2. there are double quotes in an assignment statement (eg x = "hi there";) 3. the warning was not already issued. The standalone app, aelparse, does not (yet) issue this warning. I'd have to have it read in the asterisk.conf file, and that's a bit of hassle. I'll add it if users request it, tho. 2008-08-19 15:58 +0000 [r138814] Philippe Sultan * res/res_jabber.c: Mention JID rather than SreenName in help messages 2008-08-19 00:10 +0000 [r138775-138780] Sean Bright * channels/chan_sip.c: Let it compile now, too (woops) * channels/chan_sip.c: And remove code we don't need anymore. * channels/chan_sip.c: While we're at it, make this machine parseable too. * channels/chan_sip.c: Change event header to RegistrationTime to be more consistent (and avoid breaking existing frameworks). Pointed out by Laureano on #asterisk-dev. 2008-08-18 21:07 +0000 [r138738] Richard Mudgett * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, doc/tex/misdn.tex, channels/chan_misdn.c, configs/misdn.conf.sample, channels/misdn/isdn_lib.c, channels/misdn_config.c: channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/tex/misdn.tex * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. 2008-08-18 20:23 +0000 [r138687-138694] Mark Michelson * configs/queues.conf.sample, apps/app_queue.c: Change the queue timeout priority logic into less ugly and confusing code pieces. Clarify the logic within queues.conf.sample. (closes issue #12690) Reported by: atis Patches: queue_timeoutpriority.patch uploaded by atis (license 242) * apps/app_queue.c: Merged revisions 138685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug 2008) | 10 lines Change the inequalities used in app_queue with regards to timeouts from being strict to non-strict for more accuracy. (closes issue #13239) Reported by: atis Patches: app_queue_timeouts_v2.patch uploaded by atis (license 242) ........ 2008-08-18 15:54 +0000 [r138631] Jason Parker * Makefile: Remove option that isn't valid here. 2008-08-18 02:13 +0000 [r138518] Jeff Peeler * channels/chan_dahdi.c: add missing define for SS7 in dahdi_restart 2008-08-17 14:12 +0000 [r138442-138482] Sean Bright * main/features.c: Move Uniqueid to the end of the event for those that rely on the position of the name/value pairs, pointed out by snuffy-home on #asterisk-commits. For those of you who rely on the position of name/value pairs in manager events... stop... that is why associative arrays were invented. * main/features.c: Add Uniqueid header to ParkedCall manager event. (closes issue #13323) Reported by: srt Patches: 13323_unique_id_for_parkedcalls_event.diff uploaded by srt (license 378) * main/rtp.c: Add missing colons to RTCPReceived and RTCPSent manager events. (closes issue #13319) Reported by: srt Patches: 13319_rtcp_manager_event_headers.diff uploaded by srt (license 378) * channels/chan_iax2.c: Fix the output of the JitterBufStats manager event. (closes issue #13324) Reported by: srt Patches: 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt (license 378) * configs/cdr_tds.conf.sample: Since it's introduction in revision 3497, cdr_tds has *never* read the port configuration option from cdr_tds.conf. So go ahead and remove it from the sample config. 2008-08-16 13:07 +0000 [r138409-138412] Tilghman Lesher * channels/chan_dahdi.c: Fix compilation warnings (found with dev-mode) * main/pbx.c: Also make sure hinting won't crash on reload. (Closes issue #13312) 2008-08-16 01:13 +0000 [r138311-138361] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 138360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15 Aug 2008) | 1 line fixes use count to properly decrement if an active dahdi channel is destroyed allowing module to be unloaded ........ * channels/chan_dahdi.c, /: Merged revisions 138119,138151,138238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) | 4 lines Fixes the dahdi restart functionality. Dahdi restart allows one to restart all DAHDI channels, even if they are currently in use. This is different from unloading and then loading the module since unloading requires the use count to be zero. Reloading the module is different in that the signalling is not changed from what it was originally configured. Also, this fixes not closing all the file descriptors for D-channels upon module unload (which would prevent loading the module afterwards). (closes issue #11017) ........ r138151 | jpeeler | 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared static mutexes using AST_MUTEX_DEFINE_STATIC macro ........ r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) | 1 line initialize condition variable ss_thread_complete using ast_cond_init ........ 2008-08-15 22:54 +0000 [r138206-138260] Tilghman Lesher * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 138258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines More fixes for realtime peers. (closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ * main/pbx.c, configs/extensions.conf.sample: Remove deprecated syntax from sample config file (closes issue #13314) Reported by: kue 2008-08-15 20:12 +0000 [r138155] Jeff Peeler * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to dfd to match 1.4 (left over from DAHDI transition) 2008-08-15 19:36 +0000 [r138086-138148] Tilghman Lesher * main/pbx.c: Change free to ast_free_ptr, too * main/pbx.c: e->data can be NULL, so use the safe version of ast_strdup() (closes issue #13312) Reported by: pj * channels/chan_sip.c: regseconds is actually stored as the epoch time, not registration length 2008-08-15 15:09 +0000 [r138028] Russell Bryant * main/autoservice.c, /: Merged revisions 138027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) | 9 lines Ensure that when a hangup occurs in autoservice, that a hangup frame gets properly deferred to be read from the channel owner when it gets taken out of autoservice. (closes issue #12874) Reported by: dimas Patches: v1-12874.patch uploaded by dimas (license 88) ........ 2008-08-15 15:03 +0000 [r138024] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 138023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008) | 8 lines Additional check for more string specifiers than arguments. (closes issue #13299) Reported by: adomjan Patches: 20080813__bug13299.diff.txt uploaded by Corydon76 (license 14) func_strings.c-sprintf.patch uploaded by adomjan (license 487) Tested by: adomjan ........ 2008-08-14 22:43 +0000 [r137987] Russell Bryant * doc/tex/Makefile: Fix a bashism that causes an error when trying to build the pdf on ubuntu 2008-08-14 18:47 +0000 [r137933] Sean Bright * cdr/cdr_sqlite3_custom.c: Fix memory leak in cdr_sqlite3_custom. (closes issue #13304) Reported by: eliel Patches: sqlite.patch uploaded by eliel (license 64) (Slightly modified by me) 2008-08-14 18:12 +0000 [r137901] Russell Bryant * CHANGES: Prepare for adding 1.6.2 changes 2008-08-14 16:52 +0000 [r137848] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 137847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) | 9 lines When creating the secondary subchannel name, it is necessary to compare to the existing channel name without the "Zap/" or "DAHDI/" prefix, since our test string is also without that prefix. (closes issue #13027) Reported by: dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525) (Slightly modified by me, to compensate for both names) ........ 2008-08-14 15:32 +0000 [r137812] Jason Parker * channels/chan_sip.c: Make sure we set the socket port, so we don't try to use :0. (closes issue #13255) Reported by: falves11 Patches: 13255-socketport.diff uploaded by qwell (license 4) Tested by: falves11 2008-08-14 15:03 +0000 [r137780] Sean Bright * cdr/cdr_tds.c: If we detect that we are no longer connected, try to reconnect a few times before giving up. This relies on the timeout settings in the freetds.conf file and, unfortunately, on a recent version of FreeTDS (0.82 or newer). I either need to change the current execs to be non-blocking (which I do not want to do) or we have to force people to run with the latest and greatest of FreeTDS. I'm on the fence... 2008-08-14 14:15 +0000 [r137732] Russell Bryant * /, configs/sip.conf.sample: Merged revisions 137731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines Comments in this config file were aligned only if your tab size was set to 8. So, convert tabs to spaces so that things should be aligned regardless of what tab size you use in your editor. ........ 2008-08-14 02:03 +0000 [r137680] Kevin P. Fleming * /, Zaptel-to-DAHDI.txt: Merged revisions 137679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug 2008) | 1 line forgot one module name that changed ........