2012-11-05 Asterisk Development Team * Asterisk 10.10.0 Released. 2012-11-05 Asterisk Development Team * Asterisk 10.10.0-rc2 Released. * Fix a bug which made ConfBridge not record the conference when the recording was initiated from an AMI/CLI command (closes issue ASTERISK-20601) Reported by: Vilius 2012-10-08 Asterisk Development Team * Asterisk 10.10.0-rc1 Released. 2012-10-08 13:46 +0000 [r374652] Matthew Jordan * apps/confbridge/conf_state.c (added), apps/confbridge/conf_state_single.c (added), apps/confbridge/conf_state_inactive.c (added), apps/confbridge/conf_state_single_marked.c (added), apps/confbridge/include/confbridge.h, apps/confbridge/include/conf_state.h (added), apps/confbridge/conf_state_multi.c (added), apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c (added), apps/confbridge/conf_state_empty.c (added): Resolve issues in ConfBridge regarding marked, waitmarked, and unmarked users Thank's to Neil Tallim (flan)'s tireless testing, issue reporting, and patches it became clear that app_confbridge had some complex logic in how it handled interactions between marked, waitmarked, and unmarked users. In particular, there were some areas in which the interactions between the users resulted in inconsistent behavior, and app_confbridge was missing logic in how to handle some corner cases. Some areas included: * Poor handling of mixing unmarked and waitmarked users * Inconsistencies in how MOH and muting was applied to various users * Handling of various announcements for different user profile options flan's patches seem to fix the various issues, but highlighted how hard the code could be to maintain. In an attempt to make things easier to maintain and to more fully enumerate the various cases that exist, this patch breaks up the logic into a state machine-like setup. Please note that the various state transitioned are documented on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes Review: //https://reviewboard.asterisk.org/r/2072/ Note that for the following issues, mjordan uploaded the patch, although it was written by twilson. Any contributor license discrepency is due to that. (closes issue ASTERISK-19562) Reported by: flan Tested by: flan, mjordan, jrose patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) (closes issue ASTERISK-19726) Reported by: flan Tested by: flan patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) (closes issue ASTERISK-20181) Reported by: Jonathan White Tested by: Jonathan White patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) 2012-10-05 20:23 +0000 [r374586] dlee : * main/manager.c, /: Multiple revisions 374570,374581 ........ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines Improve AMI long line error handling In AMI's parser, when it receives a long line (> 1024 characters), it discards that line, but continues to process the message normally. Typically, this is not a problem because a) who has lines that long and b) usually a discarded line results in an invalid message. But if that line is specifying an optional field, then the message will be processed, you get a 'Response: Success', but things don't work the way you expected them to. This patch changes the behavior when a line-too-long parse error occurs. * Changes the log message to avoid way-too-long (and truncated anyways) log messages * Adds a 'parsing' status flag to Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long * Responds with an appropriate error if parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line I've committed too much. Reverting part of r374570. ........ Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-10-05 18:25 +0000 [r374537] Richard Mudgett * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-10-04 20:17 +0000 [r374476-374481] Alec L Davis * main/dsp.c, /, configs/dsp.conf.sample, CHANGES: dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of a recompile, allow values to be adjusted in dsp.conf For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2144/ ........ Merged revisions 374479 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/dsp.c, /: dsp.c fix incorrect DTMF Digit_Duration. it's always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2 (issue ASTERISK-16003) Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2145/ ........ Merged revisions 374475 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-10-04 17:44 +0000 [r374457] Joshua Colp * /, channels/chan_sip.c: Fix a regression from direct media ACLs where the directrtpsetup option no longer works. A check was added for direct media ACLs that immediately forbid remote bridging if there was no bridged channel. This caused directrtpsetup to no longer function as it needs this information before bridging actually occurs. Logic has now been adjusted so if there is no bridged channel a remote bridge will still be attempted. (closes issue ASTERISK-20511) Reported by: kristoff Review: https://reviewboard.asterisk.org/r/2146/ ........ Merged revisions 374456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-10-04 15:37 +0000 [r374427] dlee : * /, res/res_agi.c, main/db.c: Fix DBDelTree error codes for AMI, CLI and AGI The AMI DBDelTree command will return Success/Key tree deleted successfully even if the given key does not exist. The CLI command 'database deltree' had a similar problem, but was saved because it actually responded with '0 database entries removed'. AGI had a slightly different error, where it would return success if the database was unavailable. This came from confusion about the ast_db_deltree retval, which is -1 in the event of a database error, or number of entries deleted (including 0 for deleting nothing). * Changed some poorly named res variables to num_deleted * Specified specific errors when calling ast_db_deltree (database unavailable vs. entry not found vs. success) * Fixed similar bug in AGI database deltree, where 'Database unavailable' results in successful result (closes issue AST-967) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2138/ ........ Merged revisions 374426 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-10-04 04:41 +0000 [r374370-374385] Alec L Davis * main/dsp.c, /, configs/dsp.conf.sample, CHANGES: dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries. Various countries have different specifications for the maximum power level differences between the DTMF low group and high group of frequencies. Power level difference between frequencies for different Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03) Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T specifications Add's the following variables to dsp.conf ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51 ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98 (closes issue ASTERISK-20442) Reported by: tbsky Tested by: tbsky,alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2141/ ........ Merged revisions 374384 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/dsp.c, /: _dsp_init: bring inline with trunk preparation for clean merge of DTMF TWIST patch No functional changes, just style. alecdavis (license 585) Reported by: Alec Davis Tested by: alecdavis related https://reviewboard.asterisk.org/r/2141 ........ Merged revisions 374365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-10-04 02:11 +0000 [r374178-374336] Matthew Jordan * /, res/res_jabber.c: Check for presence of buddy in info/dinfo handlers The res_jabber resource module uses the ASTOBJ library for managing its ref counted objects. After calling ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to the object has to be checked to see if the buddy existed. Prior to this patch, the buddy object was not checked for NULL; with this patch in both aji_client_info_handler and aji_dinfo_handler the pointer is checked before used and, if no buddy object was found, the handlers return an error code. This patch does not take the approach that our JID can be used to log in from another resource. If that approach is desired, an improvement could be made to this patch to create the buddy on the fly. This patch seeks only to prevent Asterisk from crashing. Note that multiple people have proposed patches for this issue; the patch being committed here is based on those. (closes issue ASTERISK-19532) Reported by: Karsten Wemheuer Tested by: Byron Clark patches: fix-jabber uploaded by Karsten Wemheuer (license #5930) xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark (license #6157) (closes issue ASTERISK-19557) Reported by: ulugutz ........ Merged revisions 374335 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/ccss.c: Destroy the generic_monitors container after the core_instances in ccss For each item in core_instances disposed of in the shutdown of ccss, any generic monitor instances referenced by the objects will be removed from generic_monitors during their destruction. Hilarity ensues if generic_monitors no longer exists. Thanks to the Asterisk Test Suite's generic_ccss test for complaining loudly when it ran into this. * /, main/asterisk.c: Ensure Shutdown AMI event is still fired during Asterisk shutdown Richard pointed out that having the manager dispose of itself gracefully during shutdown meant that the Shutdown event will no longer get fired. This patch moves the AMI event just prior to running the atexit callbacks. ........ Merged revisions 374230 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/message.c: Fix findings from check-in on r374177 Richard pointed out two problems with the check-in from r374177: * The ast_msg_shutdown function declaration doesn't match the prototype in main/message.c. * The ref/alloc function usage in astobj2 (in 11+) can use the ao2_t_* variants of the functions to allow the REF_DEBUG flag to enable/disable their debug counterparts. * main/channel.c, main/format.c, main/data.c, main/pbx.c, main/manager.c, /, main/ccss.c, channels/chan_agent.c, main/features.c, main/cel.c, main/format_pref.c, main/indications.c, main/message.c, main/asterisk.c, main/db.c: Fix a variety of ref counting issues This patch resolves a number of ref leaks that occur primarily on Asterisk shutdown. It adds a variety of shutdown routines to core portions of Asterisk such that they can reclaim resources allocate duringd initialization. Review: https://reviewboard.asterisk.org/r/2137 ........ Merged revisions 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-10-01 17:52 +0000 [r374132-374135] Sean Bright * include/asterisk/astdb.h, tests/test_db.c, apps/app_queue.c, main/db.c: app_queue: Support persisting and loading of long member lists. Greenlight in #asterisk brought up that he was receiving an error message "Could not create persistent member string, out of space" when running app_queue in Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to store the generated string, but with queues that have large member lists this is not always the case. This patch removes the limitation and uses ast_str instead of a fixed sized buffer. The complicating factor comes from the fact that ast_db_get requires a buffer and buffer size argument, which doesn't let us pull back more than what we pass in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d copy of the value from astdb. As an aside, I did some testing on the maximum size of data that we can store in the BDB library we distribute and was able to store a 10MB string and retrieve it with no problems, so I feel this is a safe patch. Review: https://reviewboard.asterisk.org/r/2136/ ........ Merged revisions 374108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/db.c: Use ast_copy_string instead of strncpy to guarantee a NUL terminated string. 2012-09-28 19:21 +0000 [r374045] Jonathan Rose * /, res/res_jabber.c: res_jabber: Remove CLI command 'jabber test' The opinion of development was that it is both improper to have Matt's personal email address used in the source and that the command wouldn't be useful without it. (closes issue AST-467) Reported by: Malcolm Davenport ........ Merged revisions 374032 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-28 12:15 +0000 [r373990] Joshua Colp * /, res/res_agi.c: Update documentation to make it explicit that "stream file" will not restart musiconhold. (issue ASTERISK-17367) Reported by: oej ........ Merged revisions 373989 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-27 22:12 +0000 [r373946] Richard Mudgett * /, apps/app_senddtmf.c: Fix SendDTMF crash and channel reference leak using channel name parameter. The SendDTMF channel name parameter has two issues. 1) Crashes if the channel name does not exist. 2) Leaks a channel reference if the channel is the current channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF documentation. * Renamed app to senddtmf_name and tweaked the type. ........ Merged revisions 373945 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-27 16:50 +0000 [r373879-373910] Joshua Colp * main/loader.c, /: loader: Ensure dependent modules are properly initialized. If an Asterisk module specifies a dependency in ast_module_info.nonoptreq, it is possible for Asterisk to skip calling the modules's .load function. Asterisk was loading and linking the module via load_dynamic_module() but was not adding the module to the resource_heap. Therefore the module was not initialized based on it's priority along with the other modules in the heap. Now use load_resource() instead of load_dynamic_module() for non-optional requirement. This will add the module to the resource_heap so the module can be properly initialized in the correct order. This is required if there are any module global data structures initialized in the .load() callback for the module on platforms which do not support weak references. (issue ASTERISK-20439) Reported by: sruffell Patches: 0001-loader-Ensure-dependent-modules-are-properly-initial.patch uploaded by sruffell (license 5417) ........ Merged revisions 373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_local.c, /: Fix an issue where Local channels dialed by app_queue are considered in use immediately. The chan_local channel driver returns a device state of in use even if a created Local channel has not yet been dialed. This fix changes the logic to return a state of not in use until the channel itself has been dialed. (closes issue ASTERISK-20390) Reported by: tim_ringenbach Review: https://reviewboard.asterisk.org/r/2116/ ........ Merged revisions 373878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-26 21:11 +0000 [r373849] Mark Michelson * /, channels/chan_sip.c: Move handling of 408 response so there is no misleading warning message. (closes issue ASTERISK-20060) Reported by: Walter Doekes ........ Merged revisions 373848 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-26 18:15 +0000 [r373816] Richard Mudgett * /, apps/app_meetme.c: Fixed meetme tab completion and command documentation. * Removed unnecessary case sensitivity in meetme list, lock, unlock, mute, unmute, and kick commands. * Separated meetme lock/unlock, mute/unmute, and kick commands into their own registered commands to simplify tab completion and parameter checking. meetme_lock_cmd(), meetme_mute_cmd(), and meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue AST-1006) Reported by: John Bigelow Tested by: rmudgett ........ Merged revisions 373815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-25 23:08 +0000 [r373737-373774] Mark Michelson * /, main/say.c: Fix saying of date in Dutch. The Dutch say the date before the month. (closes issue ASTERISK-20353) Reported by: Teun Ouwehand ........ Merged revisions 373773 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * configs/agents.conf.sample, /, channels/chan_agent.c: Remove dead code and documentation for nonexistent feature. multiplelogin was removed from chan_agent back in 1.6.0 when AgentCallbackLogin() was removed. (closes issue AST-948) reported by Steve Pitts ........ Merged revisions 373768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: Fix error where improper IMAP greetings would be deleted. (closes issue ASTERISK-20435) Reported by: fhackenberger Patches: asterisk-20435-imap-del-greeting.diff uploaded by Michael L. Young (License #5026) (with suggested modification made by me) ........ Merged revisions 373735 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-25 20:12 +0000 [r373706] Joshua Colp * channels/chan_local.c, /: Fix T.38 support when used with chan_local in between. Users of the T.38 API can indicate AST_T38_REQUEST_PARMS on a channel to request that the channel indicate a T.38 negotiation with the parameters present on the channel. The return value of this indication is expected to be AST_T38_REQUEST_PARMS upon success but with chan_local involved this could never occur. This fix changes chan_local to always return AST_T38_REQUEST_PARMS for this situation. If the underlying channel technology on the other side does not support T.38 this would have been determined ahead of time using ast_channel_get_t38_state and an indication would not occur. (closes issue ASTERISK-20229) Reported by: wdoekes Patches: ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review: https://reviewboard.asterisk.org/r/2070/ ........ Merged revisions 373705 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-25 19:34 +0000 [r373675-373703] Kinsey Moore * res/res_rtp_asterisk.c, /: Fix an issue where media would not flow for situations where the legacy STUN code is in use. The STUN packets should *not* be blocked by strict RTP. (closes issue ASTERISK-20415) Reported-by: Michele Cicciotti Patch-by: Josh Colp (trunk r369817) ........ Merged revisions 373702 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: "show" completion option for "queue" shouldn't appear twice When tab-completing CLI commands starting with "queue", "show" appeared twice in the list due to the way that Asterisk's tab completion functions and the order in which the commands were registered. The registration order has been altered to resolve this issue. (closes issue AST-940) Reported-by: Steve Pitts ........ Merged revisions 373666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-25 17:35 +0000 [r373665] Terry Wilson * /, channels/chan_sip.c, configs/sip.conf.sample, channels/sip/include/sip.h: Properly handle UAC/UAS roles for SIP session timers The SIP session timer mechanism contains a mandatory 'refresher' parameter (included in the Session-Expires header) which is used in the session timer offer/answer signaling within a SIP Invite dialog. It looks like asterisk is interpreting the uac resp. uas role only as the initial role of client and server (caller is uac, callee is uas). The standard rfc 4028 however assigns the client role to the ((RE)-Invite) requester, the server role to the ((RE)-Invite) responder. This patch has Asterisk track the actual refresher as "us" or "them" as opposed to relying on just the configured "uas" or "uac" properties. (closes issue AST-922) Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged revisions 373652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-25 17:19 +0000 [r373633-373645] Richard Mudgett * /, codecs/ilbc/iLBC_encode.c, codecs/ilbc/iLBC_decode.c: Fix valgrind found memcpy issues in codec_ilbc. Valgrind found codec_ilbc using memcpy instead of memmove for overlapping memory blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231) Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license #5674) patch uploaded by Walter Doekes ........ Merged revisions 373640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * codecs/Makefile, /: Make rebuild GSM, ilbc, or lpc10 codecs if the respective sources change. ........ Merged revisions 373618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-25 16:24 +0000 [r373631] Jonathan Rose * /, channels/chan_sip.c: chan_sip: Set Quality of Service for video rtp instance (closes issue ASTERISK-20201) Reported by: ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license 6008) ........ Merged revisions 373617 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-25 13:28 +0000 [r373579] Kinsey Moore * configs/res_odbc.conf.sample, /: Fix documentation for default username in res_odbc This was previously stated to be "root", but is actually the name of the context if unspecified. (closes issue ASTERISK-20258) Reported by: Stefan x ........ Merged revisions 373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-25 12:00 +0000 [r373533-373551] Joshua Colp * res/res_rtp_multicast.c, /: Fix an issue where a caller to ast_write on a MulticastRTP channel would determine it failed when in reality it did not. When sending RTP packets via multicast the amount of data sent is stored in a variable and returned from the write function. This is incorrect as any non-zero value returned is considered a failure while a return value of 0 is success. For callers (such as ast_streamfile) that checked the return value they would have considered it a failure when in reality nothing went wrong and it was actually a success. The write function for the multicast RTP engine now returns -1 on failure and 0 on success, as it should. (closes issue ASTERISK-17254) Reported by: wybecom ........ Merged revisions 373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Add missing checks that I neglected. The SIP technology and SIP info technology should be considered equal. (closes issue ASTERISK-20409) Reported by: michele cicciotti privatewave ........ Merged revisions 373532 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-24 22:17 +0000 [r373505] Matthew Jordan * res/res_rtp_asterisk.c, /: Revert change to res_rtp_asterisk committed in r373236 (1.8) The change committed in r373236 attempted to account for endpoints that increased their RTP timestamp in DTMF end of event re-transmissions. This change attempted to make Asterisk continue to work with endpoints that failed to follow the RFC while maintaining the fix that allowed for out of order DTMF to be handled. Unfortunately, there is no free lunch, and this patch broke any system that sent DTMF immediately after an RTP session was established or when an SSRC is updated. As such, that patch is being reverted for the previous behavior. Endpoints that erroneously increase the RTP timestamp in DTMF end of event packets will not work properly with Asterisk. (issue ASTERISK-20424) ........ Merged revisions 373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-24 22:11 +0000 [r373501] Richard Mudgett * /, channels/chan_sip.c: Be consistent, send From: "Anonymous" When setting CALLERID(pres)=unavailable in the dialplan, the From header in the SIP message contains "Anonymous" . For consistency, Asterisk should use a lowercase a in the userpart of the URI. * Make the From header use a lowercase A in the userpart of the anonymous URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383) patch uploaded by Antti Yrjola ........ Merged revisions 373500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-24 21:05 +0000 [r373468] Jonathan Rose * funcs/func_audiohookinherit.c, /, apps/app_mixmonitor.c: func_audiohookinherit: Document some missed sources. This patch also mentions that AUDIOHOOK_INHERIT can be used to transfer MixMonitor audiohooks. There is also wiki that addresses audiohooks and the use of AUDIOHOOK_INHERIT at the following link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........ Merged revisions 373467 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-24 20:44 +0000 [r373466] Richard Mudgett * /, channels/chan_sip.c: Fix potential reentrancy problems in chan_sip. Asterisk v1.8 and later was not as vulnerable to this issue. * Made find_call() lock each private as it processes the found dialogs. (Primary cause of ABE-2876) * Made the other functions that traverse the dialogs container lock each private as it examines them. * Fix race condition in sip_call() if the thread that sent the INVITE is held up long enough for a response to be processed. The p->initid for the INVITE retransmission could be added after it was canceled by the response processing. * Made __sip_destroy() clean up resource pointers after freeing. This is primarily defensive in case someone has a stale private pointer. * Removed redundant memset() in reqprep(). The call to init_req() already does the memset() and is the first reference to req in reqprep(). * Removed useless set of req.method in transmit_invite(). The calls to initreqprep() and reqprep() have to do this because they memset() the req. JIRA ABE-2876 .......... Merged -r373423 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 373424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-24 19:17 +0000 [r373440] Joshua Colp * /, channels/chan_sip.c: Fix a deadlock caused by a race condition between removing a hint and reloading the dialplan and subscribing to the removed hint. If conditions were right it was possible for both the PBX core and chan_sip to deadlock by both having a lock that the other wants. In the case of the PBX core it had the contexts lock and wanted a SIP dialog lock, while in the case of chan_sip it had the SIP dialog lock and wanted the contexts lock. This fix unlocks the SIP dialog before getting the extension state so that the other thread will not block on trying to lock it. Once the extension state is retrieved the SIP dialog is locked again and life carries on. As the SIP dialog is reference counted it is not possible for it to go away after unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins ........ Merged revisions 373438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-21 19:08 +0000 [r373300-373343] Jonathan Rose * /, channels/iax2-provision.c: iax2-provision: Fix improper return on failed cache retrieval (closes issue ASTERISK-20337) reported by: John Covert Patches: iax2-provision.c.patch uploaded by John Covert (license 5512) ........ Merged revisions 373342 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: app_queue: Make queue reload members and variants of that work Prior to this patch, 'queue reload members' cli command did not work at all. This also affects the manager function 'QueueReload' when supplied with the 'members: yes' field. (closes issue AST-956) Reported by: John Bigelow ........ Merged revisions 373298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-20 19:14 +0000 [r373245] Joshua Colp * /, apps/app_meetme.c: Fix incorrect MeetME conference bridge reference count decrementing and sometimes premature destruction. When using the 'e' or 'E' option to MeetMe the configured conference bridges are loaded and examined to see if any are empty. If no conference bridges are empty the caller is prompted to enter the number of one. This operation left around a pointer to the last created conference bridge still containing participants. When the caller that was not able to find any empty conference bridge hung up this pointer was disposed of and the reference count of the conference bridge decremented. If there was only a single participant in the conference bridge it was ultimately destroyed prematurely. (closes issue AST-994) Reported by: John Bigelow ........ Merged revisions 373242 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-20 18:42 +0000 [r373196-373237] Matthew Jordan * res/res_rtp_asterisk.c, /: When processing RFC 2833 DTMF, accomodate increasing timestamps in End events While endpoints should not be changing the source timestamp between DTMF event packets, the fact is there exists those endpoints that do exactly that. To work around this, we absorb timestamps within the expected re-transmit period. Note that this period only affects End of Event packets, so it should not prevent the detection of new DTMF digits that happen to arrive right on top of each other. (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson Tested by: mjordan, Vladimir Mikhelson Review: https://reviewboard.asterisk.org/r/2124 ........ Merged revisions 373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/confbridge/conf_config_parser.c: Ensure that all ConfBridge sounds can be set using CONFBRIDGE function The CONFBRIDGE function can be used to set the sounds in a ConfBridge bridge profile. Unfortunately, three sounds were missed in the portion of the code that applies the settings passed in from the function: sound_only_one, join, and leave. This patch makes sure that the sounds passed from the function are applied to the bridge profile. (closes issue ASTERISK-20404) Reported by: univ Tested by: mjordan 2012-09-19 17:05 +0000 [r373179] Joshua Colp * /, channels/chan_sip.c: Fix a regression where direct media was not permitted for calls using SIP INFO DTMF. A change was committed to fix direct media ACL support. This change wrongly assumed that only a single channel technology structure exists for chan_sip. This is in fact false as a second exists for calls using SIP INFO DTMF. The code which performs direct media ACL checking now checks for both the non-INFO DTMF and INFO DTMF channel technology structures. (closes issue ASTERISK-20409) Reported by: michele cicciotti privatewave ........ Merged revisions 373165 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-18 20:13 +0000 [r373132] Sean Bright * main/manager.c, /: Don't crash when passing a NULL message to __astman_get_header. Before this commit, __astman_get_header would blindly dereference the passed in 'struct message *' to traverse the header list. There are cases, however, such as '*CLI> sip qualify peer foo' where the message pointer is NULL, so we need to check for that. ........ Merged revisions 373131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-15 00:20 +0000 [r373101] Richard Mudgett * /, channels/sig_ss7.c: Made companding law for SS7 calls only determined by SS7 signaling type. For SS7, the companding law for a call was chosen inconsistently depending upon ss7type (ITU vs ANSI) and the DAHDI companding default (T1 vs E1). For incoming calls, the companding law was determined by ss7type. For outgoing calls, the companding law was determined by the DAHDI default. With the wrong combination you would get A-law/u-law conflicts. An A-law/u-law conflict sounds like bad static on the line. SS7 ITU signaling with E1 line: ok SS7 ITU signaling with T1 line: noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling with T1 line: ok * Fix the companding law used to be determined by the SS7 signaling type only. ........ Merged revisions 373090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-14 19:12 +0000 [r373059-373062] Matthew Jordan * main/tcptls.c, /, channels/chan_sip.c, main/ssl.c: Resolve memory leaks in TLS initialization and TLS client connections This patch resolves two sources of memory leaks when using TLS in Asterisk: 1) It removes improper initialization (and multiple re-initializations) of portions of the SSL library. Asterisk calls SSL_library_init and SSL_load_error_strings during SSL initialization; collectively this obviates the need for calling any of the following during initialization or client connection handling: * ERR_load_crypto_strings (handled by SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for SSL_library_init) 2) Failure to completely clean up all memory allocated by Asterisk and by the SSL library for TLS clients. This included not freeing the SSL_CTX object in the SIP channel driver, as well as not clearing the error stack when the TLS client exited. Note that these memory leaks were found by Thomas Arimont, and this patch was essentially written by him with some minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas Arimont (license 5525) Review: https://reviewboard.asterisk.org/r/2105 ........ Merged revisions 373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, include/asterisk/astobj2.h, main/astobj2.c: Constify __ao2_ref_debug in astobj2 When REF_DEBUG is enabled in certain files - most notably ccss.c - the 'tag' parameter passed to __ao2_ref_debug will be a const char *. The function currently expects that parameter to not be const. This causes a warning when compiling, as the const qualifier is being discarded. With dev-mode enabled, this prevents compiling Asterisk. This patch makes __ao2_ref_debug's tag and file parameters const. (closes issue ASTERISK-20408) Reported by: mjordan ........ Merged revisions 372959 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-13 18:44 +0000 [r373025] dlee : * main/channel.c, /, include/asterisk/channel.h: Fix timeouts for ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass its timeout to ast_waitfor_nandfds, expecting it to decrement the timeout by however many milliseconds were waited. This is a problem if it consistently waits less than 1ms. The timeout will never be decremented, and we wait... FOREVER! This patch makes ast_waitfordigit_full manage the timeout itself. It maintains the previously undocumented behavior that negative timeouts wait forever. (closes issue ASTERISK-20375) Reported by: Mark Michelson Tested by: Mark Michelson Review: https://reviewboard.asterisk.org/r/2109/ ........ Merged revisions 373024 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-13 Asterisk Development Team * Asterisk 10.9.0-rc1 Released. 2012-09-12 14:53 +0000 [r372933] Mark Michelson * /, channels/chan_sip.c: Add channel name to a warning to make debugging easier. The "autodestruct with owner in place" message is typically indicative of a channel reference leak. Printing out the name of the channel in the message may be helpful when trying to debug the issue. ........ Merged revisions 372932 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-11 22:23 +0000 [r372916] Jonathan Rose * channels/chan_local.c, /: chan_local: Switch from using a random 4 digit hex identifier to unique id Changes chan_local channels to use an 8 digit hex identifier generated atomically and sequentially in order to eliminate the chance of having multiple channels with the same name during high call volume situations. (issue ASTERISK-20318) Reported by: Dan Cropp Review: https://reviewboard.asterisk.org/r/2104/ ........ Merged revisions 372902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-11 21:04 +0000 [r372885] Mark Michelson * include/asterisk/_private.h, main/message.c, main/asterisk.c: Fix inability to shutdown gracefully due to an unending channel reference. message.c makes use of a special message queue channel that exists in thread storage. This channel never goes away due to the fact that the taskprocessor used by message.c does not get shut down, meaning that it never ends the thread that stores the channel. This patch fixes the problem by shutting down the taskprocessor when Asterisk is shut down. In addition, the thread storage has a destructor that will release the channel reference when the taskprocessor is destroyed. (closes issue AST-937) Reported by Jason Parker Patches: AST-937.patch uploaded by Mark Michelson (License #5049) Tested by Jason Parker 2012-09-11 17:14 +0000 [r372863] dlee : * Makefile: Corrects the astsbindir setting when installing the sample asterisk.conf. (closes issue ASTERISK-20406) 2012-09-11 15:30 +0000 [r372841] Mark Michelson * /, main/features.c: Fix bad channel application data reference. When channels get bridged due to an AMI bridge action or a DTMF attended transfer, the two channels that get bridged have their application data pointing to the other channel's name. This means that if one channel is hung up but the other moves on, it means that the channel that moves on will have its application data pointing at freed memory. (issue ASTERISK-20335) Reported by: aragon ........ Merged revisions 372840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-10 20:56 +0000 [r372805] Kinsey Moore * /, channels/chan_iax2.c: Ensure iax2 debug output is displayed when expected When IAX2 debug was changed from iax_showframe to iax_outputframe, some instances were missed (or added afterward). This was causing debug output to not be displayed when expected. (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by: John Covert ........ Merged revisions 372804 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-10 18:41 +0000 [r372767] Jonathan Rose * /, apps/app_meetme.c: app_meetme: Document that 'p' option will continue in dialplan. (closes issue AST-991) Reported by John Bigelow ........ Merged revisions 372765 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-10 18:32 +0000 [r372764] Kinsey Moore * /, channels/chan_sip.c: Warn on CLI when UDPTL init fails This adds a CLI warning when a SDP offer is rejected due to UDPTL initialization failure. Previously, there was no indication of the reason for offer rejection in this case. (closes issue ASTERISK-20357) Reported-by: Francesco Usseglio Gaudi ........ Merged revisions 372763 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-10 17:14 +0000 [r372737] Jonathan Rose * main/channel.c, /: Masquerade: Retain parkinglot settings made by CHANNEL function. Prior to this patch, the user would have a parkinglot set on a channel that was parked and when the channel was retrieved, any attempt by that channel to park would simply use the default. This patch makes parkinglot values set in this way be retained through the masquerade. (closes issue AST-990) Reported by: Nick Huskinson Patches: masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose (license 6182) ........ Merged revisions 372736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-09 01:24 +0000 [r372710] Matthew Jordan * channels/sip/sdp_crypto.c, /: Only re-create an SRTP session when needed In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an SDP offer and the ability to re-create an SRTP session when the crypto keys changed. In certain circumstances - most notably when a phone is put on hold after having been bridged for a significant amount of time - the act of re-creating the SRTP session causes problems for certain models of phones. The patch committed in r356604 always re-created the SRTP session regardless of whether or not the cryptographic keys changed. Since this is technically not necessary, this patch modifies the behavior to only re-create the SRTP session if Asterisk detects that the remote key has changed. This allows models of phones that do not handle the SRTP session changing to continue to work, while also providing the behavior needed for those phones that do re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported by: Nicolo Mazzon Tested by: Nicolo Mazzon Review: https://reviewboard.asterisk.org/r/2099 ........ Merged revisions 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-08 05:21 +0000 [r372695] dlee : * /, main/Makefile: Add OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c. Without this flag, those files will compile with the system installed OpenSSL headers (if they exist). This is a real bummer if a different path was specified using --with-ssl= (closes issue ASTERISK-20392) ........ Merged revisions 372682 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-07 23:06 +0000 [r372621-372656] Richard Mudgett * /, main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup(). (closes issue ASTERISK-20349) Reported by: Brent Eagles ........ Merged revisions 372655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, funcs/func_math.c: Remove annoying unconditional debug message from INC/DEC functions. (closes issue AST-1001) Reported by: Guenther Kelleter ........ Merged revisions 372628 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Fix exception path typo in app_queue.c try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy Pepper Patches: fix-local-channel-locking.patch (license #6350) patch uploaded by Jeremy Pepper ........ Merged revisions 372624 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: Fix VoicemailUserEntry event headers ServerEmail and MailCommand reported values. The AMI action VoicemailUsersList VoicemailUserEntry event headers ServerEmail and MailCommand did not report the global values if they were not overridden. The VoicemailUserEntry event header ServerEmail was not populated with the global value if the voicemail user did not override it. The VoicemailUserEntry event header MailCommand was never populated with a value. * Removed unused struct ast_vm_user member mailcmd[]. (closes issue AST-973) Reported by: John Bigelow Tested by: rmudgett ........ Merged revisions 372620 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-07 02:25 +0000 [r372555-372582] Matthew Jordan * /, apps/app_minivm.c: Free ast_str objects when temp file fails to be created in MiniVM The previous commit (r372554) was from a patch that was written before r366880, which ensured that ast_str objects allocated in the sendmail routine were free'd in off nominal paths. This commit frees the string objects in the off nominal path introduced in r372554. (issue ASTERISK-17133) Reported by: Tzafrir Cohen ........ Merged revisions 372581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_minivm.c: Fix file descriptor leak and pointer scope issue in MiniVM when sending mail When MiniVM sends an e-mail and it has the volgain option set, it will spawn sox in a separate process to handle the manipulation of the sound file. In doing so, it creates a temporary file. There are two problems here: 1) The file descriptor returned from mkstemp is leaked 2) The finalfilename character pointer points to a buffer that loses scope once volgain processing is finished. Note that in r316265, Russell fixed some gcc warnings by using the return value of the mkstemp call. A warning was placed in minivm that the file descriptor was going to be leaked. This patch reverts that change, as it handles the leak and 'uses' the file descriptor returned from mkstemp. (closes issue ASTERISK-17133) Reported by: Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir Cohen (license #5035) ........ Merged revisions 372554 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-06 22:10 +0000 [r372522] Richard Mudgett * channels/sig_pri.c: Fix loss of MOH on an ISDN channel when parking a call for the second time. Using the AMI redirect action to take an ISDN call out of a parking lot causes the MOH state to get confused. The redirect action does not take the call off of hold. When the call is subsequently parked again, the call no longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on repeated AST_CONTROL_HOLD frames if it is already in a state where it is supposed to be sending MOH. The MOH may have been stopped by other means. (Such as killing the generator.) This simple fix is done rather than making the AMI redirect action post an AST_CONTROL_UNHOLD unconditionally when it redirects a channel and thus potentially breaking something with an unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches: jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 372521 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier 2012-09-06 21:40 +0000 [r372518] Kinsey Moore * /, apps/app_queue.c: Ensure listed queues are not offered for completion When using tab-completion for the list of queues on "queue reset stats" or "queue reload {all|members|parameters|rules}", the tab-completion listing for further queues erroneously listed queues that had already been added to the list. The tab-completion listing now only displays queues that are not already in the list. (closes issue AST-963) Reported-by: John Bigelow ........ Merged revisions 372517 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-06 18:54 +0000 [r372499] dsessions : * channels/chan_sip.c, configs/res_ldap.conf.sample: LDAP Realtime Peers Cannot Register Prior to 1.8, it was not necessary for an explicit "type" to be set for an asterisk LDAP realtime peer. Now the routine find_peer actually checks the type field during registration and fails to find the peer if it is not set. The attached patches make the realtime type equal whatever type is being searched for if the type is 0 upon return from routine build_peer. (closes issue ASTERISK-17222) Reported by: John Covert Patch by: David Vossel Tested by: Darren Sessions Review: https://reviewboard.asterisk.org/r/2095/ 2012-09-06 15:54 +0000 [r372472] Jonathan Rose * /, UPGRADE-1.8.txt: chan_sip: Note change in behavior to how directmediapermit/deny ACL works r366547 introduced a change to the directmedia ACL for chan_sip which modified the behavior significantly. Prior to the patch, this option would bridge peers with directmedia if a peer's IP address matched its own directmedia ACL. After that patch, the peer would check the bridged peer's ACL instead. This change has been present since 1.8.14.0. That patched failed to document the change in Upgrade.txt, so this patch adds mention of that change to UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876) ........ Merged revisions 372471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-06 14:29 +0000 [r372445] Kinsey Moore * /, apps/app_queue.c: Ensure "rules" is tab-completable for "queue show" Previously, tabbing at the end of "queue show" produced a list of available queues about which information could be shown, but did not include an alternative command, "rules", to access information about queue rules. The "rules" item should now be shown in the list of tab-completable items. (closes issue AST-958) Reported-by: John Bigelow ........ Merged revisions 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-06 02:49 +0000 [r372391-372418] Matthew Jordan * /, pbx/pbx_dundi.c: Fix DUNDi message routing bug when neighboring peer is unreachable Consider a scenario where DUNDi peer PBX1 has two peers that are its neighbors, PBX2 and PBX3, and where PBX2 and PBX3 are also neighbors. If the connection is temporarily broken between PBX1 and PBX3, PBX1 should not include PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER message, as it cannot send messages to PBX3. If it does, PBX2 will assume that PBX3 already received the message and fail to forward the message on to PBX3 itself. This patch fixes this by only including peers in a DPDISCOVER message that are reachable by the sending node. This includes all peers with an empty address (00:00:00:00:00:00) and that are have been reached by a qualify message. This patch also prevents attempting to qualify a dynamic peer with an empty address until that peer registers. (closes issue ASTERISK-19309) Reported by: Peter Racz patches: dundi_routing.patch uploaded by Peter Racz (license 6290) The patch uploaded by Peter was modified slightly for this commit. ........ Merged revisions 372417 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_followme.c: Allow configured numbers for FollowMe to be greater than 90 characters When parsing a 'number' defined in followme.conf, FollowMe previously parsed the number in the configuration file into a buffer with a length of 90 characters. This can artificially limit some parallel dial scenarios. This patch allows for numbers of any length to be defined in the configuration file. Note that Clod Patry originally wrote a patch to fix this problem and received a Ship It! on the JIRA issue. The patch originally expanded the buffer to 256 characters. Instead, the patch being committed duplicates the string in the config file on the stack before parsing it for consumption by the application. (closes issue ASTERISK-16879) Reported by: Clod Patry Tested by: mjordan patches: followme_no_limit.diff uploaded by Clod Patry (license #5138) Slightly modified for this commit. ........ Merged revisions 372390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-05 19:42 +0000 [r372372] Richard Mudgett * main/dsp.c: Fix compile error. 2012-09-05 19:22 +0000 [r372358] Kinsey Moore * main/manager.c, /: Correct documentation for ModuleLoad AMI action The documentation incorrectly listed 'rtp' as a reloadable subsystem and left out many other reloadable subsystems. It is now also documented that subsystems may only be reloaded, not loaded or unloaded. (closes issue AST-977) Reported-by: John Bigelow ........ Merged revisions 372354 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-05 18:43 +0000 [r372341] Alec L Davis * main/dsp.c, /: dsp.c: in ast_mf_detect_init incorrectly sets goertzel samples to 160, should be MF_GSIZE Related https://reviewboard.asterisk.org/r/2097/ ........ Merged revisions 372339 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-05 18:30 +0000 [r372338] Kinsey Moore * main/pbx.c, /: Ensure counts generated in manager_show_dialplan_helper are correct When manager_show_dialplan_helper was written, the counter increment for the total number of contexts was placed with the extensions increment instead of in the enclosing loop. This function should now generate correct context counts. (closes issue AST-970) Reported-by: John Bigelow ........ Merged revisions 372337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-05 13:42 +0000 [r372288] Matthew Jordan * apps/app_voicemail.c, /: Fix memory leaks in app_voicemail when using IMAP storage or realtime config This patch fixes two memory leaks: 1. When find_user is called with NULL as its first parameter, the voicemail user returned is allocated on the heap. The inboxcount2 function uses find_user in such a fashion when counting new messages, and fails to free the resulting voicemail user object. 2. When populate_defaults is called on a voicemail user, it wipes whatever flags have been set on the object by copying over the global flags object. If the VM_ALLOCED flag was ste on the voicemail user prior to doing so, that flag is removed. This leaks the voicemail user when free_user is later called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277) Patch slightly modified for this commit. Review: https://reviewboard.asterisk.org/r/2096 ........ Merged revisions 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-05 07:37 +0000 [r372213-372240] Alec L Davis * main/dsp.c, /: dsp.c: Fix multiple issues when no-interdigit delay is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss detector to original -r349249 method with some changes, remove unnecessary; 1. reseting of hits=0, when no signal, only need to set it once. 2. incrementing of hits, when the hit is the same as the current hit. 3. setting of lasthit, when it's the same as before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3 spelling mistakes (closes issue ASTERISK-19610) alecdavis (license 585) Reported by: Jean-Philippe Lord Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2085/ ........ Merged revisions 372239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/dsp.c, /: dsp.c: optimize goerztzel sample loops, in dtmf_detect, mf_detect and tone_detect use a temporary short int when repeatedly used to call goertzel_sample. alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2093/ ........ Merged revisions 372212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-05 04:47 +0000 [r372198] Michael L. Young * res/res_rtp_asterisk.c, /: Fix Incrementing Sequence Number For Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in place to increment the sequence number for retransmitted DTMF end packets. With the introduction of the RTP engine API in 1.8, the sequence number was no longer being incremented. This patch fixes this regression as well as cleans up a few lines that were not doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh Bansal Tested by: Michael L. Young Patches: 01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license 6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2083/ ........ Merged revisions 372185 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-05 02:19 +0000 [r372165] Matthew Jordan * cel/cel_pgsql.c, /: Fix memory leak when CEL is successfully written to PostgreSQL database PQClear is not called when the result object of a call to PQExec has a status of PGRES_COMMAND_OK. Interestingly enough, the off nominal case was handled properly, so this memory leak only occurred when CEL records were successfully written. This patch properly clears the result in the nominal code path. (closes issue ASTERISK-19991) Reported by: Etienne Lessard Tested by: Etienne Lessard patches: mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license #6394) ........ Merged revisions 372158 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-30 20:53 +0000 [r372049-372090] Mark Michelson * /, apps/app_queue.c: Prevent crash on shutdown due to refcount error on queues container. When app_queue is unloaded, the queues container has its refcount decremented, potentially to 0. Then the taskprocessor responsible for handling device state changes is unreferenced. If the taskprocessor happens to be just about to run its task, then it will create and destroy an iterator on the queues container. This can cause the refcount on the queues container to increase to 1 and then back to 0. Going back to 0 a second time results in double frees. This failure was seen periodically in the testsuite when Asterisk would shut down. ........ Merged revisions 372089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Help prevent ringing queue members from being rung when ringinuse set to no. Queue member status would not always get updated properly when the member was called, thus resulting in the member getting multiple calls. With this change, we update the member's status at the time of calling, and we also check to make sure the member is still available to take the call before placing an outbound call. (closes issue ASTERISK-16115) reported by nik600 Patches: app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409) ........ Merged revisions 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-30 16:22 +0000 [r371962-372020] Matthew Jordan * /, channels/chan_iax2.c: AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers When an IAX2 call is made using the credentials of a peer defined in a dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are not applied to the call attempt. This allows for a remote attacker who is aware of a peer's credentials to bypass the ACL rules set for that peer. This patch ensures that the ACLs are applied for all peers, regardless of their storage mechanism. (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by: mjordan, Alan Frisch ........ Merged revisions 372015 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/manager.c, /, README-SERIOUSLY.bestpractices.txt: AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR The AMI Originate action can allow a remote user to specify information that can be used to execute shell commands on the system hosting Asterisk. This can result in an unwanted escalation of permissions, as the Originate action, which requires the "originate" class authorization, can be used to perform actions that would typically require the "system" class authorization. Previous attempts to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought to do so by inspecting the names of applications and functions passed in with the Originate action and, if those applications/functions matched a predefined set of values, rejecting the command if the user lacked the "system" class authorization. As noted by IBM X-Force Research, the "ExternalIVR" application is not listed in the predefined set of values. The solution for this particular vulnerability is to include the "ExternalIVR" application in the set of defined applications/functions that require "system" class authorization. Unfortunately, the approach of inspecting fields in the Originate action against known applications/functions has a significant flaw. The predefined set of values can be bypassed by creative use of the Originate action or by certain dialplan configurations, which is beyond the ability of Asterisk to analyze at run-time. Attempting to work around these scenarios would result in severely restricting the applications or functions and prevent their usage for legitimate means. As such, any additional security vulnerabilities, where an application/function that would normally require the "system" class authorization can be executed by users with the "originate" class authorization, will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has been updated to reflect that the AMI Originate action can result in commands requiring the "system" class authorization to be executed. Proper system configuration can limit the impact of such scenarios. (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM X-Force Research ........ Merged revisions 371998 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * doc/CODING-GUIDELINES (added), /: Restore CODING-GUIDELINES to doc folder In r294740, the CODING-GUIDELINES was removed from the doc folder in favor of the content on the Asterisk wiki. Some folks still look in the doc folder initially for coding guideline suggestions; as such, this patch adds a CODING-GUIDELINES file back into the doc folder. The content of the file merely points to the correct page on the Asterisk wiki where the coding guidelines currently live. (closes issue ASTERISK-20279) Reported by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985) ........ Merged revisions 371961 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-29 20:58 +0000 [r371920] Jonathan Rose * /, apps/app_meetme.c: app_meetme: Adding test events for following activity in MeetMe. ........ Merged revisions 371919 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-29 19:40 +0000 [r371861-371890] Richard Mudgett * main/channel.c, /: Initialize file descriptors for dummy channels to -1. Dummy channels usually aren't read from, but functions like SHELL and CURL use autoservice on the channel. (closes issue ASTERISK-20283) Reported by: Gareth Palmer Patches: svn-371580.patch (license #5169) patch uploaded by Gareth Palmer (modified) ........ Merged revisions 371888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_dial.c, /: Fix hangup cause passthrough regression. The v1.8 -r369258 change to fix the F and F(x) action logic introduced a regression in passing the hangup cause from the called channel to the caller channel. (closes issue ASTERISK-20287) Reported by: Konstantin Suvorov Patches: app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged revisions 371860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-29 17:07 +0000 [r371825] Jonathan Rose * /, channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout instead of 603 (closes issue ASTERISK-20124) Reported by: Walter Doekes ........ Merged revisions 371824 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-27 21:49 +0000 [r371748-371789] Mark Michelson * configs/agents.conf.sample, /: Fix misleading documentation in agents.conf.sample regarding ackcall usage. The documentation made it sound as if the DTMF acknowledgment was needed at the time the agent logs in, rather than when the agent is called. This is likely a relic from the days when there were multiple ways of logging in agents. (closes issue AST-962) reported by Steve Pitts ........ Merged revisions 371787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/manager.c, /: Fix incorrect documentation of the MailboxStatus manager command. The "Waiting" field was misdocumented as reporting the number of messages waiting. In reality, it simply indicated the presence or absence of waiting messages. ........ Merged revisions 371782 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configs/queues.conf.sample: Fix incorrectly documented option in queues.conf sharedlastcall defaults to "no" not "yes" (closes issue AST-979) reported by Steve Pitts ........ Merged revisions 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-27 16:43 +0000 [r371719] dlee : * main/lock.c, /: Fixes ast_rwlock_timed[rd|wr]lock for BSD and variants. The original implementations simply wrap pthread functions, which take absolute time as an argument. The spinlock version for systems without those functions treated the argument as a delta. This patch fixes the spinlock version to be consistent with the pthread version. (closes issue ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch uploaded by Egor Gorlin (license 6416) ........ Merged revisions 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-27 13:57 +0000 [r371691] Kinsey Moore * /, main/utils.c: Implement workaround for BETTER_BACKTRACES crash When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes crash when "core show locks" is run. This happens regularly in the testsuite since several tests run "core show locks" to help with debugging. This seems to be a fault with libraries on certain operating systems (notably CentOS 6.2/6.3) running on virtual machines and utilizing gcc 4.4.6. (closes issue ASTERISK-20090) ........ Merged revisions 371690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-26 23:06 +0000 [r371663] Alec L Davis * main/dsp.c, /: mf_detect: incorrectly used DTMF_GSIZE instead of MF_GSIZE ........ Merged revisions 371662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-21 20:40 +0000 [r371591] Mark Michelson * cdr/cdr_tds.c, main/xmldoc.c, apps/app_dial.c, channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c, main/file.c, main/utils.c, apps/app_queue.c, pbx/pbx_config.c, res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c, res/res_config_sqlite.c: Fix misuses of asprintf throughout the code. This fixes three main issues * Change asprintf() uses to ast_asprintf() so that it pairs properly with ast_free() and no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() fails, set the pointer NULL if it will be referenced later. * Fix some memory leaks that were spotted while taking care of the first two points. (Closes issue ASTERISK-20135) reported by Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 ........ Merged revisions 371590 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-20 15:27 +0000 [r371545] Kinsey Moore * main/udptl.c, /: Ignore recovered zero-length secondary UDPTL packets In some cases, recovering lost packets using the secondary packet recovery mechanism with UDPTL/T.38 can result in the recovery of zero-length packets. These must be ignored or the frame generated from them can cause segfaults and allocation failures. (closes issue ASTERISK-19762) (closes issue ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob Gagnon (rgagnon) ........ Merged revisions 371544 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-18 02:34 +0000 [r371491-371529] Matthew Jordan * main/http.c: Remove old debug code from http configuration loading (closes issue ASTERISK-20254) Reported by: Andrew Latham Patches: http.diff uploaded by Andrew Latham (license #5985) * main/xmldoc.c, /: Fix memory leak in XML documentation When formatting documentation fields, the XML documentation parser calls xmldoc_get_formatted. This function allocates a string buffer at the beginning of its routine. Unfortunately, on certain code paths, it also calls xmldoc_string_cleanup, which assumes that it will create the string buffer. The previously allocated string buffer is then leaked by the xmldoc_string_cleanup routine. Now: we don't do that. (closes issue AST-932) Reported by: Alexander Homig ........ Merged revisions 371469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-17 15:51 +0000 [r371437] Kinsey Moore * main/loader.c, /: Add instrumentation to subsystem reloads When Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now generate TestEvent AMI events on subsystem reloads such as cdr, dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-16 22:50 +0000 [r371398] Terry Wilson * /, main/config.c: Handle integer over/under-flow in ast_parse_args The strtol family of functions will return *_MIN/*_MAX on overflow. To detect when an overflow has happened, errno must be set to 0 before calling the function, then checked afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged revisions 371392 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-16 22:42 +0000 [r371394] Kinsey Moore * main/loader.c, /: Add module reload instrumentation for TEST_FRAMEWORK This adds AMI events for module reloads when Asterisk is built with TEST_FRAMEWORK enabled and corrects generation of the module load AMI event. (issue PQ-1126) ........ Merged revisions 371393 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-16 19:05 +0000 [r371338-371358] Jonathan Rose * /, channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING flag was used instead, which will frequently flip during reinvites. (closes issue AST-897) Reported by: Thomas Arimont ........ Merged revisions 371357 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK Under certain conditions, a SIP transaction involving directmedia wouldn't trigger a re-invite because the SDP answer was included in an ACK instead of in a message that we would have triggered the invite with. This patch just queues a source change control frame if the dialog is using directmedia when we find sdp for an ACK. (closes issue AST-913) Reported by: Thomas Arimont ........ Merged revisions 371337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-15 23:19 +0000 [r371313] Mark Michelson * /, apps/app_queue.c: Fix bug where final queue member would not be removed from memory. If a static queue had realtime members, then there could be a potential for those realtime members not to be properly deleted from memory. If the queue's members were loaded from realtime and then all the members were deleted from the backend, then the queue would still think these members existed. The reason was that there was a short- circuit in code such that if there were no members found in the backend, then the queue would not be updated to reflect this. Note that this only affected static queues with realtime members. Realtime queues with realtime members were unaffected by this issue. (closes issue ASTERISK-19793) reported by Marcus Haas ........ Merged revisions 371306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-15 20:15 +0000 [r371271] Kinsey Moore * /, channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog destruction The other instance of this bug was fixed by jcolp/file in r121496. If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed. (closes issue ASTERISK-20119) Patch-by: Misha Vodsedalek ........ Merged revisions 371270 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-15 01:43 +0000 [r371250-371251] Michael L. Young * channels/chan_sip.c: Reverting this change that was meant for branch 11. (issue ASTERISK-20221) * channels/chan_sip.c: Fix Segfault When Registering SIP Over WebSockets The helper function, get_address_family_filter, in chan_sip for dns resolution by address family was not recognizing the websockets transport and resulting in a null pointer being sent to functions in netsock2, in an attempt to determine if we are bound to ANY address ([::]) or not. This patch fixes this issue by handling the transport types SIP_TRANSPORT_WS and SIP_TRANSPORT_WSS which results in a sock address being set properly for use in determining the address family. (closes issue ASTERISK-20221) Reported by: Sven Beisiegel Tested by: Sven Beisiegel, James Mortensen Patches: asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young (license 5026) 2012-08-13 20:04 +0000 [r371203] Kinsey Moore * main/loader.c, /, apps/app_meetme.c: Add test instrumentation This adds test instrumentation for loading and unloading of modules and for certain actions in MeetMe to be used in the testsuite or any other consumer of AMI events. These will only be generated when Asterisk is built with TEST_FRAMEWORK enabled. (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-13 19:51 +0000 [r371199] Mark Michelson * /, channels/chan_sip.c: Fix problem where incorrect pointer was checked for nullity. ........ Merged revisions 371198 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-10 21:23 +0000 [r371142] Mark Michelson * /, apps/app_queue.c: Fix a couple of documentation problems in app_queue.c * The RemoveQueueMember app made mention of options that could be passed in, but no options are supported. I have removed the listing of options from the documentation. * The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value that could be set. (closes issue AST-949) reported by Steve Pitts (closes issue AST-954) reported by Steve Pitts ........ Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-10 16:46 +0000 [r371022-371090] Alexandr Anikin * addons/chan_ooh323.c, /: remove ALREADYGONE flag on ooh323 call data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone there really. This indication arrive from asterisk core not h.323 stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov Patches: ASTERISK-19308.patch ........ Merged revisions 371089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * addons/ooh323c/src/ooGkClient.c, /: Send re-register packets by GRQ (gatekeeper request) interval (close issue ASTERISK-20094) Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * addons/ooh323c/src/ooGkClient.c, /: Fix to resend GRQ/RRQ if RRJ (registration reject) is received (close issue ASTERISK-20094) Patches: ASTERISK-20094.patch ........ Merged revisions 371011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-09 19:11 +0000 [r371013] Richard Mudgett * channels/chan_dahdi.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c, channels/sig_ss7.c: Use better libss7 detection test and move libpri compile test. ........ Merged revisions 371012 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-09 18:05 +0000 [r370989] Alexandr Anikin * /, addons/ooh323c/src/ooh323ep.c: change opening h323 logfile with append mode instead of overwrite ........ Merged revisions 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-09 17:39 +0000 [r370986] Kinsey Moore * /, apps/app_meetme.c: Correct documentation for the MeetMe x flag The documentation for the x flag for MeetMe incorrectly described its function as closing down the conference when the last marked user left. It actually causes the users with that flag to leave the conference when the last marked user exits. The functionality of this flag is not changing. ........ Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-08 22:42 +0000 [r370954] Michael L. Young * /, apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When a channel hangs up while being spied upon and the option to exit the ChanSpy application when the spied on channel hangs up is set, ast_autochan_destroy is not being called and therefore a reference to the spied upon channel is not removed. The symptom being reported was that when using func_group in the dialplan and calling "group show channels" at the cli, the spied upon channel was still being shown while "core show channels" showed that the channel was not up. This patch calls ast_autochan_destroy when a spied upon channel hangs up and the option to exit the ChanSpy application is set, removing the reference to the channel allowing the count for the group that the spied channel was part of to be decremented. (closes issue ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-08 20:29 +0000 [r370924] Kinsey Moore * main/channel.c, /: Do not define a cause that doesn't actually exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause information. As such, it should not be defined and translatable as a cause. ........ Merged revisions 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-08 20:04 +0000 [r370901] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Fix the analog dial *0 flash-hook of bridged peer feature. The flash-hook the bridged peer feature now correctly determines if the bridged peer is another chan_dahdi channel, that it is an analog channel, and that it has the correct signaling for an FXO port. It now also flash-hooks the correct channel. ........ Merged revisions 370900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-07 19:21 +0000 [r370858] Kinsey Moore * main/channel.c, /: Add missing AST_CAUSE_* -> text translations ........ Merged revisions 370856 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-06 15:02 +0000 [r370798] Mark Michelson * /, channels/chan_sip.c: Improve debug message for temporary outbound proxies. Thanks to Paul Belanger for pointing this out. ........ Merged revisions 370797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-03 21:50 +0000 [r370772] Mark Michelson * /, channels/chan_sip.c, channels/sip/config_parser.c, channels/sip/include/sip.h: Multiple revisions 370769-370771 ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri, 03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a SIP dialstring. This is based on the review request posted by Walter Doekes (referenced lower in the commit message) The main fix here is to treat the IPorHost portion of the dial string as a temporary outbound proxy. This ensures requests get sent to the proper location. Due to the age of the request, some parts were no longer relevant. For instance, the request moved outbound proxy parsing code into a single method. This is done in a previous commit, so it was not necessary to do again. Also, the review request fixed some errors with regards to request routing for CANCEL and ACK requests. This has also been fixed in more recent commits. (closes issue ASTERISK-19677) reported by Walter Doekes Review https://reviewboard.asterisk.org/r/1859 ........ r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug 2012) | 3 lines Remove unused variable. ........ r370771 | mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5 lines Seriously? Another compilation error fixed. Somebody beat me. ........ Merged revisions 370769-370771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-08-01 02:26 +0000 [r370698] Kinsey Moore * /, utils/extconf.c: Revert alloca changes for utils These changes were a tad overzealous in the utils directory. Unfortunately, these don't compile with a "make". ........ Merged revisions 370697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-31 21:19 +0000 [r370672] Matthew Jordan * /, channels/chan_sip.c: Schedule pokes of registered SIP peers within a given timespan after SIP reload With a large number of SIP peers registered, performing a SIP reload causes a flood of SIP OPTIONS request packets. These are immediately sent out, and, as responses come back, can cause peers to be flagged as 'lagged' due to handling of the many response messages. This fix prevents this "packet storm" and schedules the pokes for a random time. That time varies between 1 ms and the peer's qualify time, or, if the qualify time is unknown, the global qualifyfreq setting. The committed patch has some very small modifications to the patch schmidts wrote for the review. (closes issue ASTERISK-19154) Reported by: Nicolo Mazzon patches: issue19154.patch license #6034 uploaded by schmidts Review: https://reviewboard.asterisk.org/r/1652 ........ Merged revisions 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-31 19:57 +0000 [r370643] Kinsey Moore * apps/app_while.c, include/asterisk/utils.h, main/pbx.c, res/res_config_pgsql.c, channels/chan_sip.c, apps/app_festival.c, pbx/pbx_lua.c, funcs/func_cut.c, tests/test_linkedlists.c, apps/app_getcpeid.c, funcs/func_global.c, channels/chan_jingle.c, main/tcptls.c, funcs/func_channel.c, apps/app_directed_pickup.c, main/callerid.c, main/file.c, apps/app_macro.c, main/astmm.c, apps/app_sms.c, main/event.c, pbx/pbx_dundi.c, include/asterisk/strings.h, utils/extconf.c, apps/app_mixmonitor.c, main/asterisk.c, main/dsp.c, addons/res_config_mysql.c, apps/app_voicemail.c, addons/app_mysql.c, /, apps/app_meetme.c, apps/app_dictate.c, main/say.c, main/threadstorage.c, funcs/func_strings.c, main/utils.c, funcs/func_logic.c, channels/chan_gtalk.c, cdr/cdr_pgsql.c, channels/chan_iax2.c, res/res_jabber.c, main/config.c, main/channel.c, res/ael/pval.c, apps/app_osplookup.c, main/manager.c, pbx/pbx_spool.c, main/strcompat.c, apps/app_minivm.c, main/features.c, res/res_agi.c, main/http.c, main/logger.c, pbx/pbx_ael.c, main/app.c, channels/chan_alsa.c, pbx/pbx_realtime.c, addons/chan_mobile.c, main/db.c: Clean up and ensure proper usage of alloca() This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-31 15:31 +0000 [r370619] Mark Michelson * /, channels/chan_sip.c, configs/sip.conf.sample, channels/sip/include/sip.h: Help mitigate potential reinvite glare scenarios. When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. For those who are having some deja vu, that's because this patch was originally committed to trunk since there is a new configuration option added. After seeing a bug report about audio being slow to set up on SIP calls, it became apparent that this patch would be the best solution for resolving the issue. The patch is unintrusive and will have no effect unless the option is explicitly enabled. (closes issue AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857) reported by Matt Jordan ........ Merged revisions 370618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-09-13 Asterisk Development Team * Asterisk 10.8.0 Released. 2012-09-11 Asterisk Development Team * Asterisk 10.8.0-rc2 Released. * AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers * AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR * r371861: Fix hangup cause passthrough regression. The v1.8 -r369258 change to fix the F and F(x) action logic introduced a regression in passing the hangup cause from the called channel to the caller channel. (closes issue ASTERISK-20287) Reported by: Konstantin Suvorov Patches: app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified) Tested by: rmudgett * r372710: Only re-create an SRTP session when needed; respond with correct crypto policy In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an SDP offer and the ability to re-create an SRTP session when the crypto keys changed. In certain circumstances - most notably when a phone is put on hold after having been bridged for a significant amount of time - the act of re-creating the SRTP session causes problems for certain models of phones. The patch committed in r356604 always re-created the SRTP session regardless of whether or not the cryptographic keys changed. Since this is technically not necessary, this patch modifies the behavior to only re-create the SRTP session if Asterisk detects that the remote key has changed. This allows models of phones that do not handle the SRTP session changing to continue to work, while also providing the behavior needed for those phones that do re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported by: Nicolo Mazzon Tested by: Nicolo Mazzon Review: https://reviewboard.asterisk.org/r/2099 * r372841: Fix bad channel application data reference. When channels get bridged due to an AMI bridge action or a DTMF attended transfer, the two channels that get bridged have their application data pointing to the other channel's name. This means that if one channel is hung up but the other moves on, it means that the channel that moves on will have its application data pointing at freed memory. (issue ASTERISK-20335) 2012-07-31 Asterisk Development Team * Asterisk 10.8.0-rc1 Released. 2012-07-30 16:49 +0000 [r370564] Richard Mudgett * channels/chan_misdn.c, /: Release B channel allocation on error path in chan_misdn. ........ Merged revisions 370563 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-30 14:50 +0000 [r370547] Jonathan Rose * apps/app_meetme.c: app_meetme: Change app_meetme support level to extended from deprecated (closes issue ASTERISK-20134) Reported by: Leif Madsen 2012-07-25 21:12 +0000 [r370495] Jonathan Rose * /, res/res_agi.c: res_agi: Add message indicating need for \n character in verbose message The while loop responsible for reading AGI messages from a fastAGI service can end up looping indefinitely when an AGI script fails to indicate the end of a message with a \n character. This patch adds an indication that we are expecting a \n character to end the message to make it more clear to users that this is necessary if they are receiving this warning over and over. (issue ASTERISK-20061) Reported by: Eike Kuiper ........ Merged revisions 370494 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-24 17:08 +0000 [r370432] Tzafrir Cohen * /, channels/chan_oss.c: chan_oss: fix "sample rate" error message Merged revisions 370428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-24 16:54 +0000 [r370361-370430] Kevin P. Fleming * main/frame.c, /: Rewrite a comment that didn't adequately explain the code it was documenting. ........ Merged revisions 370429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, funcs/func_shell.c: Improve documentation for the SHELL() dialplan function. ........ Merged revisions 370383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/channel.c, /: Free any datastores attached to dummy channels. Revision 370205 added the use of a datastore attached to a dummy channel to resolve a memory leak, but ast_dummy_channel_destructor() in this branch did not free datastores, resulting in a continued (but slightly smaller) memory leak. This patch backports the change to free said datastores from the Asterisk trunk. (related to issue AST-916) ........ Merged revisions 370360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-19 22:11 +0000 [r370277] Richard Mudgett * /, main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has problems casting away constness. ........ Merged revisions 370275 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-19 22:01 +0000 [r370271-370274] Matthew Jordan * /, main/cel.c: Fix compilation error when MALLOC_DEBUG is enabled To fix a memory leak in CEL, a channel datastore was introduced whose destruction function pointer was pointed to the ast_free macro. Without MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free. With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a different place then utils.h, and became undefined. This patch resolves this by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be ast_free, which is defined to be free. (issue AST-916) Reported by: Thomas Arimont ........ Merged revisions 370273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * res/res_rtp_asterisk.c, /: Handle extremely out of order RFC 2833 DTMF The current implementation of RFC 2833 DTMF handling in res_rtp_asterisk will, if a packet arrives out of order, drop the packet. This is to prevent duplicate ton generation in the Asterisk core. Since the RTP layer does not buffer data itself, this is the only option the RTP layer currently has for handling packets that arrive out of order. For the most part, this doesn't matter. For a particular digit, so long as a BEGIN packet arrives before the first END packet, the digit will be produced. If subsequent BEGIN packets arrive interleaved with the ENDs, they will be dropped; likewise, if the BEGIN or END packets themselves are out of order, those packets are dropped but sufficient information is conveyed to the Asterisk core to produce the appropriate digit. For certain sequences of DTMF packets - most notably when, for a particular digit, an END packet arrives before any BEGIN packet for that digit - this is a real problem. When an END arrives before any BEGINs, the END packet is dropped - but at the same time, it causes subsequent BEGIN packets for that digit to be ignored. When the next in order END packet arrives, it too is dropped - Asterisk believes that there was no initial BEGIN. The solution this patch provides is to trust the END packet to convey the information needed for the Asterisk core to produce the DTMF digit. If we receive an END packet, and it: * Has a timestamp greater then the last timestamp received from an END packet * Does not have the same sequence number as the last received sequence number (and is thus not an END packet retransmission) Then we send the END frame up to the Asterisk core. It contains enough DTMF information for Asterisk to produce the digit. On the other hand, if we receive a BEGIN or continuation packet that occurs with a timestamp equal to or less then the last END timestamp, then we've received something out of order - but we already have received enough information to produce the digit. These packets are dropped. Much thanks goes to Olle Johansson (oej) for providing the idea for this solution. Review: https://reviewboard.asterisk.org/r/2033/ (issue ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt Jordan ........ Merged revisions 370252 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-18 19:14 +0000 [r370184-370206] Kevin P. Fleming * /, main/cel.c: Resolve severe memory leak in CEL logging modules. A customer reported a significant memory leak using Asterisk 1.8. They have tracked it down to ast_cel_fabricate_channel_from_event() in main/cel.c, which is called by both in-tree CEL logging modules (cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event that they log. The cause was an incorrect assumption about how data attached to an ast_channel would be handled when the channel is destroyed; the data is now stored in a datastore attached to the channel, which is destroyed along with the channel at the proper time. (closes issue AST-916) Review: https://reviewboard.asterisk.org/r/2053/ ........ Merged revisions 370205 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/channel.c, addons/app_mysql.c, main/pbx.c, funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c, funcs/func_lock.c, apps/app_macro.c, channels/chan_iax2.c, apps/app_mixmonitor.c, apps/app_stack.c, funcs/func_global.c, res/res_odbc.c: Ensure that all ast_datastore_info structures are 'const'. While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-16 19:52 +0000 [r370132] Walter Doekes * /, channels/chan_sip.c: Code cleanup and bugfix in chan_sip outboundproxy parsing. The bug was clearing the global outboundproxy when a peer-specific outboundproxy was bad. The cleanup reduces duplicate code. Review: https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark Michelson ........ Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-16 13:51 +0000 [r370082] Kinsey Moore * /, UPGRADE.txt, CHANGES, UPGRADE-1.8.txt: Add comments about the BUILD_NATIVE change This is a significant change and mention of it should have gone into UPGRADE.txt and CHANGES. ........ Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-12 20:20 +0000 [r370025] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /: Add missing ast_hangup() calls on some analog exception paths. Make starting analog_ss_thread() or __analog_ss_thread() failure paths hangup the channel. ........ Merged revisions 370017 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-12 20:05 +0000 [r369994-370015] Kinsey Moore * /, channels/chan_sip.c: Include Expires header for SIP PUBLISH requests RFC3903 requres SIP PUBLISH requests to have Expires headers, so add them. Review: https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth ........ Merged revisions 370014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Prevent double uri_escaping in chan_sip when pedantic is enabled If pedantic mode is enabled, outbound invites will have double-escaped contacts. This avoids setting an already-escaped string into a field where it is expected to be unescaped. (closes issue ASTERISK-20023) Reported-by: Walter Doekes ........ Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-12 14:25 +0000 [r369971] Michael L. Young * /, funcs/func_math.c: Correct Documentation For DEC Function The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) ........ Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-11 17:12 +0000 [r369938] Tilghman Lesher * funcs/func_realtime.c, /: Allow the REALTIME() function to report errors back to the caller. Also, do more error checking on the arguments specified to the REALTIME() function and clarify the documentation. While I was editing the file, a few coding guidelines fixups, as well. Review: https://reviewboard.asterisk.org/r/2031/ ........ Merged revisions 369937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-30 Asterisk Development Team * Asterisk 10.7.0 Released. 2012-07-11 Asterisk Development Team * Asterisk 10.7.0-rc1 Released. 2012-07-10 13:35 +0000 [r369871] Kinsey Moore * main/pbx.c, /, apps/app_stack.c: Improve Goto and GotoIf related documentation Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged revisions 369869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-09 17:06 +0000 [r369819] Jason Parker * configs/sip_notify.conf.sample, /: Add Digium phones context to sip_notify sample config. This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) ........ Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-09 14:43 +0000 [r369793] Jonathan Rose * /, channels/chan_sip.c: chan_sip: Fix small behavioral change accidentally introduced in r369750 When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed the return value, which would likely make the indication not be sent in audio. This fixes that while still removing the warning message. ........ Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-06 21:02 +0000 [r369751] Jonathan Rose * /, channels/chan_sip.c: chan_sip: Add case for FLASH control frames so that we don't display a warning. chan_sip channels can receive flash control frames when connected to analog phones and possibly for other reasons. There really isn't a reason to warn when these frames are received, we can safely ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-06 18:47 +0000 [r369709-369732] Mark Michelson * main/tcptls.c, /: Remove a superfluous and dangerous freeing of an SSL_CTX. The problem here is that multiple server sessions share a SSL_CTX. When one session ended, the SSL_CTX would be freed and set NULL, leaving the other sessions unable to function. The code being removed is superfluous because the SSL_CTX structures for servers will be properly freed when ast_ssl_teardown is called. (closes issue ASTERISK-20074) Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license #5049) Testers: Trevor Helmsley ........ Merged revisions 369731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/bridging.c: Fix bridging thread leak. The bridge thread was exiting but was never being reaped using pthread_join(). This has been fixed now by calling pthread_join() in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012 ........ Merged revisions 369708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-05 19:12 +0000 [r369653] Kinsey Moore * apps/app_voicemail.c, /: AST-2012-011: Resolve heap corruption issue with voicemail The heard and deleted arrays in the voicemail state structure were not handled properly following the memory leak fix in r354890 and a fix for an invalid free in r356797. This could result in accessing and writing into freed memory. The allocation for these arrays has been reworked to avoid the possibility of invalid frees, access of freed memory, and crashes that were occurring as a result of this. Locking around accesses and modifications of the voicemail state structure members dh_arraysize, heard, and deleted has been added to prevent simultaneous modification and access when IMAP storage is in use. If IMAP storage is not in use, this locking is not compiled in. Review: https://reviewboard.asterisk.org/r/1994/ (closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license 6273) ........ Merged revisions 369652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-05 17:02 +0000 [r369627] Matthew Jordan * /, channels/chan_sip.c: Do not send a BYE when a provisional response arrives during a re-INVITE Commits r369557 and r369579 were done to improve handling of re-INVITEs when the UA that was supposed to receive the re-INVITE fails to respond. A limitation of those patches occurred when a UA sent a provisional response to the re-INVITE. This triggered a sending of a BYE in check_pending. This patch tweaks the handling of the re-INVITE such that a BYE is not sent in response to those messages. (issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies patches: (reinvite_tweak.diff license #5012 by Steve Davies) ........ Merged revisions 369626 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-03 17:02 +0000 [r369558-369580] Terry Wilson * /, channels/chan_sip.c: More improvements to re-INVITEs timing out after a provisional response There is no need to call check_pendings() on a final response to an INVITE when destroying the scheduler entry as it will be done later during normal processing. (issue ASTERISK-19992) ........ Merged revisions 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c, channels/sip/include/sip.h: Better handle re-INVITEs with provisional but no final repsonses A previous attempt at fixing this issue had negative side effects related to attended transfers which this patch should resolve. Many thanks to Steve Davies for all of the good suggestions and testing. (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson Review: https://reviewboard.asterisk.org/r/2009/ ........ Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-29 20:28 +0000 [r369511] Mark Michelson * main/rtp_engine.c: Fix apparent copy and paste error where incorrect "glue" is used. 2012-06-29 16:54 +0000 [r369472-369491] Joshua Colp * /, channels/chan_sip.c: With some configurations a transport is not actually specified so assume UDP in these cases. ........ Merged revisions 369490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Make the address family filter specific to the transport. (closes issue ASTERISK-16618) Reported by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........ Merged revisions 369471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-27 21:10 +0000 [r369437] Terry Wilson * /, channels/chan_sip.c, channels/sip/include/sip.h: AST-2012-010: Clean up after a reinvite that never gets a final response The basic problem is that if a re-INVITE is sent by Asterisk and it receives a provisional response, but no final response, then the dialog is never torn down. In addition to leaking memory, this also leaks file descriptors and will eventually lead to Asterisk no longer being able to process calls. This patch just keeps track of whether there is an outstanding re-INVITE, and if there is goes ahead and cleans up everything as though there was no outstanding reinvite. Review: https://reviewboard.asterisk.org/r/2009/ (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson ........ Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-26 13:22 +0000 [r369369-369391] Matthew Jordan * /, main/adsi.c: Fix crash in unloading of res_adsi module When res_adsi is unloaded, it removes the ADSI functions that it previously installed by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing it to check whether or not the version of the functions matches what the core was expecting it. This patch makes it so that the version is only checked if a potentially valid adsi_funcs pointer was passed in. Passing in NULL removes the installed functions, bypassing the version check. ........ Merged revisions 369390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/cdr.c, /: Fix incorrect duration reporting in CDRs created in batch mode Certain places in core/cdr.c would, if the duration value were 0, calculate the duration as being the delta between the current time and the time at which the CDR record was started. While this does not typically cause a problem in non-batch mode, this can cause an issue in batch mode where CDR records are gathered and written long after those calls have ended. In particular, this affects calls that were never answered, as those are expected to have a duration of 0. Often, this would result in CDR logs with a significant number of calls with lengthy durations, but dispositions of "BUSY". Note that this does not affect cdr_csv, as that backend does not use ast_cdr_getvar and instead directly reports the duration value. The affected core backends include cdr_apative_odbc and cdr_custom; other extended or deprecated CDR backends may potentially still directly manipulate the duration values. (issue ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883) Reported by: Thomas Arimont Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1996/ ........ Merged revisions 369351 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-25 19:16 +0000 [r369353] Mark Michelson * /, channels/chan_sip.c, channels/sip/include/sip.h: Re-fix how local tag is generated when sending a 481 to an INVITE. Match our local tag to whatever to-tag was sent in the initial INVITE. Because the size of the to-tag may not fit in the buffer in the sip_pvt, it has been changed to a string field. (closes issue ASTERISK-19892) reported by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977 ........ Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-25 15:59 +0000 [r369328] Richard Mudgett * /, main/features.c: Fix Bridge application occasionally returning to the wrong location. * Fix do_bridge_masquerade() getting the resume location from the zombie channel. The code must not touch a clone channel after it has masqueraded it. The clone channel has become a zombie and is starting to hangup. (closes issue ASTERISK-19985) Reported by: jamicque Patches: jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: jamicque ........ Merged revisions 369327 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-25 15:52 +0000 [r369303-369325] Mark Michelson * include/asterisk/adsi.h, /, main/Makefile, res/res_adsi.c, main/adsi.c (added), res/res_adsi.exports.in (removed): Multiple revisions 369323-369324 ........ r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines Eliminate embedding of res_adsi.so module. The way this is done is to stop using the optional API. Instead, res_adsi.so, when loaded fills in a table of function pointers. Review: https://reviewboard.asterisk.org/r/1991 ........ r369324 | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2 lines Forgot to svn add this file in my last commit. ........ Merged revisions 369323-369324 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Be more consistent with the return code for requests received from invalid domain. When Asterisk receives an INVITE from an external domain when allowexternaldomains=no send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving a REGISTER in this situation. (Closes issue ASTERISK-19601) Reported by Matthew Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049) ........ Merged revisions 369302 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-23 00:12 +0000 [r369236-369283] Richard Mudgett * /, main/features.c: Fix Bridge application and AMI Bridge action error handling. * Fix AMI Bridge action disconnecting the AMI link on error. * Fix AMI Bridge action and Bridge application not checking if their masquerades were successful. * Fix Bridge application running the h-exten when it should not. * Made do_bridge_masquerade() return if the masquerade was successful so the Bridge application and AMI Bridge action could deal with it correctly. * Made bridge_call_thread_launch() hangup the passed in channels if the bridge_call_thread fails to start. Those channels would have been orphaned. * Made builtin_atxfer() check the success of the transfer masquerade setup. ........ Merged revisions 369282 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Explicitly check caller hangup in app Queue rather than a polluted res2 value. ........ Merged revisions 369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_dial.c, /: Check if PBX was started and fix F and F(x) action logic in Dial application. ........ Merged revisions 369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/ccss.c: Check if PBX was started for generic CCSS recall. ........ Merged revisions 369238 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Change incorrect chan_sip zombie hangup debug message. They are all zombies now. ........ Merged revisions 369235 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-22 19:34 +0000 [r369215] Terry Wilson * /, channels/chan_sip.c: Don't crash on a guest directmedia call A sip_pvt may not have relatedpeer set if a call doesn't match up with a peer. If there is no relatedpeer, there is no direct media ACL to apply, so just return that it is allowed. (closes issue ASTERISK-20040) Reported by: Terry Wilson ........ Merged revisions 369214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-22 17:23 +0000 [r369206] Kinsey Moore * /, channels/chan_sip.c: Don't parse media stream state for SIP video streams The sendonly/recvonly/sendrecv/inactive media stream attributes were parsed for video, but nothing was ever done with them. With this code removed, an UNSUPPORTED message is produced when these attributes are used in conjunction with a video stream which is the better behavior since they were never really supported in the first place. ........ Merged revisions 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-20 17:36 +0000 [r369147] Alexandr Anikin * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, /: fix locking issue on empty callList (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch ........ Merged revisions 369146 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-20 02:04 +0000 [r369109] Michael L. Young * main/netsock2.c, /, include/asterisk/netsock2.h: Fix NULL pointer segfault in ast_sockaddr_parse() While working with ast_parse_arg() to perform a validity check, a segfault occurred. The segfault occurred due to passing a NULL pointer to ast_sockaddr_parse() from ast_parse_arg(). According to the documentation in config.h, "result pointer to the result. NULL is valid here, and can be used to perform only the validity checks." This patch fixes the segfault by checking for a NULL pointer. This patch also adds documentation to netsock2.h about why it is necessary to check for a NULL pointer. (Closes issue ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1990/ ........ Merged revisions 369108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-19 23:32 +0000 [r369091] Alexandr Anikin * addons/chan_ooh323.c: check rtptimeouts in ooh323 channels as per config file (rtp voice, video, udptl except rtcp) (closes issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches: 19179-ooh323-ast10.patch 2012-06-19 15:37 +0000 [r369067] Mark Michelson * /, channels/chan_sip.c: Fix request routing issue when outboundproxy is used. Asterisk was incorrectly setting the destination of CANCELs and ACKs for error responses to the URI of the initial INVITE. This resulted in further requests, such as INVITEs with authentication credentials, to be routed incorrectly. Instead, when these CANCEL or ACKs are to be sent, we should simply keep the destination the same as what it previously was. There is no need to alter it any. (closes issue ASTERISK-20008) Reported by Marcus Hunger Patches: ASTERISK-20008.patch uploaded by Mark Michelson (license #5049) ........ Merged revisions 369066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-18 18:11 +0000 [r369044] Richard Mudgett * /, main/features.c: Fix monitoring calls put in a parking lot. * Fix a regression that was introduced by -r366167 which effectively disabled monitoring parked calls. (closes issue ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett ........ Merged revisions 369043 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-15 16:07 +0000 [r369005] Kevin P. Fleming * channels/sip/sdp_crypto.c, main/slinfactory.c, main/translate.c, main/jitterbuf.c, main/acl.c, channels/iax2-provision.c, channels/sig_pri.c, utils/astdb2bdb.c, main/chanvars.c, main/data.c, main/hashtab.c, channels/chan_misdn.c, main/abstract_jb.c, main/fixedjitterbuf.c, channels/sip/dialplan_functions.c, main/test.c, res/snmp/agent.c, main/event.c, main/astmm.c, channels/sip/config_parser.c, channels/vgrabbers.c, main/alaw.c, main/asterisk.c, main/dsp.c, main/timing.c, main/udptl.c, main/autoservice.c, main/fskmodem_float.c, main/frame.c, main/security_events.c, main/ccss.c, main/threadstorage.c, main/say.c, channels/console_video.c, channels/sip/reqresp_parser.c, main/devicestate.c, main/astfd.c, main/ssl.c, main/taskprocessor.c, main/autochan.c, channels/misdn/isdn_lib.c, main/enum.c, main/format_pref.c, main/astobj2.c, main/indications.c, main/fskmodem.c, channels/misdn_config.c, apps/confbridge/conf_config_parser.c, main/io.c, main/cli.c, main/ulaw.c, main/dial.c, main/framehook.c, main/format_cap.c, main/strcompat.c, main/heap.c, channels/misdn/ie.c, main/plc.c, main/logger.c, main/stdtime/localtime.c, channels/sig_ss7.c, main/sched.c, main/datastore.c, main/lock.c, main/strings.c, main/pbx.c, main/stun.c, channels/sip/srtp.c, main/dnsmgr.c, channels/vcodecs.c, channels/sip/security_events.c, utils/astdb2sqlite3.c, main/aoc.c, pbx/dundi-parser.c, main/cel.c, channels/iax2-parser.c, build_tools/find_missing_support_level (added), main/netsock.c, main/tcptls.c, main/srv.c, main/privacy.c, main/callerid.c, main/file.c, channels/misdn/portinfo.c, main/audiohook.c, main/xmldoc.c, main/netsock2.c, main/format.c, main/global_datastores.c, main/rtp_engine.c, /, res/ais/clm.c, main/utils.c, channels/misdn/isdn_msg_parser.c, main/xml.c, main/config.c, main/loader.c, main/term.c, main/channel.c, main/cdr.c, res/ael/pval.c, channels/sig_analog.c, main/tdd.c, channels/console_gui.c, res/ais/evt.c, main/fskmodem_int.c, channels/console_board.c, main/syslog.c, main/app.c, main/image.c, main/dns.c, main/message.c, main/db.c, main/bridging.c: Multiple revisions 369001-369002 ........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines Add support-level indications to many more source files. Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-14 17:31 +0000 [r368947] Matthew Jordan * channels/chan_skinny.c: AST-2012-009: Fix crash in chan_skinny due to Key Pad Button Message handling AST-2012-008 (r367844) fixed a denial of service attack exploitable in the Skinny channel driver that occurred when certain messages are sent after a previously registered station sends an Off Hook message. Unresolved in that patch is an issue in the Asterisk 10 releases, wherein, if a Station Key Pad Button Message is processed after an Off Hook message, the channel driver will inappropriately dereference a NULL pointer. This patch fixes those places where the message handling or the channel callback functions would attempt to dereference the line's pointer to the device. (issue ASTERISK-19905) Reported by: Christoph Hebeisen Tested by: mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff uploaded by mjordan (license 6283) 2012-06-14 15:25 +0000 [r368899-368928] Mark Michelson * /, main/Makefile: Revert Makefile change to remove embedding res_adsi.so The change has resulted in a linking error for certain versions of GCC. This is much worse than the original issue, so for now, temporarily revert the change. A more thorough change will be sought out. ........ Merged revisions 368927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, funcs/func_volume.c: Fix a deadlock that occurs when func_volume is used on a local channel. This was discovered by trying to perform a call forward to an extension that makes use of func_volume. When the local channel is optimized away, the datastore on the local;2 channel would have its audiohook destroyed rather than detaching the audiohook from the channel and then destroying it. With this patch, func_volume's datastore destructor takes the proper route of detaching the audiohook and then destroying it. (closes issue ASTERISK-19611) reported by Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark Michelson (license #5049) ........ Merged revisions 368898 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-13 20:27 +0000 [r368895] Matthew Jordan * res/res_smdi.c, /, res/res_adsi.c: Mark res_smdi/res_adsi as 'core' supported modules Recently, various issues surrounding weak symbols have caused problems with modules that rely on that feature to be enabled in menuselect. This includes app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi, which, in certain circumstances, may not be enabled by default in menuselect. Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this patch marks both as 'core' supported modules. This will allow both app_voicemail and chan_dahdi to be enabled as well, regardless of whether or not that system supports weak symbols. (issue AST-900) Reported by: Thomas Arimont (issue AST-885) Reported by: Denis Alberto Martinez ........ Merged revisions 368894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-13 19:36 +0000 [r368885] Mark Michelson * /, main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+ the result is that Asterisk has a phantom module loaded at startup, claiming to be res_adsi. (closes issue ASTERISK-19920) reported by Leif Madsen ........ Merged revisions 368873 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-13 14:30 +0000 [r368831-368853] Matthew Jordan * Makefile, /: Do not install empty directories; add ASTLIBDIR r368830 modified the installation script to only create a directory if that directory does not exist. If some directory variable was empty, it would attempt to create the empty location. It also failed to create the ASTLIBDIR directory. This patch fixes it such that the correct directories are made and only created if a value specifying them actually exists. ........ Merged revisions 368852 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * Makefile, /: Do not perform install on existing directories If a directory already exists, performing a 'make install' will remove the permissions associated with the current directory and replace them with the permissions of the user executing the install. This patch changes this behavior to only perform an install on the directory if the directory does not exist. Thus, if a user later changes the permissions on that directory, those permissions will be preserved in subsequent installs. Review: https://reviewboard.asterisk.org/r/1986 Review: https://reviewboard.asterisk.org/r/1864 (closes issue ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger, Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded by mjordan) ........ Merged revisions 368830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-12 15:37 +0000 [r368808] Mark Michelson * /, channels/chan_sip.c: Set the Caller ID "tag" on peers even if remote party information is present. On incoming calls, we were setting the cid_tag on the dialog only if there was no remote party information (Remote-Party-ID or P-Asserted-Identity) present. The Caller ID tag is an invented parameter, though, and should be set no matter the circumstance. (closes issue ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884) Reported by Trey Blancher ........ Merged revisions 368807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-11 17:08 +0000 [r368760] Richard Mudgett * main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c, /, channels/chan_sip.c, include/asterisk/channel.h, channels/chan_iax2.c: Fix deadlock potential with ast_set_hangupsource() calls. Calling ast_set_hangupsource() with the channel lock held can result in a deadlock because the function also locks the bridged channel. (issue ASTERISK-19537) (closes issue AST-891) Reported by: Guenther Kelleter Tested by: Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec Davis ........ Merged revisions 368759 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-11 15:15 +0000 [r368721-368739] Kinsey Moore * apps/app_voicemail.c, main/udptl.c, channels/sip/sdp_crypto.c, /, channels/chan_sip.c, main/say.c, res/res_fax.c, funcs/func_strings.c, channels/sip/reqresp_parser.c, apps/app_queue.c, main/loader.c, channels/chan_dahdi.c, res/res_config_odbc.c, channels/sip/dialplan_functions.c, pbx/pbx_config.c, apps/app_directory.c, res/res_speech.c, res/res_odbc.c: Fix coverity UNUSED_VALUE findings in core support level files Most of these were just saving returned values without using them and in some cases the variable being saved to could be removed as well. (issue ASTERISK-19672) ........ Merged revisions 368738 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/md5.c: Fix compilation in dev-mode Backport a compilation fix in md5.c from trunk that only showed up in dev-mode under certain compiler versions. ........ Merged revisions 368719 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-07-10 Asterisk Development Team * Asterisk 10.6.0 Released. 2012-07-06 Asterisk Development Team * Asterisk 10.6.0-rc2 Released. * AST-2012-009: Skinny Channel Driver Remote Crash Vulnerability * AST-2012-010: Possible Resource Leak on Uncompleted Re-INVITE Transactions * AST-2012-011: Remote Crash Vulnerability in VoiceMail Application * Fix crash on a guest directmedia call A sip_pvt may not have relatedpeer set if a call doesn't match up with a peer. If there is no relatedpeer, there is no direct media ACL to apply, so just return that is is allowed. (closes issue ASTERISK-20040) * Fix request routing issue when outboundproxy is used Asterisk was incorrectly setting the destination of CANCELs and ACKs for error responses to the URI of the initial INVITE. This resulted in further requests, such as INVITEs with authentication credentials, to be routed incorrectly. Instead when these CANCEL or ACKs are to be esnt, we should simply keep the destination the same as what it previously was. There is no need to alter it any. (closes issue ASTERISK-20008) * Fix monitoring calls put in a parking lot Fix a regression that was introduced by r366167 which effectively disabled monitoring parked calls. (closes issue ASTERISK-20012) 2012-06-08 Asterisk Development Team * Asterisk 10.6.0-rc1 Released. 2012-06-06 21:32 +0000 [r368645] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /: Fix POTS flash hook to orignate a second call deadlock. A deadlock can occur when a POTS phone tries to flash hook to originate a second call for 3-way or transfer. If another process is scanning the channels container when the POTS line flash hooks then a deadlock will occur. * Release the channel and private locks when creating a new channel as a result of a flash hook. (closes issue ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 368644 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-06 19:18 +0000 [r368629] Mark Michelson * /, channels/chan_sip.c: Fix a specific scenario where ACKs are not matched. If a dialog-starting INVITE contains a to-tag, then Asterisk will respond with a 481. In this case, the resulting incoming ACK would not be matched, so Asterisk would continue retransmitting the 481 until the transaction times out. There were two issues. Asterisk, upon creating a sip_pvt would generate a local tag. However, when the time came to transmit the 481, since there was a to-tag in the INVITE, Asterisk would place this original to-tag in the 481 response. When the ACK came in, Asterisk would attempt to match the to-tag in the ACK to the generated local tag. Unfortunately, Asterisk never actually transmitted a response with the generated local tag, so the to-tag in the ACK would not match. The other problem was that when the 481 was sent, nothing was set on the sip_pvt to indicate what CSeq is expected in the ACK. To fix the first problem, we zero out the to-tag seen in the incoming INVITE. This way, Asterisk, when time to send a response, will send its generated local tag instead. To fix the second problem, we set the sip_pvt's pendinginvite to the CSeq of the INVITE when we send a 481. (closes issue ASTERISK-19892) Reported by Mark Michelson ........ Merged revisions 368625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-06 17:21 +0000 [r368605] Matthew Jordan * /, build_tools/make_version: Add feature modifier to versions produced from branches Certain branches, such as Certified Asterisk, may have a modifier added to them that specifies the features available in that branch. For branches, this modifier is expected to be reflected in the location of the branch in subversion. For example, a subversion of URL of /certified/branches/1.8.11 would have a feature modifier of 'certified'. This is slightly different then how features are determined for tags, where the feature is part of the actual tag name, e.g., "10.5.0-digiumphones". In keeping with the nomenclature used for tags, the feature specifier for branches is translated and placed after the revision numbers. For the example given previously, this would result in a branch version of "Asterisk SVN-branch-1.8.11-cert-rXXXXXX". ........ Merged revisions 368604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-06 16:09 +0000 [r368587] Kinsey Moore * /, channels/chan_sip.c: Ensure overlapping hold flags do not conflict When changing between different modes of hold, the flags were not being cleared out properly causing a failure to change hold states. (closes issue ASTERISK-19919) Patch-by: Morten Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-06 01:10 +0000 [r368568] Richard Mudgett * /, main/features.c: Fix parked call performing a DTMF blind transfer after being retrieved. When a parked call was retrieved from the parking lot, it could not do a blind transfer because it caused the involved calls to be hung up unconditionally. * Made the ParkedCall application return the ast_bridge_call() return value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc ........ Merged revisions 368567 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-05 15:27 +0000 [r368524-368536] Kinsey Moore * /, apps/app_minivm.c: Resolve some build warnings My newly upgraded compiler caught these usages of uninitialized values. They weren't actually used. ........ Merged revisions 368533 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: Ensure that pages and emails are sent using RFC822-compliant date format When localization was added to app_voicemail, these headers were altered when they should have remained in en_US format for RFC compliance. This reverts the changes to those two lines. (closes issue ASTERISK-19876) ........ Merged revisions 368520 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-04 22:02 +0000 [r368499] Mark Michelson * /, channels/chan_sip.c: Relay proper SIP responses on calling side. Revision 351130 broke corect HANGUPCAUSE setting for the 404 case in chan_sip. Other cases were also potentially broken. This patch fixes the relaying of causes to be what they used to be. (closes issue ASTERISK-19914) Reported by Pavel Troller Tested by Walter Doekes (via a reviewboard test to be committed later) Patches: chan_sip.diff uploaded by Pavel Troller (license #6302) ........ Merged revisions 368498 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-04 21:11 +0000 [r368407-368470] Richard Mudgett * /, UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call ........ Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/channel.c, /: Fix potential deadlock between masquerade and chan_local. * Restructure ast_do_masquerade() to not hold channel locks while it calls ast_indicate(). * Simplify many calls to ast_do_masquerade() since it will never return a failure now. If it does fail internally because a channel driver callback operation failed, the only thing ast_do_masquerade() can do is generate a warning message about strange things may happen and press on. * Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This change fixes half of the deadlock reported in ASTERISK-19801 between masquerades and chan_iax. (closes issue ASTERISK-19537) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1915/ ........ Merged revisions 368405 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-01 23:24 +0000 [r368310] Richard Mudgett * /, apps/app_stack.c: Fix deadlock when Gosub used with alternate dialplan switches. Attempting to remove a channel from autoservice with the channel lock held will result in deadlock. * Restructured gosub_exec() to not call ast_parseable_goto() and ast_exists_extension() with the channel lock held. (closes issue ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 368308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-01 20:22 +0000 [r368267] Kevin P. Fleming * /, channels/chan_sip.c: Improve SDP parsing warning messages * 'Unsupported media type' is only reported when that is in fact the case, not when a supported media type is included in an 'm' line that has an invalid format. * All warning messages related to parsing 'm' lines now include the 'm' line contents. * (minor bugfix) newline added to port-number-zero warning messages. * Warning messages improved to use RFC-specified terminology for various items. * Warnings for offers that include more than one port for a single media type now include the media type. Review: https://reviewboard.asterisk.org/r/1811/ ........ Merged revisions 368218 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-01 03:28 +0000 [r368093] Michael L. Young * /, funcs/func_channel.c: Add documentation to function CHANNEL for options echocan_mode and buffers The ability to set "echocan_mode" and "buffers" through the dialplan was added to chan_dahdi some time ago. This patch adds some documentation to func_channel. (Closes issue ASTERISK-19911) Reported by: Dale Noll Tested by: Michael L. Young Patches: asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1949/ ........ Merged revisions 368092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-31 18:20 +0000 [r368042] Richard Mudgett * res/ael/pval.c, main/tcptls.c, main/manager.c, res/res_config_odbc.c, /, channels/chan_sip.c, channels/chan_agent.c, funcs/func_math.c, main/features.c, apps/app_queue.c, channels/chan_iax2.c, pbx/pbx_config.c: Coverity Report: Fix issues for error type REVERSE_INULL (core modules) * Fixes findings: 0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt Jordan ........ Merged revisions 368039 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-30 18:07 +0000 [r367907-367981] Richard Mudgett * /, channels/sig_pri.c, channels/sig_ss7.c: Use the DEADLOCK_AVOIDANCE() macro instead. (issue ASTERISK-19854) ........ Merged revisions 367980 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when executing CLI "pri show channels" and "ss7 show channels" commands. * Fix sig_pri_lock_owner() to avoid deadlock properly. * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid deadlock properly. * Code ss7_grab() better. (closes issue ASTERISK-19854) Reported by: Jaxon Patches: jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7) Tested by: Jaxon ........ Merged revisions 367976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_meetme.c: Coverity Report: Fix issues for error type REVERSE_INULL (deprecated modules) * Fix only issue pointed out by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user(). * Change use of %i to %d in sscanf() in find_user(). The use of %i gives unexpected parsing because it can accept hex, octal, and decimal integer formats. * Changed other uses of %i in app_meetme() to use %d for consistency. (issue ASTERISK-19648) Reported by: Matt Jordan ........ Merged revisions 367906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-29 18:33 +0000 [r367844] Matthew Jordan * channels/chan_skinny.c: AST-2012-008: Fix remote crash vulnerability in chan_skinny When a skinny session is unregistered, the corresponding device pointer is set to NULL in the channel private data. If the client was not in the on-hook state at the time the connection was closed, the device pointer can later be dereferened if a message or channel event attempts to use a line's pointer to said device. The patches prevent this from occurring by checking the line's pointer in message handlers and channel callbacks that can fire after an unregistration attempt. (closes issue ASTERISK-19905) Reported by: Christoph Hebeisen Tested by: mjordan, Damien Wedhorn Patches: AST-2012-008-1.8.diff uploaded by mjordan (license 6283) AST-2012-008-10.diff uploaded by mjordan (licesen 6283) 2012-05-25 16:30 +0000 [r367782] Richard Mudgett * /, channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD without suggested MOH class crash. * Made schedule_delivery() set the received frame f->data.ptr to NULL if the datalen is zero. * Fix queue_signalling() memcpy() size error. * Made queue_signalling() not use C++ keyword variable names. (closes issue ASTERISK-19597) Reported by: mgrobecker Patches: jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Michael L. Young ........ Merged revisions 367781 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-25 02:29 +0000 [r367731] Michael L. Young * /, channels/chan_sip.c: Fix pvt_sip for inbound call to use peer's allowtransfer setting The pvt_sip allowtransfer was not being set to that of the peer's setting. Therefore, the global allowtransfer setting was being used instead which would lead to calls not being transfered if the global setting was set to 'no' despite the setting on the peer being 'yes' and vice versa, calls would be allowed to transfer even if the peer's setting was 'no' but the global setting was 'yes'. (Closes issue ASTERISK-19856) Reported by: Jacek Tested by: Michael L. Young, Jacek Patches: issue-asterisk-19856-branch10-v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1923/ ........ Merged revisions 367730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-24 22:29 +0000 [r367679] Richard Mudgett * apps/app_dial.c, /, apps/app_queue.c: Fix Dial I option ignored if dial forked and one fork redirects. The Dial and Queue I option is intended to block connected line updates and redirecting updates. However, it is a feature that when a call is locally redirected, the I option is disabled if the redirected call runs as a local channel so the administrator can have an opportunity to setup new connected line information. Unfortunately, the Dial and Queue I option is disabled for *all* forked calls if one of those calls is redirected. * Make the Dial and Queue I option apply to each outgoing call leg independently. Now if one outgoing call leg is locally redirected, the other outgoing calls are not affected. * Made Dial not pass any redirecting updates when forking calls. Redirecting updates do not make sense for this scenario. * Made Queue not pass any redirecting updates when using the ringall strategy. Redirecting updates do not make sense for this scenario. * Fixed deadlock potential with chan_local when Dial and Queue send redirecting updates for a local redirect. * Converted the Queue stillgoing flag to a boolean bitfield. (closes issue ASTERISK-19511) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1920/ ........ Merged revisions 367678 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-24 13:32 +0000 [r367562] Matthew Jordan * apps/app_confbridge.c: Fix crash in ConfBridge when user announcement is played for more than 2 users A patch introduced in r354938 made it so that ConfBridge would not attempt to play sound files if those files did not exist. Unfortunately, ConfBridge uses the same underlying function, play_sound_helper, to playback both sound files and numbers to callers. When a number is being played back, the name of the sound file is expected to be NULL. This NULL value was passed into a function that tested for the existance of a sound file and is not tolerant to NULL file names, causing a crash. This patch fixes the behavior, such that if a sound file does not exist we do not attempt to play it, but we only attempt that check if the a sound file was specified in the first place. If a sound file was not specified, we use the 'play number' logic in the helper function. (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested by: Florian Gilcher patches: asterisk-19899.diff uploaded by mjordan (license 6283) 2012-05-23 23:16 +0000 [r367470] Richard Mudgett * main/pbx.c, /: Fix WaitExten(x,m(musicclass)) string termination. The AST_CONTROL_HOLD MOH class from the WaitExten application can now be queued onto a channel, passed over local channels with the /m option, and passed over IAX channels. ........ Merged revisions 367469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-23 20:29 +0000 [r367417] Mark Michelson * main/tcptls.c, /: Only call SSL_CTX_free if DO_SSL is defined. Thanks to Paul Belanger for pointing out this error. ........ Merged revisions 367416 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-23 13:25 +0000 [r367369] Matthew Jordan * /, channels/chan_sip.c, CHANGES, channels/sip/include/sip.h: Re-add LastMsgsSent value for SIP peers Previously, MWI logic utilized a counter called 'lastmsgssent' to know whether or not MWI NOTIFY requests had been sent to a specific peer. When MWI notifications were changed to use the internal event framework, this value was no longer needed for its original purpose. Hence, it was no longer updated with the new/old message counts for a peer. The value was previously removed for Asterisk 10; however, since it was still present in Asterisk 1.8 and still useful for reporting purposes, it was decided to re-add the value. This patch re-adds the 'LastMsgsSent' field in the response to an AMI/CLI 'sip show peer [peer]' command, and makes it so that the value of lastmsgssent is updated appropriately. The value should now display the new/old message counts for a particular peer. (closes issue ASTERISK-17866) Reported by: Steve Davies patches by: ast-17866-rb1272.patch (License #5041 by irroot) Modified slightly for this commit Review: https://reviewboard.asterisk.org/r/1939 ........ Merged revisions 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-22 17:21 +0000 [r367267-367299] Terry Wilson * main/channel.c, /, include/asterisk/cel.h, include/asterisk/channel.h, main/cel.c, main/asterisk.c: Fix race condition for CEL LINKEDID_END event This patch fixes to situations that could cause the CEL LINKEDID_END event to be missed. 1) During a core stop gracefully, modules are unloaded when ast_active_channels == 0. The LINKDEDID_END event fires during the channel destructor. This means that occasionally, the cel_* module will be unloaded before the channel is destroyed. It seemed generally useful to wait until the refcount of all channels == 0 before unloading, so I added a channel counter and used it in the shutdown code. 2) During a masquerade, ast_channel_change_linkedid is called. It calls ast_cel_check_retire_linkedid which unrefs the linkedid in the linkedids container in cel.c. It didn't ref the new linkedid. Now it does. Review: https://reviewboard.asterisk.org/r/1900/ ........ Merged revisions 367292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Resolve crash in subscribing for MWI notifications ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the variable should definitely not be used after that. To solve this in the two cases that affect subscribing for MWI notifications, we instead save the ref locally, and unref them in the error conditions. (closes issue ASTERISK-19827) Reported by: B. R Review: https://reviewboard.asterisk.org/r/1940/ ........ Merged revisions 367266 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-18 17:50 +0000 [r367003-367028] Mark Michelson * channels/chan_dahdi.c, /, main/say.c: Address MISSING_BREAK static analysis reports some more. This addresses core findings 4 and 6. Moises Silva helped me by stating that a break could be safely added to the case where it is added in chan_dahdi.c In say.c, I have added a comment indicating that static analysis complains but that it is currently unknown if this is correct. This fixes all core findings of this type. (closes issue ASTERISK-19662) reported by Matthew Jordan ........ Merged revisions 367027 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h: Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX structures were allocated but never freed. This was a bigger issue for clients than servers since new SSL_CTX structures could be allocated for each connection. Servers, on the other hand, typically set up a single SSL_CTX for their lifetime. This is solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an ssl_ctx on it, it is freed so that a new one can take its place. 2. A companion to ast_ssl_setup() called ast_ssl_teardown() has been added so that servers can properly free their SSL_CTXs. (issue ASTERISK-19278) ........ Merged revisions 367002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-18 15:45 +0000 [r366948] Matthew Jordan * main/cli.c, /, channels/chan_sip.c, funcs/func_odbc.c: Fix more memory leaks This patch adds to what was fixed in r366880. Specifically, it addresses the following: * chan_sip: dispose of an allocated frame in off nominal code paths in sip_rtp_read * func_odbc: when disposing of an allocated resultset, ensure that any rows that were appended to that resultset are also disposed of * cli: free the created return string buffer in another off nominal code path (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922/ ........ Merged revisions 366944 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-18 14:18 +0000 [r366884] Kinsey Moore * /, channels/sip/config_parser.c: Reorder and renumber tests appropriately It appears that a patch did not apply properly when adding tests 12 and 13 and test 11 was duplicated. These tests have been reordered and renumbered such that they make sense. ........ Merged revisions 366882 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-18 14:01 +0000 [r366881] Matthew Jordan * main/xmldoc.c, apps/app_voicemail.c, res/res_calendar.c, main/netsock2.c, res/res_rtp_asterisk.c, main/pbx.c, res/res_calendar_exchange.c, res/res_calendar_icalendar.c, apps/app_page.c, /, funcs/func_dialgroup.c, channels/chan_sip.c, apps/app_record.c, res/res_calendar_caldav.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c, main/editline/term.c, main/enum.c, main/config.c, res/res_srtp.c, main/cli.c, main/editline/tokenizer.c, main/data.c, channels/chan_dahdi.c, funcs/func_odbc.c, main/features.c, apps/app_minivm.c, main/editline/readline.c, channels/sip/config_parser.c: Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-17 14:41 +0000 [r366792] Jonathan Rose * /, channels/chan_sip.c: chan_sip: Fix missed locking of opposing pvt for directmedia acl from r366547 It also required deadlock avoidance since two sip_pvts structs needed to be locked simultaneously. Trunk handles it differently, so this is a 1.8 and 10 patch only. ........ (issue AST-876) Merged revisions 366791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-17 12:57 +0000 [r366741] Matthew Jordan * channels/chan_dahdi.c, /, res/res_calendar_ews.c: Fix checking bounds of array index after using it; improper sizeof This patch fixes two problems pointed out by a static analysis tool. * In chan_dahdi, when an event is handled the index of the sub channel is first obtained. In very off nominal cases, the method that determines the index can return a negative value. In the event handling code, whether or not the index returned is valid was being checked after that value was used to index into an array. This patch makes it so the value is checked before any indexing is done. * In res_calendar_ews, sizeof was being passed a pointer instead of the struct to determine the amount of memory to allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes issue ASTERISK-19671) Reported by: Matt Jordan ........ Merged revisions 366740 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-15 23:39 +0000 [r366598] Mark Michelson * /, channels/chan_sip.c: Correct misuse of ast_strip_quoted() when getting a Diversion header's reason parameter. The use here was assuming that the pointer would be updated, but the updated string is actually returned by ast_strip_quoted() instead. ........ Merged revisions 366597 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-15 20:44 +0000 [r366591] Jonathan Rose * /, channels/chan_sip.c: chan_sip: Check the right channel's host address for directmediapermit/deny Prior to this patch, when checking the addresses for directmediapermit and denydirectmediadeny, Asterisk would check the host address of the channel permit/deny was specified, which defers from the expectations of both our users and the development team. Instead, directmediapermit/deny now checks against the address of the channel that the peer with the ACL is connected to. (issue AST-876) Review: https://reviewboard.asterisk.org/r/1899/ ........ Merged revisions 366547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-14 20:06 +0000 [r366390-366412] Mark Michelson * /, pbx/dundi-parser.c: Fix two more coverity constant expression result findings. These correspond to findings 0 and 1 in the core findings of ASTERISK-19649. After contacting Mark Spencer, he was unsure of what the intent behind these lines of code were, so they are being axed. For Asterisk 1.8 and 10, the output of debugging DUNDi frames will not be changed, but for trunk the "Retry" portion will be omitted since it does not properly distinguish retransmissions from initial frames. (closes issue ASTERISK-19649) Reported by Matthew Jordan ........ Merged revisions 366409 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Fix broken reinvite glare scenario. To make a long story short, reinvite glares were broken because Asterisk would invert the To and From headers when ACKing a 491 response. The reason was because the initreq of the dialog was being changed to the incoming glared reinvite instead of being set to the outgoing glared reinvite. This change has three parts * In handle_incoming, we never will reject an ACK because it has a to-tag present, even if we think the request may be out of dialog. * In handle_request_invite, we do not change the initreq when receiving a reinvite to which we will respond with a 491. * In handle_request_invite, several superflous settings up pendinginvite have been removed since this is dones automatically by transmit_response_reliable Review: https://reviewboard.asterisk.org/r/1911 ........ Merged revisions 366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-11 23:59 +0000 [r366297] Russell Bryant * /, addons/format_mp3.c: format_mp3: Fix a possible crash mp3_read(). This patch fixes a potential crash in mp3_read() by not assuming that dbuf has enough data to finish filling up the output buffer. The patch also makes sure that the dbuf state gets reset after we know we read everything out of it already. In passing, this patch includes some other cleanups of this module, including stripping trailing whitespace, formatting fixes based on coding guidelines, and removing a number of unused members from the private state struct. (closes issue ASTERISK-19761) Reported by: Chris Maciejewsk Tested by: Chris Maciejewsk ........ Merged revisions 366296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-10 23:42 +0000 [r366241] Richard Mudgett * main/channel.c, /: * Made ast_change_name() hold the channels container lock while changing the channel name. * Eliminate redundant list not empty check in clone_variables(). ........ Merged revisions 366240 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-10 20:54 +0000 [r366168] Kinsey Moore * main/xmldoc.c, apps/app_voicemail.c, funcs/func_speex.c, main/pbx.c, res/res_calendar_icalendar.c, /, channels/chan_sip.c, funcs/func_lock.c, channels/chan_agent.c, channels/sip/reqresp_parser.c, main/devicestate.c, pbx/dundi-parser.c, channels/chan_iax2.c, channels/iax2-parser.c, main/config.c, res/res_monitor.c, main/cdr.c, main/channel.c, res/ael/pval.c, main/data.c, channels/chan_dahdi.c, main/tcptls.c, main/manager.c, main/features.c, main/app.c, main/event.c, pbx/pbx_dundi.c, res/res_odbc.c: Resolve FORWARD_NULL static analysis warnings This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-10 16:55 +0000 [r366106] Jonathan Rose * main/xmldoc.c, apps/app_voicemail.c, main/pbx.c, channels/sig_analog.c, /, channels/chan_sip.c, funcs/func_lock.c, main/features.c, main/acl.c, channels/iax2-provision.c, apps/app_queue.c, channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c, main/asterisk.c: Coverity Report: Fix issues for error type CHECKED_RETURN for core (issue ASTERISK-19658) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1905/ ........ Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-10 16:13 +0000 [r366053] Mark Michelson * /, channels/chan_sip.c: Close the proper tcptls_session when session creation fails. (issue AST-998) Reported by: Thomas Arimont Tested by: Thomas Arimont ........ Merged revisions 366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-10 15:43 +0000 [r365990-366049] Jonathan Rose * /, apps/app_page.c, funcs/func_cdr.c, main/features.c, apps/app_disa.c, apps/app_chanspy.c: Coverity Report: Fix issues for error type UNINIT in Core supported modules (issue ASTERISK-19652) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1909/ ........ Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, codecs/codec_dahdi.c: Block on frameout if the hardware has enough samples to complete a frame. Fixes some problems with skipping audio in elaborate scenarios involving multiple codecs by making codec_dahdi operate in a more synchronous fashion similar to codec_g729. This change also fixes the use of file conversion tools from Asterisk's CLI. This change may cause the thread responsible for transcoding audio to block briefly (Shaun Ruffell describes this as 'several milliseconds') while waiting for the hardware transcoder. (closes issue ASTERISK-19643) reported by: Shaun Ruffell Patches: 0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch uploaded by Shaun Ruffell (license 5417) ........ Merged revisions 365989 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-09 16:15 +0000 [r365898] Mark Michelson * /, channels/chan_sip.c: Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt. chan_sip was coded under the assumption that a SIP dialog with an owner channel will always be destroyed after the owner channel has been hung up. However, there are situations where the SIP dialog can time out and auto destruct before the corresponding channel has hung up. A typical example of this would be if the 'h' extension in the dialplan takes a long time to complete. In such cases, __sip_autodestruct() would complain about the dialog being auto destroyed with an owner channel still in place. The problem is that even once the owner channel was hung up, the sip_pvt would still be linked in its ao2_container because nothing would ever unlink it. The fix for this is that if __sip_autodestruct() is called for a sip_pvt that still has an owner channel in place, the destruction is rescheduled for 10 seconds in the future. This will continue until the owner channel is finally hung up. (closes issue ASTERISK-19425) reported by David Cunningham Patches: ASTERISK-19425.patch uploaded by Mark Michelson (License #5049) (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by Dean Vesvuio ........ Merged revisions 365896 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-08 20:25 +0000 [r365632-365701] Richard Mudgett * /, apps/app_followme.c: * Fix FollowMe memory leak on error paths in app_exec(). * Fix FollowMe leaving recorded caller name file on error paths in app_exec(). * Use correct buffer dimension define in struct call_followme.moh[] and struct fm_args.namerecloc[]. This fixes unexpected namerecloc filename length restriction. ........ Merged revisions 365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_followme.c: * Fix accept/decline DTMF buffer overwrite in FollowMe. * Made use MAX_YN_STRING define to make all accept/decline DTMF buffers the same size. Just using 20 isn't good enough when someone didn't get the memo. * Fix stupid use of a global variable in FollowMe. (ynlongest) * Fix bit field declarations in FollowMe. ........ Merged revisions 365631 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-08 15:51 +0000 [r365575] Mark Michelson * /, channels/chan_sip.c: Send more accurate identification information in dialog-info SIP NOTIFYs. This uses the calling channel's caller ID and connected line information to populate the remote and local identities in the dialog-info NOTIFY when an extension is ringing. There is a bit of an oddity here, and that is that we seed the remote target with the To header of the outbound call rather than the from header. This is because it was reported that seeding with the from header caused hints to be broken with certain SNOM devices. A comment has been added to the code to explain this. (closes issue ASTERISK-16735) reported by Maciej Krajewski patches: local_remote_hint2.diff uploaded by Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark Michelson (license #5049) Tested by Niccolo Belli ........ Merged revisions 365574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-07 18:43 +0000 [r365478] Richard Mudgett * /, tests/test_config.c: Fix type punned compiler warning in test_config.c ........ Merged revisions 365476 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-07 18:39 +0000 [r365475] Matthew Jordan * apps/app_voicemail.c, main/pbx.c, /: Support VoiceMail d() option when extension does not exist in channel's context The VoiceMail d([c]) option is documented to accept digits for a new extension in context , if played during the greeting. This option works fine if the extension being redirected to has an extension with the same initial digit in the channel's current context. If that digit did not happen to exist in some extension, a dialplan match would fail and the user would not be redirected. This patch fixes it such that if the option is used, the extensions are matched in that context as opposed to the caller's original context. (closes issue ASTERISK-18243) Reported by: mjordan Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1892 ........ Merged revisions 365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-04 22:15 +0000 [r365399] Kinsey Moore * apps/app_voicemail.c, /, channels/chan_sip.c, funcs/func_aes.c, main/features.c, apps/app_followme.c, channels/chan_iax2.c, channels/sip/config_parser.c, pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c, main/config.c: Fix many issues from the NULL_RETURNS Coverity report Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-04 16:28 +0000 [r365320] Richard Mudgett * channels/chan_local.c, /: Fix local channel chains optimizing themselves out of a call. * Made chan_local.c:check_bridge() check the return value of ast_channel_masquerade(). In long chains of local channels, the masquerade occasionally fails to get setup because there is another masquerade already setup on an adjacent local channel in the chain. * Made the outgoing local channel (the ;2 channel) flush one voice or video frame per optimization attempt. * Made sure that the outgoing local channel also does not have any frames in its queue before the masquerade. * Made do the masquerade immediately to minimize the chance that the outgoing channel queue does not get any new frames added and thus unconditionally flushed. * Made block indication -1 (Stop tones) event when the local channel is going to optimize itself out. When the call is answered, a chain of local channels pass down a -1 indication for each bridge. This blizzard of -1 events really slows down the optimization process. (closes issue ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec Davis Review: https://reviewboard.asterisk.org/r/1894/ ........ Merged revisions 365313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-04 15:51 +0000 [r365299] Mark Michelson * res/res_rtp_asterisk.c, /: Fix core FINDING 2, FINDING 3, and FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report. These three all are in RTP code that attempts to print the number of sequence number cycles in an RTCP RR report. The code was masking out the upper 16 bits and then shifting the number right by 16 bits. This led to an all zero result in all cases. The fix is to do the shift without the bit masking. (issue ASTERISK-19649) ........ Merged revisions 365298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-03 15:01 +0000 [r365155-365160] Alexandr Anikin * addons/ooh323c/src/ooh323.c, /, addons/ooh323c/src/h323/H323-MESSAGES.h, addons/ooh323c/src/h323/H323-MESSAGESEnc.c: Fix warning of Coverity Static analysis, change H225ProtocolIdentifier from value to pointer per functions that use this. (close issue ASTERISK-19670) Reported by: Matt Jordan Patches: ASTERISK-19670.patch (License #5415) ........ Merged revisions 365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * addons/ooh323c/src/ooq931.c, /: Fix coverity static analysis warning, allocate full ie structure instead of without data buffer (close issue ASTERISK-19674) Reported by: Matt Jordan Patches: ASTERISK-19674.patch (License #5415) ........ Merged revisions 365143 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-06-04 Asterisk Development Team * Asterisk 10.5.0 Released. 2012-05-30 Asterisk Development Team * Asterisk 10.5.0-rc2 Released. * Resolve crash in subscribing for MWI notifications. ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the variable shoudl definitely not be used after that. To solve this in the two cases that affect subscribing for MWI notifications, we instead save the ref locally, and unref them in the error conditions. (closes issue ASTERISK-19827) Reported by: B. R. Review: https://reviewboard.asterisk.org/r/1940/ * Fix crash in ConfBridge when user announcement is played for more than 2 users A patch introduced in r354938 made it so that ConfBridge would not attempt to play sound files if those files did not exist. Unfortunately, ConfBridge uses the same underlying fucntion, play_sound_helper, to playback both the sound files and numbers to callers. When a number is being played back, the name of the sound file is expected to be NULL. This NULL value was passed into a function that tested for the existance of a sound file and is not tolerant to NULL file names, causing a crash. This patch fixes the behavior, such that if a sound file does not exist we do not attempt to play it, but we only attempt that check if the sound file was specified in the first place. If a sound file was not specified, we use the 'play number' logic in the helper function. (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested by: Florian Gilcher patches: ASTERISK-19899.diff uploaded by mjordan (license 6283) * AST-2012-007 * AST-2012-008 2012-05-03 Asterisk Development Team * Asterisk 10.5.0-rc1 Released. 2012-05-02 17:29 +0000 [r365083] Terry Wilson * channels/chan_local.c, /, main/cel.c: Multiple revisions 365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race and local channel linkedids This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes the race condition by no longer scanning the channel list for "other" channels with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings and uses the refcount of the string as a counter of how many channels with the linkedid exist. Not only does this eliminate the race condition, but it also allows us to look up the linkedid by the hashed key instead of traversing the entire channel list. Review: https://reviewboard.asterisk.org/r/1895/ ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines Don't leak a ref if out of memory and can't link the linkedid If the ao2_link fails, we are most likely out of memory and bad things are going to happen. Before those bad things happen, make sure to clean up the linkedid references. This patch also adds a comment explaining why linkedid can't be passed to both local channel allocations and combines two ao2_ref calls into 1. Review: https://reviewboard.asterisk.org/r/1895/ ........ Merged revisions 365006,365068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-02 16:16 +0000 [r365014] Michael L. Young * tests/test_security_events.c: Update security events unit tests The security events framework API was changed in Asterisk 10 but the unit tests were not updated at the same time. This patch does the following: * Adds two more security events that were added to the API * Add challenge, received_challenge and received_hash in the inval_password security event unit test (issue ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael L. Young Patches: issue-asterisk-19760-branch10.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1877/ 2012-05-02 02:44 +0000 [r364965] Matthew Jordan * main/audiohook.c: Only log a failure to get read/write samples from factories if it didn't happen In audiohook_read_frame_both, anytime samples are obtained from the read/write factories a debug statement is logged stating that samples were not obtained from the factories. This statement used to only occur if option_debug was turned on and no samples were obtained; in some refactoring when the option_debug statement was removed, the "else" clause was removed as well. This patch makes it so that those debug log statements only occur if the condition leading up to them actually happened. 2012-05-01 23:14 +0000 [r364903] Richard Mudgett * /, main/astobj2.c: Fixed __ao2_ref() validating user_data twice. (closes issue ASTERISK-19755) Reported by: Gunther Kelleter Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter ........ Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-01 23:10 +0000 [r364900] Mark Michelson * /, funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON error. As it turned out, this wasn't a huge deal. We were calling ast_app_parse_options() for a set of options of which none took arguments. The proper thing to do for this case is to pass NULL for the "args" parameter here. We were instead passing a seemingly-randomly chosen char * from the function. While this would never get written to, you can rest assured things would have gotten bad had new options (which took arguments) been added to func_volume. (closes issue ASTERISK-19656) ........ Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-01 21:50 +0000 [r364845] Richard Mudgett * channels/chan_local.c, /: * Fix error path resouce leak in local_request(). * Restructure local_request() to reduce indentation. ........ Merged revisions 364840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-01 21:44 +0000 [r364842] Jason Parker * main/manager.c, /: Prevent a potential crash when using manager hooks. Found by me while poking at DPMA-127. ........ Merged revisions 364841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-01 19:07 +0000 [r364787] Kinsey Moore * /, apps/app_confbridge.c: Play conf-placeintoconf message to the correct channel Correct the code in app_confbridge to play the conf-placeintoconf message to the marked user entering the bridge instead of to the conference while the marked user hears silence. (closes issue ASTERISK-19641) Reported-by: Mark A Walters ........ Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-01 18:23 +0000 [r364777] Jonathan Rose * /, main/app.c: Fix bad check in voicemail functions for ast_inboxcount2_func Check looks for ast_inboxcount_func instead of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes issue ASTERISK-19718) Reported by: Corey Farrell Patches: ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-30 19:42 +0000 [r364707] Mark Michelson * /, channels/chan_sip.c: Revert improved identities sent in dialog-info NOTIFY requests in r360862 Revision 360862 was intended to improve identities sent in dialog-info NOTIFY requests. Some users reported that hint became broken once this was done. It's not clear exactly what part of the patch has caused this regression, but broken hints are bad. For now, this revision is being reverted so that the next releases of Asterisk do not have bad behavior in them. The original reported issue will have to be fixed differently in the next version of Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-30 16:48 +0000 [r364651] Alexandr Anikin * /, addons/ooh323cDriver.c: Fix use freed pointer in return value from call thread (issue ASTERISK-19663) Reported by: Matt Jordan Patches: ASTERISK-19663-ooh323.patch (License #5415) ........ Merged revisions 364649 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-30 16:43 +0000 [r364650] Mark Murawki * /, main/logger.c: Merged revisions 364635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) | 10 lines Sanatize result from bfd_find_nearest_line (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file to null resulting in a crash when strrchr(file) runs (closes issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark Murawski ........ 2012-04-29 19:43 +0000 [r364579] Matthew Jordan * formats/format_g719.c, formats/format_siren7.c, formats/format_g729.c, formats/format_ilbc.c, /, formats/format_sln.c, formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, formats/format_g723.c, formats/format_h263.c, formats/format_h264.c, formats/format_wav_gsm.c, formats/format_siren14.c, formats/format_gsm.c: Fix error that caused truncate operations to fail Another very inappropriate placement of a ')' (again introduced in r362151) caused the various truncate operations to attempt to truncate the sound file at a position of '0'. (issue ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810) Reported by: colbec ........ Merged revisions 364578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-29 02:21 +0000 [r364536] Michael L. Young * apps/confbridge/conf_config_parser.c: Fix configuring custom sound_leader_has_left in confbridge.conf The configuration option to specify a custom sound_leader_has_left file for a conference bridge was not being parsed. This patch fixes it so that a custom sound file will now be used. (closes issue ASTERISK-19771) Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380) Review: https://reviewboard.asterisk.org/r/1884/ 2012-04-27 22:33 +0000 [r364365-364369] Terry Wilson * tests/test_config.c (added): Add missing test_config.c * /, main/config.c: Fix ast_parse_arg numeric type range checking and add tests ast_parse_arg wasn't checking for strto* parse errors or limiting the results by the actual range of the numeric types. This patch fixes that and adds unit tests as well. Review: https://reviewboard.asterisk.org/r/1879/ ........ Merged revisions 364340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-27 21:58 +0000 [r364342] Mark Michelson * /, channels/chan_sip.c: Don't attempt to make use of the dynamic_exclude_static ACL if DNS lookup fails. (closes issue ASTERISK-18321) Reported by Dan Lukes Patches: ASTERISK-18321.patch by Mark Michelson (license #5049) ........ Merged revisions 364341 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-27 19:30 +0000 [r364285] Matthew Jordan * include/asterisk/time.h, /: Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds, between two timeval structs, and return the difference in a 64-bit integer. Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval struct are large enough to hold the calculated values before it returns. On 64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit machines, however, a long may be less (32-bits), in which case, the calculation can overflow. This overflow caused significant problems in MixMonitor, which uses the method to determine if an audio factory, which has not presented audio to an audiohook, is merely late in providing said audio or will never provide audio. In an overflow situation, the audiohook would incorrectly determine that an audio factory that will never provide audio is merely late instead. This led to situations where a MixMonitor never recorded any audio. Note that this happened most frequently when that MixMonitor was started by the ConfBridge application itself, or when the MixMonitor was attached to a Local channel. (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark Murawski Tested by: Michael L. Young Patches: 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan) (closes issue ASTERISK-19471) Reported by: feyfre Tested by: feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1889/ ........ Merged revisions 364277 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-27 18:58 +0000 [r364259] Kinsey Moore * /, channels/chan_sip.c: Allow SIP pvts involved in Replaces transfers to fall out of reference sooner Unref the SIP pvt stored in the refer structure as soon as it is no longer needed so that the pvt and associated file descriptors can be freed sooner. This change makes a reference decrement unnecessary in code that handles SIP BYE/Also transfers which should not touch the reference anyway. (related to issue ASTERISK-19579) ........ Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-27 14:44 +0000 [r364204] Matthew Jordan * /, channels/chan_sip.c: Allow for reloading SRTP crypto keys within the same SIP dialog As a continuation of the patch in r356604, which allowed for the reloading of SRTP keys in re-INVITE transfer scenarios, this patch addresses the more common case where a new key is requested within the context of a current SIP dialog. This can occur, for example, when certain phones request a SIP hold. Previously, once a dialog was associated with an SRTP object, any subsequent attempt to process crypto keys in any SDP offer - either the current one or a new offer in a new SIP request - were ignored. This patch changes this behavior to only ignore subsequent crypto keys within the current SDP offer, but allows future SDP offers to change the keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Review: https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-27 12:54 +0000 [r364163] Stefan Schmidt * res/res_calendar_icalendar.c, res/res_calendar_caldav.c: fix a wrong behavior of alarm timezones in caldav and icalendar when an alarm doesnt use utc. This change uses the same timezone from the start time. 2012-04-26 21:10 +0000 [r364065-364109] Richard Mudgett * /, apps/app_directed_pickup.c: Update Pickup application documentation. (With feeling this time.) ........ Merged revisions 364108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/features.c: Fix DTMF atxfer running h exten after the wrong bridge ends. When party B does an attended transfer of party A to party C, the attending bridge between party B and C should not be running an h exten when the bridge ends. Running an h exten now sets a softhangup flag to ensure that an AGI will run in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the attending bridge between party B and C. (closes issue AST-870) (closes issue ASTERISK-19717) Reported by: Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario ........ Merged revisions 364060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-26 19:30 +0000 [r364047] Terry Wilson * /, main/asterisk.c: Add more constness to the end_buf pointer in the netconsole issue ASTERISK-18308 Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 364046 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-26 13:27 +0000 [r363987] Kinsey Moore * /, channels/chan_sip.c: Fix reference leaks involving SIP Replaces transfers The reference held for SIP blind transfers using the Replaces header in an INVITE was never freed on success and also failed to be freed in some error conditions. This caused a file descriptor leak since the RTP structures in use at the time of the transfer were never freed. This reference leak and another relating to subscriptions in the same code path have now been corrected. (closes issue ASTERISK-19579) ........ Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-26 09:46 +0000 [r363935] Alec L Davis * /, channels/chan_sip.c: chan_sip: [general] maxforwards, not checked for a value greater than 255 The peer maxforwards is checked for both '< 1' and '> 255', but the default 'maxforwards' in the [general] section is only checked for '< 1' alecdavis (license 585) Reported by: alecdavis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1888/ ........ Merged revisions 363934 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-26 03:11 +0000 [r363376-363876] Richard Mudgett * /, apps/app_directed_pickup.c: Update Pickup application documentation. (Even better) ........ Merged revisions 363875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_directed_pickup.c: Update Pickup application documentation. ........ Merged revisions 363788 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_dahdi.c, /, channels/sig_pri.c: Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call. Some switches may not handle the call-deflection/call-rerouting message if the call is disconnected too soon after being sent. Asteisk was not waiting for any reply before disconnecting the call. * Added a 5 second delay before disconnecting the call to wait for a potential response if the peer does not disconnect first. (closes issue ASTERISK-19708) Reported by: mehdi Shirazi Patches: jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Clear ISDN channel resetting state if the peer continues to use it. Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in response to a RESTART request. * Made the second SETUP received after sending a RESTART request clear the channel resetting state as if the peer had sent the expected RESTART ACKNOWLEDGE before continuing to process the SETUP. The peer may not be sending the expected RESTART ACKNOWLEDGE. (issue ASTERISK-19608) (issue AST-844) (issue AST-815) Patches: jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified) ........ Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/features.c: Fix recalled party B feature flags for a failed DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF atxfer to C 3) B hangs up 4) C does not answer 5) B is called back 6) B answers 7) B cannot initiate transfers anymore * Add dial features datastore to recalled party B channel that is a copy of the original party B channel's dial features datastore. * Extracted add_features_datastore() from add_features_datastores(). * Renamed struct ast_dial_features features_caller and features_callee members to my_features and peer_features respectively. These better names eliminate the need for some explanatory comments. * Simplified code accessing the struct ast_dial_features datastore. (closes issue ASTERISK-19383) Reported by: lgfsantos ........ Merged revisions 363428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/features.c: Hangup affected channel in error paths of bridge_call_thread(). ........ Merged revisions 363375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-23 16:06 +0000 [r363212] Tilghman Lesher * /, main/astfd.c: On some platforms, O_RDONLY is not a flag to be checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX specification does not mandate how these 3 flags must be specified, only that one of the three must be specified in every call. ........ Merged revisions 363209 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-23 14:39 +0000 [r363156] Jonathan Rose * main/manager.c, /: AST-2012-004: Fix an error that allows AMI users to run shell commands sans authorization. As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI actions were able to run system commands by going through other AMI commands which did not require that authorization. Specifically, GetVar and Status allowed users to do this by setting their variable/s options to the SHELL or EVAL functions. Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that allowed users with originate permission to run MixMonitor and supply a shell command in the Data argument. That flaw is fixed in those versions of this patch. (closes issue ASTERISK-17465) Reported By: David Woolley Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182) ........ Merged revisions 363117 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 363141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-23 14:07 +0000 [r363103-363107] Matthew Jordan * /, channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE handling when no channel owner exists If Asterisk receives a SIP UPDATE request after a call has been terminated and the channel has been destroyed but before the SIP dialog has been destroyed, a condition exists where a connected line update would be attempted on a non-existing channel. This would cause Asterisk to crash. The patch resolves this by first ensuring that the SIP dialog has an owning channel before attempting a connected line update. If an UPDATE request is received and no channel is associated with the dialog, a 481 response is sent. (closes issue ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 363106 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_skinny.c: Reference skinny_subchannel object instead of skinny_device for r363103 The check-in to resolve ASTERISK-19592 (r363103) failed to switch to the skinny_subchannel object instead of the skinny_device when attempting to reference the buffer for the keypad digits. This patch fixes that. (issue ASTERISK-19592) Reported by: Russell Bryant * /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable heap overflow in keypad button handling When handling a keypad button message event, the received digit is placed into a fixed length buffer that acts as a queue. When a new message event is received, the length of that buffer is not checked before placing the new digit on the end of the queue. The situation exists where sufficient keypad button message events would occur that would cause the buffer to be overrun. This patch explicitly checks that there is sufficient room in the buffer before appending a new digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant ........ Merged revisions 363100 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 363102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-21 01:45 +0000 [r362998] Richard Mudgett * apps/app_dial.c, /: Update app_dial M and U option GOTO return value documentation. ........ Merged revisions 362997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-20 16:47 +0000 [r362918] Michael L. Young * main/event.c: Add missing payload type to events API The Security Events Framework API was changed while adding the generation of security events in chan_sip. A payload type and name was missed from being added to struct ie_maps. (closes issue ASTERISK-19759) Reported by: Michael L. Young Patches: issue-asterisk-19759.diff uploaded by Michael L. Young (license 5026) 2012-04-20 16:12 +0000 [r362816-362869] Terry Wilson * /, main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr (closes issue ASTERISK-19758) Reported by: Barry Miller Tested by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller (license 5434) ........ Merged revisions 362868 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_speech_utils.c, /: Document Speech* apps hangup on failure and suggest TryExec The Speech API apps return -1 on failure, which will hang up the channel. This may not be desirable behavior for some, but it isn't something that can be changed without breaking people's dialplans or writing an option to all of the Speech apps that does what TryExec already does. This patch documents the hangup behavior of the apps, and suggests TryExec as the solution. (closes issue AST-813) ........ Merged revisions 362815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-19 21:59 +0000 [r362730] Walter Doekes * funcs/func_version.c, /: Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions 362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-19 21:11 +0000 [r362681] Michael L. Young * /, tests/test_linkedlists.c, tests/test_poll.c: Add leading and trailing backslashes A couple of unit tests did not have have leading or trailing backslashes when setting their test category resulting in a warning message being displayed. Added the backslash where needed. ........ Merged revisions 362680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-19 21:00 +0000 [r362678] Richard Mudgett * /, configs/queues.conf.sample: Update membermacro and membergosub documentation in queues.conf.sample. ........ Merged revisions 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-19 16:04 +0000 [r362587] Sean Bright * /, apps/app_externalivr.c: Prevent a crash in ExternalIVR when the 'S' command is sent first. If the first command sent from an ExternalIVR client is an 'S' command, we were blindly removing the first element from the play list and deferencing it, even if it was NULL. This corrects that and also locks appropriately in one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski ........ Merged revisions 362586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-19 14:31 +0000 [r362537] Terry Wilson * /, main/asterisk.c: Handle multiple commands per connection via netconsole Asterisk would accept multiple NULL-delimited CLI commands via the netconsole socket, but would occasionally miss a command due to the command not being completely read into the buffer. This patch ensures that any partial commands get moved to the front of the read buffer, appended to, and properly sent. (closes issue ASTERISK-18308) Review: https://reviewboard.asterisk.org/r/1876/ ........ Merged revisions 362536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-19 02:27 +0000 [r362496] Matthew Jordan * channels/chan_unistim.c, /, main/tdd.c, main/jitterbuf.c, apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c, addons/chan_mobile.c, main/format_pref.c, main/asterisk.c: Fix a variety of potential buffer overflows * chan_mobile: Fixed an overrun where the cind_state buffer (an integer array of size 16) would be overrun due to improper bounds checking. At worst, the buffer can be overrun by a total of 48 bytes (assuming 4-byte integers), which would still leave it within the allocated memory of struct hfp. This would corrupt other elements in that struct but not necessarily cause any further issues. * app_sms: The array imsg is of size 250, while the array (ud) that the data is copied into is of size 160. If the size of the inbound message is greater then 160, up to 90 bytes could be overrun in ud. This would corrupt the user data header (array udh) adjacent to ud. * chan_unistim: A number of invalid memmoves are corrected. These would move data (which may or may not be valid) into the ends of these buffers. * asterisk: ast_console_toggle_loglevel does not check that the console log level being set is less then or equal to the allowed log levels of 32. * format_pref: In ast_codec_pref_prepend, if any occurrence of the specified codec is not found, the value used to index into the array pref->order would be one greater then the maximum size of the array. * jitterbuf: If the element being placed into the jitter buffer lands in the last available slot in the jitter history buffer, the insertion sort attempts to move the last entry in the buffer into one slot past the maximum length of the buffer. Note that this occurred for both the min and max jitter history buffers. * tdd: If a read from fsk_serial returns a character that is greater then 32, an attempt to read past one of the statically defined arrays containing the values that character maps to would occur. * localtime: struct ast_time and tm are not the same size - ast_time is larger, although it contains the elements of tm within it in the same layout. Hence, when using memcpy to copy the contents of tm into ast_time, the size of tm should be used, as opposed to the size of ast_time. * extconf: this treats ast_timing's minmask array as if it had a length of 48, when it has defined the size of the array as 24. pbx.h defines minmask as having a size of 48. (issue ASTERISK-19668) Reported by: Matt Jordan ........ Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-18 16:27 +0000 [r362429] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, channels/sig_pri.c: Add ability to ignore layer 1 alarms for BRI PTMP lines. Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 ........ Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-17 21:22 +0000 [r362360-362377] Matthew Jordan * /, main/format_pref.c: Handle case where an unknown format is used to get the preferred codec size In ast_codec_pref_getsize, if an unknown format is passed to the method, no preferred codec will be selected and a negative number will be used to index into the format list. The method now logs an unknown format as a warning, and returns an empty format list. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ * res/res_rtp_asterisk.c, /, res/res_agi.c, res/res_musiconhold.c: Fix places in resources where a negative return value could impact execution This patch addresses a number of modules in resources that did not handle the negative return value from function calls adequately. This includes: * res_agi.c: if the result of the read function is a negative number, indicating some failure, the result would instead be treated as the number of bytes read. This patch now treats negative results in the same manner as an end of file condition, with the exception that it also logs the error code indicated by the return. * res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor to srcfd, and instead assigns a negative value, that file descriptor could later be passed to functions that require a valid file descriptor. If spawn_mp3 fails, we now immediately retry instead of continuing in the logic. * res_rtp_asterisk.c: if no codec can be matched between two RTP instances in a peer to peer bridge, we immediately return instead of attempting to use the codec payload type as an index to determine the appropriate negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/manager.c, /, main/asterisk.c: Fix places in main where a negative return value could impact execution This patch addresses a number of modules in main that did not handle the negative return value from function calls adequately, or were not sufficiently clear that the conditions leading to improper handling of the return values could not occur. This includes: * asterisk.c: A negative return value from the read function would be used directly as an index into a buffer. We now check for success of the read function prior to using its result as an index. * manager.c: Check for failures in mkstemp and lseek when handling the temporary file created for processing data returned from a CLI command in action_command. Also check that the result of an lseek is sanitized prior to using it as the size of a memory map to allocate. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-17 20:57 +0000 [r362357] Jonathan Rose * res/res_config_curl.c, res/res_config_pgsql.c, res/res_config_odbc.c, /: Make use of va_args more appropriate to form in various res_config modules plus utils. A number of va_copy operations weren't matched with a corresponding va_end in res_config_odbc. Also, there was a potential for va_end to be invoked twice on the same va_arg in utils, which would mean invoking va_end on an undefined variable... which is bad. va_end is removed from various functions in config_pgsql and config_curl since they aren't making their own copy. The invokers of those functions are responsible for calling va_end on them. (issue ASTERISK-19451) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1848/ ........ Merged revisions 362354 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-17 20:56 +0000 [r362305-362356] Matthew Jordan * /, funcs/func_env.c: Fix places where a negative return from ftello could be used as invalid input In a variety of locations in both reading and writing a file, the result from the C library function ftello is used as input to other functions. For the parameters and functions in question, a negative value is invalid input. This patch checks the return value from the ftello function to determine if we were able to determine the current position in the file stream and, if not, fail gracefully. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362355 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * formats/format_g719.c, formats/format_siren7.c, /, formats/format_sln.c, formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, formats/format_wav_gsm.c, formats/format_siren14.c, formats/format_gsm.c: Fix error that caused seek format operations to set max file size to '1' or '0' A very inappropriate placement of a ')' (introduced in r362151) caused the maximum size of a file to be set as the result of a comparison operation, as opposed to the result of the ftello operation. This resulted in seeking being restricted to the beginning of the file, or 1 byte into the file. Thanks to the Asterisk Test Suite for properly freaking out about this on at least one test. (issue ASTERISK-19655) Reported by: Matt Jordan ........ Merged revisions 362304 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-17 14:53 +0000 [r362264] Michael L. Young * /, channels/chan_sip.c: Turn off warning message when bind address is set to any. When a bind address is set to an ANY address (udpbindport=::), a warning message is displayed stating that "Address remapping activated in sip.conf but we're using IPv6, which doesn't need it. Please remove 'localnet' and/or 'externaddr' settings." But if one is running dual stack, we shouldn't be told to turn those settings off. This patch checks if the bind address is an ANY address or not. The warning message will now only be displayed if the bind address is NOT an ANY address and IPv6 is being used. Also, updated the copyright year. (closes issue ASTERISK-19456) Reported by: Michael L. Young Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 362253 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-16 21:57 +0000 [r362152-362205] Matthew Jordan * channels/chan_dahdi.c, /, channels/chan_agent.c: Fix negative return handling in channel drivers In chan_agent, while handling a channel indicate, the agent channel driver must obtain a lock on both the agent channel, as well as the channel the agent channel is using. To do so, it attempts to lock the other channel first, then unlock the agent channel which is locked prior to entry into the indicate handler. If this unlock fails with a negative return value, which can occur if the object passed to agent_indicate is an invalid ao2 object or is NULL, the return value is passed directly to strerror, which can only accept positive integer values. In chan_dahdi, the return value of dahdi_get_index is used to directly index into the sub-channel array. If dahd_get_index returns a negative value, it would use that value to index into the array, which could cause an invalid memory access. If dahdi_get_index returns a negative number, we now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362204 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: Fix handling of negative return code when storing voicemails in ODBC storage When storing a voicemail message using an ODBC connection to a database, the voicemail message is first stored on disk. The sound file associated with the message is read into memory before being transmitted to the database. When this occurs, a failure in the C library's lseek function would cause a negative value to be passed to the mmap as the size of the memory map to create. This would almost certainly cause the creation of the memory map to fail, resulting in the message being lost. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863 ........ Merged revisions 362201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * formats/format_g719.c, formats/format_siren7.c, formats/format_g729.c, formats/format_ilbc.c, /, formats/format_sln.c, formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, formats/format_g723.c, formats/format_h263.c, formats/format_h264.c, formats/format_wav_gsm.c, formats/format_siren14.c, formats/format_gsm.c: Check for IO stream failures in various format's truncate/seek operations For the formats that support seek and/or truncate operations, many of the C library calls used to determine or set the current position indicator in the file stream were not being checked. In some situations, if an error occurred, a negative value would be returned from the library call. This could then be interpreted inappropriately as positional data. This patch checks the return values from these library calls before using them in subsequent operations. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged revisions 362151 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-13 16:04 +0000 [r362080-362084] Jonathan Rose * apps/app_forkcdr.c, /: Make ForkCDR e option not set end time of the newly forked CDR log Prior to this patch, ForkCDR's e option would immediately set the end time of the forked CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time being roughly the same as it's beginning time (which is in turn roughly the same as the original's end time). (closes issue ASTERISK-19164) Reported by: Steve Davies Patches: cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012) ........ Merged revisions 362082 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_meetme.c: Send relative path named recordings to the meetme directory instead of sounds Prior to this patch, no effort was made to parse the path name to determine a proper destination for recordings of MeetMe's r option. This fixes that. Review: https://reviewboard.asterisk.org/r/1846/ ........ Merged revisions 362079 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-12 16:22 +0000 [r361956-361981] Kinsey Moore * /, channels/chan_iax2.c: Make trunkfreq take effect when set Previously, setting trunkfreq had no effect on initial load or on reload and only ever used the default value. This causes trunkfreq to be used appropriately on initial load and reload. (closes issue ASTERISK-19521) Patch-by: Jaco Kroon ........ Merged revisions 361972 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * codecs/lpc10/Makefile, Makefile, build_tools/cflags.xml, /, build_tools/menuselect-deps.in, codecs/gsm/src/k6opt.s, configure, codecs/gsm/Makefile, configure.ac, Makefile.rules, makeopts.in: Simplify build system architecture optimization This change to the build system rips out any usage of PROC along with architecture-specific optimizations in favor of using -march=native where it is supported. This fixes broken builds on 64bit Intel systems and results in better optimized code on systems running GCC 4.2+. Review: https://reviewboard.asterisk.org/r/1852/ (closes issue ASTERISK-19462) ........ Merged revisions 361955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-11 16:07 +0000 [r361907] Jonathan Rose * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8 Prior to this patch, in order to restore that behavior, a function would have to be used on the QueueMember to make the ringinuse option do anything, which is pretty unreasonable. (closes issue ASTERISK-19536) reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1860/ 2012-04-10 21:47 +0000 [r361855] Richard Mudgett * channels/chan_dahdi.c, /: Prevent invalid access of free'd memory if DAHDI channel during an MWI event In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. * Rework the -r361705 patch to better manage the cs and mtd allocated resources. * Fixed use of mwimonitoractive flag to be correct if the mwi_thread() fails to start. ........ Merged revisions 361854 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-10 19:57 +0000 [r361658-361804] Matthew Jordan * /, main/http.c: Fix crash caused by unloading or reloading of res_http_post When unlinking itself from the registered HTTP URIs, res_http_post could inadvertently free all URIs registered with the HTTP server. This patch modifies the unregister method to only free the URI that is actually being unregistered, as opposed to all of them. ........ Merged revisions 361803 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * funcs/func_curl.c, /: Allow func_curl to exit gracefully if list allocation fails during write If the global_curl_info data structure could not be allocated, the datastore associated with the operation would be free'd, but the function would not return. This would later dereference the datastore, almost certainly causing Asterisk to crash. With this patch, if the data structure is not allocated the method will return an error code, and not attempt any further operation. ........ Merged revisions 361753 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_dahdi.c, /: Prevent invalid access of free'd memory if DAHDI channel during an MWI event In the MWI processing loop, when a valid event occurs the temporary caller ID information is deallocated. If a new DAHDI channel is successfully created, the event is passed up to the analog_ss_thread without error and the loop exits. If, however, the DAHDI channel is not created, then the caller ID struct has been free'd, and the gains reset to their previous level. This will almost certainly cause an invalid access to the free'd memory, either in subsequent calls to callerid_free or calls to callerid_feed. This patch makes it so that we only free the caller ID structure if a DAHDI channel is successfully created, and we bump the gains back up if we fail to make a DAHDI channel. ........ Merged revisions 361705 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, funcs/func_global.c: Change SHARED function to use a safe traversal when modifying a variable When the SHARED function modifies a variable, it removes it from its list of variables and reinserts the new value at the head of the list of variables. Doing this inside a standard list traversal can be dangerous, as the standard list traversal does not account for the list being changed. While the code in question should not cause a use after free violation due to its breaking out of the loop after freeing the variable, it could lead to a maintenance issue if the loop was modified. This also fixes a violation reported by a static analysis tool, which also makes this code easier to maintain in the future. ........ Merged revisions 361657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-06 22:00 +0000 [r361560-361607] Matthew Jordan * /, res/res_calendar_ews.c: Fix memory leak in res_calendar_ews when event email address node is empty If the XML calendar data returned by a Microsoft Exchange Web Service specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address is provided, a condition existed where an ast_calendar_attendee struct would be allocated but not appended to the list of attendees. Because of that, the memory associated with the attendee would never be freed. This patch frees the memory if no e-mail address is provided. ........ Merged revisions 361606 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_meetme.c: Fix memory leak when using MeetMeAdmin 'e' option with user specified A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command (eject last user that joined) is used in conjunction with a specified user. Regardless of the command being executed, if a user is specified for the command, MeetMeAdmin will look up that user. Because the 'e' option kicks the last user that joined, as opposed to the one specified, the reference to the user specified by the command would be leaked when the user variable was assigned to the last user that joined. ........ Merged revisions 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-06 19:47 +0000 [r361522] Richard Mudgett * main/message.c: Don't add an empty MESSAGE_DATA(key) header if it doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key header did not already exist. If it already existed it would delete it. * Made msg_set_var_full() exit early if the named variable did not already exist and the value to set is empty. 2012-04-06 18:13 +0000 [r361472] Kinsey Moore * channels/chan_mgcp.c, main/xmldoc.c, apps/app_voicemail.c, res/res_clioriginate.c, channels/chan_unistim.c, main/pbx.c, /, channels/chan_sip.c, funcs/func_strings.c, formats/format_ogg_vorbis.c, channels/console_video.c, channels/chan_gtalk.c, apps/app_ices.c, channels/chan_iax2.c, res/res_config_sqlite.c, res/res_srtp.c, main/cdr.c, main/tcptls.c, funcs/func_channel.c, channels/console_gui.c, apps/app_sms.c, addons/chan_mobile.c, apps/app_chanspy.c: Add missing newlines to CLI logging ........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-06 16:31 +0000 [r361422] Paul Belanger * bridges/bridge_builtin_features.c, /, funcs/func_sysinfo.c, bridges/bridge_multiplexed.c: Multiple revisions 361403,361412 ........ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ r361412 | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ Merged revisions 361403,361412 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-06 15:49 +0000 [r361381] Russell Bryant * /, configs/rpt.conf.sample (removed), configs/usbradio.conf.sample (removed), apps/rpt_flow.pdf (removed): Remove a few more files related to chan_usbradio and app_rpt. ........ Merged revisions 361380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-06 14:01 +0000 [r361333] Matthew Jordan * /, channels/chan_sip.c: Fix a typo in the warning messages for an ignored media stream Added a '\n' to the warning messages when we ignore a media stream due to the port number being '0'. (closes issue ASTERISK-19646) Reported by: Badalian Vyacheslav ........ Merged revisions 361332 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-06 13:31 +0000 [r361330] Kinsey Moore * apps/app_dial.c, /: Remove unnecessary error message in app_dial.c The error message for failure to stop autoservice after a gosub or macro call during a dial was removed for macro while Asterisk 1.4 was still being actively developed. The corresponding gosub error message was never removed. (closes issue ASTERISK-19551) ........ Merged revisions 361329 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-05 16:53 +0000 [r361208-361270] Jonathan Rose * /, apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined There were a few instances of restarting music on hold in meetme that would cause Asterisk to revert to the default class of music on hold for no adequate reason. Review: https://reviewboard.asterisk.org/r/1844/ ........ Merged revisions 361269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, addons/ooh323cDriver.c: Fix some stuff involving calls to memcpy and memset The important parts of the patch were already applied through other updates. (closes issue ASTERISK-19445) Reported by: Makoto Dei Patches: memset-memcpy-length.patch uploaded by Makoto Dei (license 5027) ........ Merged revisions 361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, funcs/func_devstate.c: Make 'help devstate change' display properly (get rid of excess comma) (closes issue ASTERISK-19444) Reported by: Makoto Dei Patches: devstate-change-usage-truncate.patch uploaded by Makoto Dei (license 5027) ........ Merged revisions 361201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-05-02 Asterisk Development Team * Asterisk 10.4.0 Released. 2012-05-01 Asterisk Development Team * Asterisk 10.4.0-rc3 Released. * channels/chan_sip.c: Revert revision 360862 Revision 360862 was intended to improve identities sent in dialog-info NOTIFY requests. Some users reported that hint became broken once this was done. It's not clear exactly what part of the patch has caused this regression, but broken hints are bad. For now, this revision is being reverted so that the next releases of Asterisk do not have bad behavior in them. The original reported issue will have to be fixed differently in the next version of Asterisk. (issue ASTERISK-16735) 2012-04-24 Asterisk Development Team * Asterisk 10.4.0-rc2 Released. * AST-2012-004 * AST-2012-005 * AST-2012-006 2012-04-04 Asterisk Development Team * Asterisk 10.4.0-rc1 Released. 2012-04-04 16:38 +0000 [r361091-361143] Jonathan Rose * main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /, channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c, apps/app_externalivr.c, channels/chan_iax2.c, res/res_fax_spandsp.c, apps/app_milliwatt.c: Replace GNU old-style field designator extensions to fix clang warnings (issue ASTERISK-19540) Reported by: Makoto Dei Patches: clang-gnu-designator.patch uploaded by Makoto Dei (license 5027) ........ Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8 Merged revisions 361142 from http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue ASTERISK-19540) * /, apps/app_meetme.c: Make the MeetMeAdmin N command (mute all nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/ ........ Merged revisions 361090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-04-03 20:08 +0000 [r360993-361041] Kinsey Moore * /, apps/app_transfer.c: Fix the display of documentation for Transfer This came up while fixing documentation generation for many other cases where the argument separator was not being displayed properly. Now that it is displayed properly, it shows up in the wrong place for Transfer since the '/' is only required if Tech is present. (related to issue ASTERISK-18168) ........ Merged revisions 361040 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Stop sending out RTCP if RTP is inactive This change prevents Asterisk from sending RTCP receiver reports during a remote bridge since it is no longer receiving media and should not be reporting anything. (related to ASTERISK-19366) ........ Merged revisions 360987 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-30 21:29 +0000 [r360934] Richard Mudgett * /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The logger_thread() had an exit path that failed to release the logmsgs list lock. * Make logger_thread() exit path unlock the logmsgs list lock. * Made ast_log() not queue any messages to the logmsgs list if the close_logger_thread flag is set. (issue ASTERISK-19463) Reported by: Matt Jordan ........ Merged revisions 360933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-29 23:33 +0000 [r360863-360885] Mark Michelson * /, main/features.c: Fix potential race condition during call pickup. Prior to this patch, a connected line update was queued during call pickup and then an answer frame was queued. The original caller would presumably then have his connected line updated and then the call would be answered. In actuality, the answer frame was not how the call ended up being answered. Rather, an odd section in app_dial that checks if the called channel's state is up. The result is that the order of the connected line update and the answer were variable. In most cases, this wasn't actually a bad thing. However, if the 'I' option was passed to dial, the connected line update would be inhibited. The fix is to queued the connected line after the answer frame is queued. This way the race in app_dial is between two conditions resulting in an answer. This way the connected line update occurs after the answer every time. (closes issue ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by Mark Michelson (license 5049) ........ Merged revisions 360884 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Improve accuracy of identifying information sent in dialog-info SIP NOTIFY requests. This change makes use of connected party information in addition to caller ID in order to populate local and remote XML elements in the dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by: Maciej Krajewski Tested by: Maciej Krajewski Patches: local_remote_hint2.diff uploaded by Mark Michelson (license 5049) ........ Merged revisions 360862 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-28 19:20 +0000 [r360717] Terry Wilson * channels/chan_jingle.c, addons/chan_ooh323.c, /, cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, channels/chan_gtalk.c: Destroy configs when they are no longer used https://reviewboard.asterisk.org/r/1834/ ........ Merged revisions 360712 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-27 18:23 +0000 [r360672] Mark Michelson * /, channels/chan_sip.c: Make a debug message regarding subscription changes more accurate. I was getting confused during some testing why Asterisk was saying that a subscription was being added when it was clearly being removed. This fixes that confusion. ........ Merged revisions 360625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-27 14:35 +0000 [r360489-360575] Jonathan Rose * /, configure: Updates config with bootstrap where I changed configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon Clark ........ Merged revisions 360574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configure.ac: Fix BETTER_BACKTRACES library detection for Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-26 18:41 +0000 [r360472-360476] Paul Belanger * /, CHANGES: Update CHANGES for r360471 ........ Merged revisions 360474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/dnsmgr.c, /: Increase verbosity level for ast_verb messages While this does not fix the issue of the CLI being flooded by 'doing dnsmgr_lookup' messages, increasing the verbosity level above 5 should help minimize it. ........ Merged revisions 360471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-24 23:47 +0000 [r360358-360414] Russell Bryant * funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error handling code path. ........ Merged revisions 360413 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_page.c: app_page: Fix a memory leak on every Page(). dial_list is a dynamically allocated array that is allocated at the beginning of Page() based on how many devices will be dialed. This was never being freed. ........ Merged revisions 360363 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_jack.c, /: app_jack: fix datastore memory leak in error handling path. ........ Merged revisions 360360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/ast_expr2.c, /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, res/ael/ael_lex.c, res/ael/ael.tab.h: Multiple revisions 360356-360357 ........ r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) | 6 lines expression parser: Fix (theoretical) memory leak. Fix a memory leak that is very unlikely to actually happen. If a malloc() succeeded, but the following strdup() failed, the memory from the original malloc() would be leaked. ........ r360357 | russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines Rebuild parsers. This is needed to include the last fix to main/ast_expr2.y. The changes look much bigger as this regeneration of the code was done with newer versions of flex and bison. ........ Merged revisions 360356-360357 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-24 00:37 +0000 [r360263-360310] Richard Mudgett * main/channel.c, /, channels/sig_pri.c: Make number not available presentation also set screening to network provided. Q.951 indicates that when the presentation indicator is "Number not available due to interworking" for a number then the screening indicator field should be "Network provided". * Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE when the presentation is "Number not available due to interworking". This fix makes Asterisk consistent and it also makes it consistent with earlier branches as far as this presentation value is concerned. * Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions handle the "Number not available due to interworking" case better in sig_pri.c. This change is possible because the minimum required libpri version (v1.4.11) has the necessary defines in libpri.h. ........ Merged revisions 360309 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Add missing initialization of update_redirecting in chan_sip.c ........ Merged revisions 360262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-21 14:52 +0000 [r360139] Jonathan Rose * contrib/scripts/install_prereq, /: Update install_prereq script to include missing GSM library for debian amd move SQLite3. (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches: debian_install_prereq.diff uploaded by Andrew Latham (license 5985) ........ Merged revisions 360138 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-21 14:21 +0000 [r360098] Tzafrir Cohen * /, configure, configure.ac: Also detect gmime 2.6 Also detect gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License #5035) ........ Merged revisions 360087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-21 13:28 +0000 [r360088] Matthew Jordan * /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending on the final response to a re-INVITE When Asterisk detects a hangup and cannot send a BYE due to a pending INVITE, it sets the pendingbye flag and waits for the final response to that INVITE. When the response is received, it transmits the BYE. If, however, that INVITE request is a pending re-INVITE, it needs to first send a CANCEL request to terminate the pending re-INVITE. In that circumstance, Asterisk was, in some scenarios, clearing the pendingbye flag after processing the CANCEL request and not checking for a pending BYE when receiving the final 487 response to the INVITE. This patch ensures that if the pendingbye flag is set, it is honored regardless of the nature of the INVITE request currently in flight. (closes issue ASTERISK-19365) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283) Review: https://reviewboard.asterisk.org/r/1807 ........ Merged revisions 360086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-20 20:37 +0000 [r360034] Kinsey Moore * /, apps/app_echo.c: Prevent Echo() from relaying control, null, and modem frames Echo()'s description states that it echoes audio, video, and DTMF except for # while it actually echoes any frame that it receives other than DTMF #. This was causing frame storms in the test suite in some circumstances where Echo() was attached to both ends of a pair of local channels and control frames were being periodically generated. Echo()'s behavior and description have been modifed so that it only echoes media and non-# DTMF frames. ........ Merged revisions 360033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-20 18:11 +0000 [r359982] Sean Bright * channels/chan_iax2.c: chan_iax2: Emit Port alongside Post in PeerStatus AMI Event. The PeerStatus event for IAX2 channels currently includes a header named Post which should have been Port. So include Port along with Post when emitting the event. We'll remove Post in trunk. 2012-03-20 17:25 +0000 [r359980] Richard Mudgett * main/manager.c, /, include/asterisk/manager.h: Allow AMI action callback to be reentrant. Fix AMI module reload deadlock regression from ASTERISK-18479 when it tried to fix the race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2 object guaranteed that there were no active callbacks that mattered when ast_manager_unregister() was called. Unfortunately, this causes the deadlock situation. The patch stops locking the ao2 object to allow multiple threads to invoke the callback re-entrantly. There is no way to guarantee a module unload will not crash because of an active callback. The code attempts to minimize the chance with the registered flag and the maximum 5 second delay before ast_manager_unregister() returns. The trunk version of the patch changes the API to fix the race condition correctly to prevent the module code from unloading from memory while an action callback is active. * Don't hold the lock while calling the AMI action callback. (closes issue ASTERISK-19487) Reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1818/ Review: https://reviewboard.asterisk.org/r/1820/ ........ Merged revisions 359979 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-16 20:20 +0000 [r359898] Jonathan Rose * /, apps/app_chanspy.c: Prevent chanspy from binding to zombie channels This patch addresses a bug with chanspy on local channels which roughly 50% of the time would create a situation where chanspy can latch onto a zombie channel, keeping the zombie alive forever and causing the channel doing the spying to never be able to hang up. (closes issue ASTERISK-19493) Reported by: lvl Review: https://reviewboard.asterisk.org/r/1819/ ........ Merged revisions 359892 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-16 08:24 +0000 [r359810] Alec L Davis * /, channels/sip/include/sip.h: Missed lastinvite CSeq int to uint32_t change from Review: https://reviewboard.asterisk.org/r/1699/ ........ Merged revisions 359809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-15 19:06 +0000 [r359694-359707] Matthew Jordan * /, main/utils.c: Fix remotely exploitable stack overflow in HTTP manager There exists a remotely exploitable stack buffer overflow in HTTP digest authentication handling in Asterisk. The particular method in question is only utilized by HTTP AMI. When parsing the digest information, the length of the string is not checked when it is copied into temporary buffers allocated on the stack. This patch fixes this behavior by parsing out pre-defined key/value pairs and avoiding unnecessary copies to the stack. (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested by: Matt Jordan ........ Merged revisions 359706 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, 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 ........ Merged revisions 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 359656 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-15 18:22 +0000 [r359620] Richard Mudgett * apps/app_dial.c, /, apps/app_queue.c: Add missing connected line macro calls to initial dial for Dial and Queue apps. The connected line interception macros do not get executed when the outgoing channel is initially created and that channel's caller-id is implicitly imported into the incoming channel's connected line data. If you are using the interception macros, you would expect that they get run for every change to a channel's connected line information outside of normal dialplan execution. Review: https://reviewboard.asterisk.org/r/1817/ ........ Merged revisions 359609 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-15 00:53 +0000 [r359454-359559] Russell Bryant * /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in try_transfer() so that the code isn't (potentially) trying to read from it while uninitialized. ........ Merged revisions 359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_gtalk.c: chan_gtalk: Fix potential use of uninitialized variable. Avoid potential use of idroster in gtalk_alloc() before it has been initialized. ........ Merged revisions 359508 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_chanisavail.c: app_chanisavail: Fix use of uninitialized variable. Ensure that status is set before it is used by resetting it during each loop iteration. This could have resulted in incorrect results from this app. ........ Merged revisions 359486 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/udptl.c, /: udptl: Ensure fec[] in udptl_build_packet() is initialized. Scan results indicated that this array could be used uninitialized. At a quick look, it looks correct. In any case, initializing it is a Good Thing (tm). ........ Merged revisions 359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * include/asterisk/app.h, /: app.h: Always initialize AST_DECLARE_APP_ARGS(). This patch ensures that the struct defined by AST_DECLARE_APP_ARGS() is always fully initialized. I'm not sure if this fixes any real bugs, but it silences a bunch of warnings from coverity, and is generally a good thing to do anyway. ........ Merged revisions 359452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-14 22:28 +0000 [r359453] Richard Mudgett * main/channel.c, /, channels/chan_agent.c, include/asterisk/channel.h: Fix deadlock potential with some ast_indicate/ast_indicate_data calls. Calling ast_indicate()/ast_indicate_data() with the channel lock held can result in a deadlock with a local channel because of how local channels need to avoid deadlock. ........ Merged revisions 359451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-14 17:42 +0000 [r359358] Matthew Jordan * /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to missed resynchronizations When a change in time occurs, such that the timestamps associated with frames being placed into an adaptive jitter buffer (implemented in jitterbuf.c) are significantly different then the previously inserted frames, the jitter buffer checks to see if it needs to be resynched to the new time frame. If three consecutive packets break the threshold, the jitter buffer resynchs itself to the new timestamps. This currently only occurs when history is calculated, and hence only on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other hand, are never passed to the history calculations. Because of this, if the jump in time is greater then the maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are dropped and no resynchronization occurs. Alterntively, if the overfill logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the buffer, but with a time reference that is not applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads from the jitter buffer reach the errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch threshold. Note that this only impacts chan_iax2, as other consumers of the adaptive jitter buffer use the abstract jitter buffer API, which does not use JB_TYPE_CONTROL frames. Review: https://reviewboard.asterisk.org/r/1814/ (closes issue ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722) ........ Merged revisions 359356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-14 17:24 +0000 [r359355] Richard Mudgett * apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and forked calls generating warnings for voice frames. When connected line support was added, the wait_for_answer() variable single changed its meaning slightly. Unfortunately, the places where single was used did not necessarily get updated to reflect that change. Also audio/video frames were sent to all forked calls when the endpoints were never made compatible. * Don't pass audio/video media frames when the channels have not been made compatible. * Added handling of AST_CONTROL_SRCCHANGE to app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also pass a requested MOH class. (closes issue ASTERISK-16901) Reported by: Chris Gentle (closes issue ASTERISK-17541) Reported by: clint Review: https://reviewboard.asterisk.org/r/1805/ ........ Merged revisions 359344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-14 10:54 +0000 [r359051-359260] Russell Bryant * include/asterisk/logger.h, /, main/logger.c: Fix bogus reads/writes of console log levels in asterisk.c This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact that logger.c implements 32 log levels (because of the custom log level stuff). asterisk.c uses this define to size an array of levels per remote console. This array is modified in ast_console_toggle_loglevel(), which is called by the "logger set level" CLI command. While the documentation for the CLI command doesn't make it terribly obvious, you can use this CLI command to toggle a custom log level on a remote console, as well. However, doing so led to an invalid array index in asterisk.c. This array is read from any time a log message is written to a console. So, all custom log level messages resulted in a bogus read if a remote console was connected. ........ Merged revisions 359259 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid reads/writes due to incorrect sizeof(). These few places in the code used sizeof() on h_addr in struct hostent. This is sizeof(char *). The correct way to get the size of this address is to use h_length. This error would result in reads/writes of 8 bytes instead of 4 on 64-bit machines. ........ Merged revisions 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/sched.c, /: Fix inaccurate sizeof() in sched.c. This code just needed sizeof(int), not sizeof(int *). ........ Merged revisions 359157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, utils/astman.c: Fix incorrect sizeof() in astman. ........ Merged revisions 359116 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, res/res_crypto.c: Fix incorrect usage of sizeof() in res_crypto. In this case, just remove the memset(). There was a redundant memset that is done correctly just 2 lines later. ........ Merged revisions 359110 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi. ........ Merged revisions 359088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/features.c: Fix incorrect sizeof() usage in features.c. This didn't actually result in a bug anywhere, luckily. The only place where the result of these memcpys was used is in app_dial, and the only field that it read out of ast_call_feature was the first one, which is an int, so these memcpys always copied just enough to avoid a problem. ........ Merged revisions 359069 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final(). ........ Merged revisions 359059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/pbx.c, /: Don't use a buffer after it goes out of scope. 's' is set to 'workspace'. Make sure 'workspace' doesn't go out of scope while the reference to it via 's' is still used. ........ Merged revisions 359056 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * res/res_ais.c, /, res/ais/clm.c, res/ais/evt.c, res/ais/ais.h: Dump cache of published events when a node joins the cluster. Also use a more reliable method for stopping the poll() thread. ........ Merged revisions 359053 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_usbradio.c (removed), /, channels/xpmr (removed), build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, apps/app_rpt.c (removed): Remove chan_usbradio and app_rpt. These modules are being maintained outside of the tree and have been for a long time now, so it doesn't make sense to keep them here. Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged revisions 359050 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-13 20:36 +0000 [r358944-358989] Terry Wilson * /, main/features.c: Fix setting CDR variables in the hangup extension A previous CDR fix for setting CDR variables during a bridge via custom dialplan features broke setting CDR variables in the hangup extension. This patch fixes the issue. Review: https://reviewboard.asterisk.org/r/1794/ ........ Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * include/asterisk/devicestate.h, /, channels/chan_sip.c, tests/test_devicestate.c, main/devicestate.c: Make hints for invalid SIP devices return Unavail, not idle This patch drastically simplifies the device state aggegation code. The old method was not only overly complex, but also made it impossible to return AST_DEVICE_INVALID from the aggregation code. The unit test update is as a result of fixing that bug. The SIP change stems from a bug introduced by removing a DNS lookup for hostname-based SIP channels. (closes issue ASTERISK-16702) Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged revisions 358943 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-13 16:58 +0000 [r358811-358860] Tilghman Lesher * /, UPGRADE.txt, CHANGES: Requested changes documenting the fixed AEL functionality. ........ Merged revisions 358859 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c, utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c: Enable macros in 1.8 to find the next highest "h" extension in a context, like in 1.4. This change restores functionality that was present in 1.4, when AEL macros were implemented with the Macro dialplan application. Macros are fraught with functionality issues, because they consume a large portion of the underlying application stack. This limits the ability of AEL users to call many layers of subroutines, an issue which Gosub does not have (originally tested to 100,000 levels deep). Therefore, starting in 1.6.0, AEL macros were implemented with Gosub. However, there were some implicit behaviors of Macro, which were not replicated at the same time as with the transition to Gosub, one of which is documented in the related issue. In particular, the "h" extension is designed to execute not in the Macro context, but in the topmost calling context. Due to legacy issues with a misapplied bugfix many years ago, when a macro exited in 1.4, it looks in all calling contexts, bubbling up from the deepest level until it finds an "h" extension. Since AEL hides the complexity of the underlying dialplan logic from the AEL programmer, it's reasonable to assume that this behavior should not change in the transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we break working AEL configurations in the transition to Asterisk 1.8 LTS. This fix is the result, which implements a search for the "h" extension in all calling Gosub contexts. Fixes ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff (License #5003) by Tilghman Lesher (with slight modifications for 1.8) Tested by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1776/ ........ Merged revisions 358810 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-08 16:50 +0000 [r358644] Jonathan Rose * /, channels/chan_sip.c: Make transfer not ignore port information with SIP. Attempting to transfer with SIP to an address like 1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from the host string and ignored. This simply keeps chan_sip from cutting off the port number during these kinds of transfers. (closes issue ASTERISK-19321) Reported by: Federico Alves Review: https://reviewboard.asterisk.org/r/1790/diff/#index_header ........ Merged revisions 358643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-07 18:28 +0000 [r358531] Richard Mudgett * /, channels/sig_ss7.c: Change directly setting _softhangup in sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue ASTERISK-19372) ........ Merged revisions 358530 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-07 16:13 +0000 [r358485] Sean Bright * /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the number of samples set properly. If the wctc4xxp returns more than a single packet, we need to update the number of samples in the returned frame accordingly. Acked-by: Shaun Ruffell ........ Merged revisions 358484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-07 15:17 +0000 [r358436-358441] Terry Wilson * /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * cel/cel_odbc.c, /, cdr/cdr_adaptive_odbc.c: Add detection for ODBC WCHAR fields Without detecting these types, cel_odbc blows up when the character set for the table is utf8. This also wraps cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR #ifdef seen in other parts of the code. ........ Merged revisions 358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-06 17:46 +0000 [r358261-358378] Richard Mudgett * channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing calls on FXS ports. * Fix referencing the wrong variable in chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for compiling with -Wshadow and finding this bug. ........ Merged revisions 358377 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/sig_ss7.c: Drop SS7 call if not connected yet when INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should clear a failed call as soon as possible. * Made SS7 hangup a call immediately if it has not connected yet for INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate inband tone. (closes issue ASTERISK-19372) Reported by: Igor Nikolaev ........ Merged revisions 358278 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c: Setup DSP when SS7 call is connected or early media is available. Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel that requires out-of-band DTMF will not work. * Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c. The new call converts conditionaled out unconverted code and shows that the code really did something useful. * Improved some chan_dahdi DTMF debug messages to help track DTMF handling. (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........ Merged revisions 358260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-05 18:58 +0000 [r358215] Jonathan Rose * main/manager.c, /: Eliminate double close of file descriptor in manager.c The process_output function in manager.c attempted to call fclose and close immediately afterwards. Since fclose implies close, this resulted in a potential double free on file descriptors. This patch changes that behavior and also adds error checking to fclose and close depending on which was deemed necessary. Also error messages. Thanks to Rosen Iliev for pointing out the location of the problem. (closes issue ASTERISK-18453) Reported By: Jaco Kroon Review: https://reviewboard.asterisk.org/r/1793/ ........ Merged revisions 358214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-05 16:42 +0000 [r358163] Joshua Colp * /, channels/chan_sip.c: Defer sending the connected line reinvite if a reinvite is already in progress. (issue ASTERISK-19355) Reported by: tomaso (closes issue AST-825) ........ Merged revisions 358162 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-05 15:59 +0000 [r358116] Kinsey Moore * /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors Asterisk was not setting pendinginvite in the upper half of handle_request_invite such that the 4xx was retransmitted repeatedly even though an ack was received for every retransmission. (closes issue ASTERISK-19303) Patch-by: Jeremiah Gowdy ........ Merged revisions 358115 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 23:28 +0000 [r357987-358033] Terry Wilson * channels/chan_usbradio.c, /, channels/xpmr/xpmr.c: Fix unused-but-set-variable warnings All of these were pretty obviously unused. Some were unused because the code that used them was #if 0'd. In those cases, I just commented out the unused-but-set variables. ........ Merged revisions 358029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Correct some set-but-unused variable warnings in the mISDN library. (from kpfleming's commit to trunk r356292) ........ Merged revisions 358011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev mode x=++x and x=x=1? Really? ........ Merged revisions 357986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 21:03 +0000 [r357941] Kinsey Moore * /, main/ccss.c, tests/test_event.c, main/event.c, include/asterisk/strings.h: Fix case-sensitivity for device-specific event subscriptions and CCSS This change fixes case-sensitivity for device-specific subscriptions such that the technology identifier is case-insensitive while the remainder of the device string is still case-sensitive. This should also preserve the original case of the device string as passed in to the event system. CCSS is the only feature affected as it is the only consumer of device-specific event subscriptions. The second part of this patch addresses similar case-sensitivity issues within CCSS itself that prevented it from functioning correctly after the fix to the events system. This adds a unit test to verify that the event system works as expected. (closes issue ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ ........ Merged revisions 357940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 18:37 +0000 [r357895] Richard Mudgett * main/channel.c, /, channels/sig_pri.c: Remove ISDN hold restriction for non-bridged calls. The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett ........ Merged revisions 357894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 15:59 +0000 [r357812] Sean Bright * /, channels/chan_iax2.c: The default value for mohinterpret is the empty string, so when resetting to default values don't explicitly set the value to "default." ........ Merged revisions 357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 15:50 +0000 [r357810] Richard Mudgett * /, apps/app_chanspy.c: Fix channel reference leak in ChanSpy. * Fix next_channel() channel reference leak in ChanSpy. (closes issue ASTERISK-19461) Reported by: Irontec Patches: app_chanspy_iteartor_next_unref.patch (license #6213) patch uploaded by Irontec (issue ASTERISK-17515) ........ Merged revisions 357809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-02 01:05 +0000 [r357762] Mark Michelson * main/channel.c, /: Fix race condition that can cause important control frames (such as a hangup) to be missed. This takes two actions. 1. Move the reading of the alertpipe in __ast_read() to immediately before the removal of frames from the readq. This means we won't do something silly like read from the alertpipe, then ignore the fact that there's a frame to get from the readq since channel's fdno is the AST_TIMING_FD. 2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because if the rate is 0 and the timingfunc is NULL, it means that the channel's timing fd is being invalidated, so any pending reads should not occur. This may actually solve more issues than the referenced one below, but it's not known at this time for sure. (closes issue ASTERISK-19223) reported by Frank-Michael Wittig Review: https://reviewboard.asterisk.org/r/1779 ........ Merged revisions 357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-01 14:18 +0000 [r357667] Kinsey Moore * /, main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a port of 0 In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. (closes issue ASTERISK-19430) ........ Merged revisions 357665 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-29 20:39 +0000 [r357576-357620] Walter Doekes * include/asterisk/stringfields.h, main/utils.c: Update stringfield documentation for removed second va_list in favor of va_copy. In r320946, the second va_list that was passed to ast_string_field_build_va and friends, was removed. This patch updates the documentation to reflect that. * apps/app_dial.c, /: Fix copying of CDR(accountcode) to local channels. In r203638, during the addition of the Channel Event Logging, in mid-2009, this got broken in trunk and ended up in asterisk 1.8 and higher. This fixes so the CDR(accountcode) from the calling channel is available to dialed channels again as well as showing up properly in the CDR's. (closes issue ASTERISK-19384) Patches: accountcode.patch (License #6033) by jamicque Review: https://reviewboard.asterisk.org/r/1775/ Reviewed by: Richard Mudgett ........ Merged revisions 357575 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 22:29 +0000 [r357458-357497] Jonathan Rose * /, configs/sip.conf.sample, UPGRADE-1.8.txt: Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) ........ Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, cdr/cdr_adaptive_odbc.c: Add additional character type types to supported data types for cdr_adaptive_odbc The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so this patch adds those along with some other character types to the list of types cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really UTF8 characters as much as it was a failure to respond to the exact type that was declared/in use on that database. (closes issue ASTERISK-19334) Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236) ........ Merged revisions 357455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 21:21 +0000 [r357421] Tilghman Lesher * /, apps/app_stack.c: Correctly reset the dialplan priority. When the stack frame is allocated, we save the address to which we should return, when the Gosub returns. However, if we just want to restore the priority, then we need to subtract 1 before setting it. Otherwise, when a Gosub goes to a nonexistent address, it will skip a priority in the dialplan. This is because when we return from an application, the PBX increments the priority for us. ........ Merged revisions 357416 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 20:58 +0000 [r357408] Richard Mudgett * /, channels/sig_pri.c: Use more reasonable cause code when rejecting incoming call waiting calls. (closes issue ASTERISK-19397) Reported by: Birger Harzenetter Patches: nochannel-cause.patch (license #5870) patch uploaded by Birger Harzenetter ........ Merged revisions 357407 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 20:42 +0000 [r357357-357405] Jonathan Rose * UPGRADE.txt: revision 357386 -- oops, accidentally made it 10.3 to 10.4 instead of 10.2 to 10.3 (issue ASTERISK-19352) reported by: jamicque * /, UPGRADE.txt, UPGRADE-1.8.txt: Moves UPGRADE.txt notes from r357356 to a new section specific to 1.8.12 (issue ASTERISK-19352) reported by: jamicque ........ Merged revisions 357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, UPGRADE-1.8.txt: Adds UPGRADE.txt notes to r357266 indicating changes to transport option (issue ASTERISK-19352) Reported by: jamicque ........ Merged revisions 357356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 19:35 +0000 [r357353] Richard Mudgett * /, apps/app_page.c: Remove dupliate 'i' option table entry in app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei Patches: app_page-duplicate-i-option.patch (license #5027) patch uploaded by Makoto Dei ........ Merged revisions 357352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 18:51 +0000 [r357318] Mark Michelson * channels/sip/security_events.c: Add a security event for the case where fake authentication challenge is sent. 2012-02-28 18:11 +0000 [r357271] Jonathan Rose * /, channels/chan_sip.c: Changes transport option in sip.conf so that using multiple instances doesn't stack. Prior to this patch, Using "transport=" multiple times would cause them to add to one another like allow/deny. This patch changes that behavior to simply use the transport option specified last. Also, if no transport option is applied now, the default will automatically be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674) Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header ........ Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-28 14:46 +0000 [r357213] Kevin P. Fleming * /, Makefile.rules: Make COMPILE_DOUBLE magic actually work. The build system has some special magic to ensure that if Asterisk is built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled with the optimizer enabled (even though the result will be thrown away), because the compiler is able to find a great deal of coding errors and bugs as a result of running its optimizers. Unfortunately at some point this mode got broken, and the 'throwaway' compile of the code was no longer done with the optimizer enabled. This patch corrects that problem. ........ Merged revisions 357212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-29 Asterisk Development Team * Asterisk 10.3.0 Released. 2012-03-26 Asterisk Development Team * Asterisk 10.3.0-rc3 Released. * AST-2012-003 * AST-2012-002 * /main/manager.c, include/asterisk/manager.h: Fix AMI deadlock regression by allowing AMI action callback to be reentrant Fix AMI module reload deadlock from ASTERISK-18479 when it tired to fix the race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13874 broken by ASTERISK-17785 change. Locking the ao2 object guaranteed that there were no active callbacks that mattered when ast_manager_unregister() was called. Unfortunately, this causes the deadlock situation. The path stops locking the ao2 object to allow multiple threads to invoke the callback re-entrantly. There is no way to guarantee a module unload will not crash because of an active callback. The code attempts to minimize the chance with the registered flag and the maximum 5 second delay before ast_manager_unregister() returns. The trunk version of the patch changes the API to fix the race condition correctly to prevent the module code from unloading from memory while an action callback is active. * Don't hold the lock while calling the AMI action callback. (closes issue ASTERISK-19487) Reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1818/ 2012-03-06 Asterisk Development Team * Asterisk 10.3.0-rc2 Released. * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a port of 0. In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. 2012-01-30 22:16 +0000 [r353369-353321] Alec L Davis * channels/sip/include/dialog.h, /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 353320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ ........ * /, channels/chan_sip.c: Merged revisions 353368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan 2012) | 2 lines prevent debug messsges displaying -ve Cseq numbers. Missed in R353320 ........ 2012-01-30 23:28 +0000 [r353397] Terry Wilson * main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h: Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it anytime an address resolves to something different. There are a couple of issues with this. First, the ast_sockaddr is usually the address of an ast_sockaddr inside a refcounted struct and we never bump the refcount of those structs when using dnsmgr. This makes it possible that a refresh could happen after the destructor for that object is called (despite ast_dnsmgr_release being called in that destructor). Second, the module using dnsmgr cannot be aware of an address changing without polling for it in the code. If an action needs to be taken on address update (like re-linking a SIP peer in the peers_by_ip table), then polling for this change negates many of the benefits of having dnsmgr in the first place. This patch adds a function to the dnsmgr API that calls an update callback instead of blindly updating the address itself. It also moves calls to ast_dnsmgr_release outside of the destructor functions and into cleanup functions that are called when we no longer need the objects and increments the refcount of the objects using dnsmgr since those objects are stored on the ast_dnsmgr_entry struct. A helper function for returning the proper default SIP port (non-tls vs tls) is also added and used. This patch also incorporates changes from a patch posted by Timo Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/ ........ Merged revisions 353371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-31 17:21 +0000 [r353463] Richard Mudgett * main/manager.c, /, include/asterisk/channel.h: Fix memory leak in error paths for action_originate(). * Fix memory leak of vars in error paths for action_originate(). * Moved struct fast_originate_helper tech and data members to stringfields. * Simplified ActionID header handling for fast_originate(). * Added doxygen note to ast_request() and ast_call() and the associated channel callbacks that the data/addr parameters should be treated as const char *. Review: https://reviewboard.asterisk.org/r/1690/ ........ Merged revisions 353454 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-01 00:00 +0000 [r353503] Terry Wilson * res/res_calendar.c, /: Allow res_calendar to be unloaded The calendaring tech modules depend on res_calendar and initially res_calendar just bumped the use count so that it couldn't be unloaded. res_calendar can potentially create many threads and I've seen issues where the Asterisk shutdown has failed where it looked like these threads could be the culprit. This patch adds unload support for res_calendar. Unloading res_calendar will also unload the dependant tech modules as well. (closes issue ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/ ........ Merged revisions 353502 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-01 15:05 +0000 [r353551] Matthew Jordan * /, contrib/init.d/etc_default_asterisk: Added clarification for the VERBOSITY setting to etc_default_asterisk Clarified that using the VERBOSITY setting in etc_default_asterisk is the same as using the -v command line switch, which causes Asterisk to launch in console mode. (closes issue ASTERISK-17030) Reported by: Jonas ........ Merged revisions 353550 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-01 15:51 +0000 [r353599] Sean Bright * /, include/asterisk/audiohook.h: Resolve an overlap in the ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused unintended side effects. This patch moves AST_AUDIOHOOK_TRIGGER_WRITE, and updates AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention. This will affect existing modules that use these flags, so be sure to recompile as necessary. (closes issue ASTERISK-19246) Reported by: feyfre ........ Merged revisions 353598 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-01 21:16 +0000 [r353771-353721] Jonathan Rose * /, channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers for various functions in chan_sip There are a number of cleaner looking wrappers for ast_sockaddr_stringify_fmt available which are slightly more readable than using a direct call to ast_sockaddr_stringify_fmt. This patch switches a number of those calls in chan_sip to use those wrappers and is generally harmless. (Closes issue ASTERISK-16930) Reported by: Michael L. Young Patches: chan_sip-broken-registration-1.8.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 353720 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Fix sip show peers port output, align columns, and fix ami port output. A previous patch I committed from ASTERISK-16930 unexpectedly changed some output for the AMI action "sippeers" which this patch changes back. Also, this aligns the output for the cli command "sip show peers" and fixes another issue that patch introduced by using ast_sockaddr_stringify calls multiple times without immediately using the pointer. I also went ahead and did a little janitorial work to clean up whitespace in _sip_show_peers. (issue ASTERISK-16930) (closes issue ASTERISK-19281) Reported by: Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by Walter Doekes (license 5674) ........ Merged revisions 353769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-02 18:48 +0000 [r353820] Mark Michelson * configs/http.conf.sample, main/manager.c, /, main/http.c, configs/manager.conf.sample, include/asterisk/manager.h: Fix TLS port binding behavior as well as reload behavior: * Removes references to tlsbindport from http.conf.sample and manager.conf.sample * Properly bind to port specified in tlsbindaddr, using the default port if specified. * On a reload, properly close socket if the service has been disabled. A note has been added to UPGRADE.txt to indicate how ports must be set for TLS. (closes issue ASTERISK-16959) reported by Olaf Holthausen (closes issue ASTERISK-19201) reported by Chris Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas Review: https://reviewboard.asterisk.org/r/1709 ........ Merged revisions 353770 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-02 20:11 +0000 [r353868] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Restore the 'w' modifier support for ISDN spans. Dial(DAHDI/g0/1234w888) This feature also causes the sending complete ie to be sent for switch types that do not automatically send the ie. (EuroISDN/ETSI) The main difference between dialing Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the sending of the sending complete ie. (closes issue ASTERISK-19176) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 353867 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-02 22:27 +0000 [r353916] Kinsey Moore * /, channels/chan_sip.c: Ensure entering T.38 passthrough does not cause an infinite loop After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is shut down and removed. If the descriptor happened to have data ready when the removal occured then Asterisk would go into an infinite loop trying to read data that it can never actually access. This change disables the audio RTCP file descriptor for the duration of the T.38 transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan Vrban ........ Merged revisions 353915 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-03 16:22 +0000 [r354000-353962] Jonathan Rose * res/res_fax.c: Fixes a segfault occuring when performing attended transfer with FAXOPT(gateway)=yes (closes issue ASTERISK-19184) Reported by: Alexandr * /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due to r335976 Bad locking order was added to chan_agent to prevent segfaults from having no locking in a patch by irroot. This patch addresses the bad locking order by releasing locks before getting the right locking order to stop deadlocks from occuring when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1708/ ........ Merged revisions 353999 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-06 17:31 +0000 [r354217-354119] Richard Mudgett * /, main/features.c: Add missing headers to AMI UnParkedCall event to uniquely identify the call. The AMI UnParkedCall event was missing the Parkinglot and Uniqueid headers that the AMI ParkedCall event contains. (closes issue ASTERISK-19240) Reported by: Michael Yara ........ Merged revisions 354116 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add extension" command. * Documented dialplan add extension ,,)> format. * Allow acceptance of command without the app-data value. There are many applications that do no need any parameters so it is silly to require that field for all commands. * Fixed a couple ast_malloc/ast_free mismatches with ast_add_extension2() calls. (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested by: rmudgett ........ Merged revisions 354216 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-07 15:19 +0000 [r354270] Jonathan Rose * /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload for cdr_pgsql. Prior to this patch, attempts to reload cdr_pgsql.so would cause the column list to keep its current data and then add a second copy during the reload. This would cause attempts to log the CDR to the database to fail. This patch also cleans up some unnecessary null checks for ast_free and deals with a few potential locking problems. (closes issue ASTERISK-19216) Reported by: Jacek Konieczny Review: https://reviewboard.asterisk.org/r/1711/ ........ Merged revisions 354263 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-07 21:17 +0000 [r354349] Terry Wilson * /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql: Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing instead of "" 2. Don't set ipaddr or port to the string "(null)" when they are empty 3. Add missing required fields, set default for lastms to 0, and modify the length of the ipaddr field to 45 in the Postgresql realtime.sql file. (closes issue ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/ ........ Merged revisions 354348 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 02:25 +0000 [r354493] Russell Bryant * main/channel.c, /: Remove some unnecessary locking from ast_hangup(). This patch removes some unnecessary locking of the channels container in ast_hangup(). The reason this came up is that this lock can very quickly block the entire system. If any of the channel cleanup code decides to block, it causes a problem for the whole system. For example, when audiohooks get destroyed, if that blocks for a while waiting on the mixmonitor thread to exit because it's busy blocking on some I/O, it causes a problem for many other threads in the meantime. Review: https://reviewboard.asterisk.org/r/1712/ ........ Merged revisions 354492 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 02:54 +0000 [r354496] Richard Mudgett * apps/app_parkandannounce.c, /: Fix crash in ParkAndAnnounce. Well, thats embarrasing. I forgot to initialize the caller_id storage. (closes issue ASTERISK-19311) Reported by: tootai Tested by: rmudgett ........ Merged revisions 354495 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 16:35 +0000 [r354543] Matthew Jordan * /, channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF events. When this occurred, the comparison of the character buffer containing the event code was changed such that the buffer was first compared again '0' and '9' to determine if it was numeric. Unfortunately, since the first character in the buffer will typically be '1' in the case of non-numeric event codes (10-16), this caused those codes to be converted to a DTMF event of '1'. This patch fixes that, and cleans up handling of both application/dtmf-relay and application/dtmf content types. Review: https://reviewboard.asterisk.org/r/1722/ (closes issue ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........ Merged revisions 354542 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 17:04 +0000 [r354546] Mark Michelson * /, res/res_fax.c: Adding reload support to res_fax.so (closes issue ASTERISK-16712) reported by Frank DiGennaro Review: https://reviewboard.asterisk.org/r/1713 ........ Merged revisions 354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 17:08 +0000 [r354548] Matthew Jordan * /, channels/chan_sip.c: Clean-up of minor formatting issues in r354542/3/4 rmudgett pointed out some formatting issues in the check-in for ASTERISK-19290. This cleans those up. Review: https://reviewboards.asterisk.org/r/1722/ ........ Merged revisions 354547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 19:54 +0000 [r354703-354656] Kinsey Moore * /, main/config.c: Make the config parser remove escaping backslashes The config parser in Asterisk does not currently remove a backslash that is used to escape a semicolon which would otherwise be interpreted as the start of a comment. The change here causes that backslash to be removed, but does not create a real escape system in the config parser. The biggest complication with a real escape system would be breaking existing configs everywhere (parsing \\ as \ and breaking on escaped non-semicolon characters) even though it would be the "right" way to do things. (closes issue ASTERISK-17121) Review: https://reviewboard.asterisk.org/r/1724/ ........ Merged revisions 354655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Fix parsing of SIP headers where compact and non-compact headers are mixed Change parsing of SIP headers so that compactness of the header no longer influences which header will be chosen. Previously, a non-compact header would be chosen instead of a preceeding compact-form header. (closes issue ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/ ........ Merged revisions 354702 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-09 22:03 +0000 [r354750] Terry Wilson * /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge is torn down CDRs cannot be modified after a bridge is torn down, (e.g. after Dial() returns) even though the CDR() function may be called. Since modifying the CDR code to change this behavior could very easily break all kinds of things, this patch just documents this limitation. (closes issues ASTERISK-16923) Review: https://reviewboard.asterisk.org/r/1720/ ........ Merged revisions 354749 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-10 18:05 +0000 [r354836] Richard Mudgett * main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting to the same exten and context. The astman_get_header() never returns NULL so the check by the code for NULL would never fail. (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches: 0018325.patch (license #6116) patch uploaded by Nuno Borges (modified) ........ Merged revisions 354835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-10 22:00 +0000 [r354890] Jason Parker * apps/app_voicemail.c, /: Fix a voicemail memory leak with heard/deleted messages. open_mailbox() was changed quite a long time ago to allocate this memory. close_mailbox() should have been changed to be responsible for freeing it. ........ Merged revisions 354889 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-13 16:41 +0000 [r354938] Joshua Colp * apps/app_confbridge.c: Don't try to play sound files that do not exist. (closes issue ASTERISK-19188) Reported by: slesru 2012-02-13 17:24 +0000 [r354959] Richard Mudgett * res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix reconnecting to pgsql database after connection loss. There can only be one database connection in res_config_pgsql just like res_config_sqlite. If the connection is lost, the connection may not get reestablished to the same database if the res_pgsql.conf and extconfig.conf files are inconsistent. * Made only use the configured database from res_pgsql.conf. * Fixed potential buffer overwrite of last[] in config_pgsql(). (closes issue ASTERISK-16982) Reported by: german aracil boned Review: https://reviewboard.asterisk.org/r/1731/ ........ Merged revisions 354953 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-13 19:51 +0000 [r355010] Joshua Colp * /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute at a time as otherwise they would share the same common local context list. (closes issue AST-758) ........ Merged revisions 355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-13 22:03 +0000 [r355057] Richard Mudgett * pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file execution. Since the dir timestamp is available at one second resolution, we cannot know if it was updated within the same second after we scanned it. Therefore, we will force another scan if the dir was just modified. * Changed to force another scan if the directory was just modified. (closes issue ASTERISK-19081) Reported by: Knut Bakke Review: https://reviewboard.asterisk.org/r/1688/ ........ Merged revisions 355056 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 09:49 +0000 [r355137] Alexandr Anikin * addons/chan_ooh323.c, /: call manager_event only if there is not null channel structure (Closes issue ASTERISK-19298) Reported by: robinfood Patches: issue19298.patch uploaded by may213 (License #5415) ........ Merged revisions 355136 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 13:33 +0000 [r355183] Sean Bright * /, channels/chan_iax2.c: Clear the high order bit from the destination call number before sending. send_apathetic_reply takes the incoming frame's source call number as the destination call number for the outgoing frame. If the incoming frame was a full frame, then the high order bit of the source call number is set and will be interpreted as a retransmit when sent back out as the destination call number. ........ Merged revisions 355182 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 15:53 +0000 [r355229] Jason Parker * /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3 CDRs by default in sample configs. ........ Merged revisions 355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 16:27 +0000 [r355271] Mark Michelson * /, channels/chan_sip.c: Properly invert the return of a strncmp call. This was causing identification that should have been made private to be public. (closes issue AST-814) reported by Patrick Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson (license 5430) ........ Merged revisions 355268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-14 18:14 +0000 [r355375-355320] Richard Mudgett * /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock in cel_sqlite_custom reload. (closes issue ASTERISK-19356) Reported by: Alex Villacis Lasso Patches: asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch (license #5617) patch uploaded by Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1740/ ........ Merged revisions 355319 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configure, include/asterisk/autoconfig.h.in, configure.ac, formats/format_ogg_vorbis.c: Fix voicemail problems when using ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file format because it did not implement the seek and tell format callbacks among other problems. Since we were already using the libvorbis and libvorbisenc libraries we can use libvorbisfile as it is also part of the vorbis library package. * Made use the libvorbisfile to handle the ogg/vorbis file stream. The format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile. (closes issue ASTERISK-16926) Reported by: sque Patches: ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque ........ Merged revisions 355365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-15 17:25 +0000 [r355530-355449] Sean Bright * /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally intended. Back in r646, TRUNK_CALL_START was added and defined as 0x4000. That same value was also hard-coded in one part of the IAX2 code instead of using the #define. TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but the hard-coded usage was never updated to match. This patch fixes that. ........ Merged revisions 355448 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_iax2.c: Only use maxtrunkcall and maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified. These variables are only accessed from the IAX_OLD_FIND path, so there is no reason to keep them updated otherwise. ........ Merged revisions 355458 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_iax2.c: When IAX2 debugging is enabled, make sure to log 'apathetic' messages too. ........ Merged revisions 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-16 18:32 +0000 [r355620-355575] Richard Mudgett * /, res/res_monitor.c: Fix AMI Monitor action without File header converting channel name into filename. * Fix potential Solaris crash if Monitor application has a urlbase and no fname_base option. ........ Merged revisions 355574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configure, include/asterisk/autoconfig.h.in, autoconf/ast_c_declare_check.m4 (added), configure.ac, formats/format_ogg_vorbis.c: Fix compile problem when old version of libvorbisfile v1.1.2 is used. The principle difference between libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the configure script to detect if libvorbisfile.h declares OV_CALLBACKS_NOCLOSE. * Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile. (closes issue ASTERISK-19370) Reported by: Jonn Taylor ........ Merged revisions 355608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-16 20:01 +0000 [r355623] Sean Bright * /, main/audiohook.c: Revert a change to audio_audiohook_write_list that had no affect. When I made this change initially, I was under the false impression that the audiohooks structure remained on the channel after all of the hooks had been detached. This is not the case, ast ast_read takes care of removing the audiohooks structure if the lists are empty. ........ Merged revisions 355622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-17 19:06 +0000 [r355733] Mark Michelson * /, channels/chan_sip.c: Fix regressions with regards to route-set creation on early dialogs. This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response. (closes issue ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034) Review: https://reviewboard.asterisk.org/r/1749 ........ Merged revisions 355732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-17 19:34 +0000 [r355794-355747] Sean Bright * /, channels/chan_iax2.c: Pass the correct value to ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but this value is in milliseconds while ast_timer_set_rate() expects the rate argument to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead. With a default of 20ms, this change makes IAX2 send trunk packets every 20ms instead of every 50ms. Tracked down by myself and Bob Wienholt. ........ Merged revisions 355746 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow trunkfreq to be greater than 1000ms. ........ Merged revisions 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-18 07:58 +0000 [r355851] Alec L Davis * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c, channels/sig_ss7.c: push 'outgoing' flag from sig_XXX up to chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-18 17:02 +0000 [r355896-355895] Paul Belanger * /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 355839 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /: Revert commit 2012-02-19 17:50 +0000 [r355998-355902] Sean Bright * /, channels/chan_iax2.c: Set the length of the ast_sockaddr, so that we can set it's port later. Without this, the call to ast_sockaddr_set_port a few lines later is a noop. ........ Merged revisions 355901 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_iax2.c: Add some boilerplate documentation for IAXVAR and IAXPEER. ........ Merged revisions 355904 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_dahdi.c, /: Change some debug messages from LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back. ........ Merged revisions 355952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_iax2.c: Remove spurious warning when 'qualifyfreqnotok' is set successfully. (closes issue ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512) ........ Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-21 04:30 +0000 [r356074] Kinsey Moore * main/ccss.c: Add missing newline to ccss state change notification Move along, nothing to see here... 2012-02-21 11:17 +0000 [r356108] Sean Bright * /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output make sense when an IP is passed in. ........ Merged revisions 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-22 14:53 +0000 [r356215] Matthew Jordan * /, channels/chan_sip.c: Merged revisions 356214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines Fix potential buffer overrun and memory leak when executing "sip show peers" The "sip show peers" command uses a fix sized array to sort the current peers in the peers ao2_container. The size of the array is based on the current number of peers in the container. However, once the size of the array is determined, the number of peers in the container can change, as the peers container is not locked. This could cause a buffer overrun when populating the array, if peers were added to the container after the array was created. Additionally, a memory leak of the allocated array would occur if a user caused the _show_peers method to return CLI_SHOWUSAGE. We now create a snapshot of the current peers using an ao2_callback with the OBJ_MULTIPLE flag. This size of the array is set to the number of peers that the iterator will iterate over; hence, if peers are added or removed from the peers container it will not affect the execution of the "sip show peers" command. Review: https://reviewboard.asterisk.org/r/1738/ (closes issue ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283) ........ 2012-02-22 21:18 +0000 [r356297] Terry Wilson * main/loader.c, res/res_calendar.c, /, include/asterisk/calendar.h: Track module use count for res_calendar If the res_calendar module was followed immediately by one of the calendar tech modules and "core stop gracefully" was run, Asterisk would crash. This patch adds use count tracking for res_calendar so that it is unloaded after the tech modules when shutting down gracefully. It is now not possible to unload all the of the calendar modules via "module unload res_calednar.so", but it is still possible to unload them all via "module unload -h res_calendar.so". Review: https://reviewboard.asterisk.org/r/1752/ ........ Merged revisions 356291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-23 03:23 +0000 [r356431-356428] Paul Belanger * /, apps/app_rpt.c: Multiple revisions 356290,356335,356337 ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed, 22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) Review: https://reviewboard.asterisk.org/r/1763/ ........ r356335 | pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2 lines Add back strsep() function for previous commit ........ r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb 2012) | 2 lines Missed one strsep() function ........ Merged revisions 356290,356335,356337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * addons/chan_ooh323.c, /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 356430 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-23 15:40 +0000 [r356476] Mark Michelson * /, channels/chan_sip.c: Fix ACK routing for non-2xx responses. When we send an ACK for a 2xx response to an INVITE, we are supposed to use the learned route set. However, when we receive a non-2xx final response to an INVITE, we are supposed to send the ACK to the same place we initially sent the INVITE. We had been doing this up until the changes went in that would build a route set from provisional responses. That introduced a regression where we would use the learned route set under all circumstances. With this change, we now will set the destination of our ACK based on the invitestate. If it is INV_COMPLETED then that means that we have received a non-2xx final response (INV_TERMINATED indicates a 2xx response was received). If it is INV_CANCELLED, then that means the call is being canceled, which means that we should be ACKing a 487 response. The other change introduced here is setting the invitestate to INV_CONFIRMED when we send an ACK *after* the reqprep instead of before. This way, we can tell in reqprep more easily what the invitestate is prior to sending the ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer patches: ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049) (with some slight modifications prior to commit) ........ Merged revisions 356475 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-23 19:52 +0000 [r356522] Richard Mudgett * /, channels/chan_sip.c, main/features.c: Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension. Custom parking extensions may not be coded such that the first and only extension priority is the Park application. These custom parking extensions will not be recognized as parking extensions. When a call is blind transferred to an extension that is not recognized as a parking extension, the normal blind transfer code causes the transferred channel to start executing dialplan. Calls that get parked in this manner do not know the original channel name that parked the call so the original parker could never be called back if the parked call is not retrieved before the timeout time. The parking space is also announced to the call being parked as a side effect of not knowing the original parking channel. * Fix handling of BLINDTRANSFER channel variable for call parking. * Fixed SIP blind transfer using the wrong dialplan context variable to check for the parking extension. (closes issue ASTERISK-19322) Reported by: aragon Tested by: rmudgett, jparker Review: https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........ Merged revisions 356521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-24 15:07 +0000 [r356651-356605] Matthew Jordan * res/res_srtp.c, channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h, main/rtp_engine.c, /, include/asterisk/rtp_engine.h: Allow SRTP policies to be reloaded Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) ........ Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * res/res_srtp.c, /: Remove srtp_shutdown from res_srtp The patch for ASTERISK-19253 included properly shutting down the libsrtp library in the case of module unload. Unfortunately, not all distributions have the srtp_shutdown call. As such, this patch removes calling srtp_shutdown. ........ Merged revisions 356650 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-24 18:27 +0000 [r356690] Richard Mudgett * /, channels/chan_sip.c, include/asterisk/tcptls.h, channels/sip/include/sip.h: Fix worker thread resource leak in SIP TCP/TLS. The SIP TCP/TLS worker threads were created joinable but noone could join them if they died on their own. * Fix the SIP TCP/TLS worker threads to not be created joinable. * _sip_tcp_helper_thread() only needs one parameter since the pvt parameter is only passed in as NULL and never used. (closes issue ASTERISK-19203) Reported by: Steve Davies Review: https://reviewboard.asterisk.org/r/1714/ ........ Merged revisions 356677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-25 17:22 +0000 [r356798] Matthew Jordan * apps/app_voicemail.c, /: Fix crash in app_voicemail during close_mailbox In r354890, a memory leak in app_voicemail was fixed by properly disposing of the allocated heard/deleted pointers. However, there are situations, particularly when no messages are found in a folder, where these pointers are not allocated and not NULL. In that case, an invalid free would be attempted, which could crash app_voicemail. As there are a number of code paths where this could occur, this patch uses the number of messages detected in the folder before it attempts to free the pointers. This resolves the crash detected in the Asterisk Test Suite's check_voicemail_nominal test. ........ Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-27 15:30 +0000 [r356961] Jonathan Rose * /, res/res_odbc.c: Remove possible segfaults from res_odbc by adding locks around usage of odbc handle (closes issue ASTERISK-19011) Reported by: Walter Doekes Patches: issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch uploaded by Walter Doekes (license 5674) review: https://reviewboard.asterisk.org/r/1719/ review: https://reviewboard.asterisk.org/r/1622/ ........ Merged revisions 356917 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-27 16:05 +0000 [r356964] Terry Wilson * /, main/features.c: Copy CDR variables when set during a bridge This patch makes sure amaflags, accountcode, and userfield get copied to the bridge CDR when set during a bridge (like via a custom feature). (closes issue ASTERISK-16990) Review: https://reviewboard.asterisk.org/r/1721/ ........ Merged revisions 356963 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-02-27 23:36 +0000 [r357095] Richard Mudgett * main/channel.c, /: Fix callerid of Originated calls. Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix ........ Merged revisions 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-03-05 Asterisk Development Team * Asterisk 10.2.0 Released. 2012-03-01 Asterisk Development Team * Asterisk 10.2.0-rc4 Released. * main/acl.c: Prevent outbound SIP NOTIFY packets from displaying a port of 0. In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. 2012-02-28 Asterisk Development Team * Asterisk 10.2.0-rc3 Released. * main/channel.c: Fix callerid of Originated calls. The callerid of originated calls (independent of mechanism) was not being passed to the outbound channel. This patch fixes that. Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix * channels/chan_sip.c: Fix ACK routing for non-2xx responses. When we send an ACK for a 2xx response to an INVITE, we are supposed to use the learned route set. However, when we receive a non-2xx final response to an INVITE, we are supposed to send the ACK to the same place we initially sent the INVITE. We had been doing this up until the changes went in that would build a route set from provisional responses. That introduced a regression where we would use the learned route set under all circumstances. With this change, we now will set the destination of our ACK based on the invitestate. If it is INV_COMPLETED then that means that we have received a non-2xx final response (INV_TERMINATED indicates a 2xx response was received). If it is INV_CANCELLED, then that means the call is being canceled, which means that we should be ACKing a 487 response. The other change introduced here is setting the invitestate to INV_CONFIRMED when we send an ACK *after* the reqprep instead of before. This way, we can tell in reqprep more easily what the invitestate is prior to sending the ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer patches: ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049) * channels/chan_sip.c: Fix regressions with regards to route-set creation on early dialogs. This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response. (closes issue ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034) Review: https://reviewboard.asterisk.org/r/1749 2012-02-10 Asterisk Development Team * Asterisk 10.2.0-rc2 Released. * channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric codes. In ASTERISK-18924, SIP INFO DTMF handling was changed to account for both lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF events. When this occurred, the comparison of the character buffer containing the event code was changed such that the buffer was first compared against '0' and '9' to determine if it was numeric. Unfortunately, since the first character in the buffer will typically be '1' in the case of non-numeric event codes (10-16), this caused those codes to be converted to a DTMF event of '1'. This patch fixes that, and cleans up handling of both application/dtmf-relay and application/dtmf content types. Review: https://reviewboard.asterisk.org/r/1722/ (closes issue ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan * apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce from uninitialized caller_id storage (closes issue ASTERISK-19311) Reported by: tootai Tested by: rmudgett * channels/chan_agent.c: Fixes deadlocks occuring in chan_agent due to r335976. Bad locking order was added to chan_agent to prevent segfaults from having no locking in a patch by irroot. This patch addresses the bad locking order by releasing locks before getting the right locking order to stop deadlocks from occuring when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1708/ * channels/chan_sip.c: Ensure entering T.38 passthrough does not cause an infinite loop. After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is shut down and removed. If the descriptor happened to have data ready when the removal occured then Asterisk would go into an infinite loop trying to read data that it can never actually access. This change disables the audio RTCP file descriptor for the duration of the T.38 transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan Vrban * channels/chan_sip.c,include/asterisk/dnsmgr.h,main/dnsmgr.c: Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it anytime an address resolves to something different. There are a couple of issues with this. First, the ast_sockaddr is usually the address of an ast_sockaddr inside a refcounted struct and we never bump the refcount of those structs when using dnsmgr. This makes it possible that a refresh could happen after the destructor for that object is called (despite ast_dnsmgr_release being called in that destructor). Second, the module using dnsmgr cannot be aware of an address changing without polling for it in the code. If an action needs to be taken on address update (like re-linking a SIP peer in the peers_by_ip table), then polling for this change negates many of the benefits of having dnsmgr in the first place. 2012-02-01 Asterisk Development Team * Asterisk 10.2.0-rc1 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK1020RCS-2 2012-01-30 12:48 +0000 [r353261] Kevin P. Fleming * /, channels/chan_sip.c: Clarify log WARNING message when port-zero SDP 'm' lines received. Previously, if an m-line in an SDP offer or answer had a port number of zero, that line was skipped, and resulted in an 'Unsupported SDP media type...' warning message. This was misleading, as the media type was not unsupported, but was ignored because the m-line indicated that the media stream had been rejected (in an answer) or was not going to be used (in an offer). ........ Merged revisions 353260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-29 02:44 +0000 [r353176] Russell Bryant * main/netsock.c, /: Find even more network interfaces. The previous change made the code look for emN and pciN in addition to what it did originally, which was search for ethN. However, it needed to be looking for pciN#N, so that's what it does now. This also moves the memset() to be before every ioctl(). ........ Merged revisions 353175 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-28 14:51 +0000 [r353127] Kevin P. Fleming * main/rtp_engine.c, /: Add 'L16-256' MIME subtype alias for slin16. Asterisk has supported the 'L16' MIME subtype for 16kHz signed linear (PCM) audio for quite some time, but some endpoints refer to it as 'L16-256'. This commit adds this as an alias for the existing format. ........ Merged revisions 353126 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-28 04:27 +0000 [r353078] Russell Bryant * main/netsock.c, /: Update ast_set_default_eid() to find more network interfaces. As of Fedora 15, ethN is not the name of ethernet interfaces. The names are emN or pciN. Update some code that searched for interfaces named ethN to look for the new names, as well. For more information about why this change was made, see this page: http://domsch.com/blog/?p=455 ........ Merged revisions 353077 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-27 21:37 +0000 [r352992-353039] Richard Mudgett * apps/app_queue.c: Audit of ao2_iterator_init() usage for v10. Missed one. * tests/test_format_api.c: Audit of ao2_iterator_init() usage for v10. Fix double format_cap iterator cleanup. 2012-01-27 19:19 +0000 [r352965] Jonathan Rose * /, res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor with no valid channel not close AMI session. I also went ahead and took a little time to make sure that the manager value AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's how we handle this stuff these days. (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches: res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766) ........ Merged revisions 352959 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-27 18:36 +0000 [r352956] Richard Mudgett * res/res_srtp.c, main/pbx.c, /, channels/chan_sip.c, include/asterisk/indications.h, res/snmp/agent.c, main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c, apps/app_chanspy.c, main/indications.c, res/res_odbc.c: Audit of ao2_iterator_init() usage for v1.8. Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ ........ Merged revisions 352955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-27 00:08 +0000 [r352863] Alec L Davis * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 352862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan 2012) | 12 lines rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer. If a BLF subscription exists for long enough, using %d may print negative version numbers. Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/ ........ 2012-01-26 20:22 +0000 [r352817] Alexandr Anikin * addons/chan_ooh323.c, /: Fix outbound DTMF for inband mode (tell asterisk core to generate DTMF sounds). (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346) ........ Merged revisions 352807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-26 19:07 +0000 [r352756] Jonathan Rose * /, channels/chan_sip.c: Copy amaflags to sip_pvt from peer during create_addr_from_peer For whatever reason, we don't have a single function for copying data like this from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the sip_pvt, but it would probably be worth discussing this function along with the others that essentially just copy some amount of data from a peer to a private. (Closes issue ASTERISK-19029) Reported by: Matt Lehner ........ Merged revisions 352755 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-26 06:33 +0000 [r352705] Alec L Davis * /, channels/chan_sip.c: Merged revisions 352704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan 2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make similar to other Notify messages. sample output: terminated Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1693/ ........ 2012-01-25 22:23 +0000 [r352651] Paul Belanger * apps/app_voicemail.c, /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 352643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-25 21:18 +0000 [r352616] Kevin P. Fleming * /, main/test.c: Avoid unnecessary rebuilds of main/test.c. main/test.c includes "asterisk/version.h", when it should include "asterisk/ast_version.h" instead (and it should use the ast_get_version() and ast_get_version_num() functions). This commit modifies it to extract the Asterisk version information using the proper APIs, and as a result means that main/test.c no longer needs to be rebuilt when a Subversion checkout is updated or modified. ........ Merged revisions 352612 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-25 17:30 +0000 [r352556] Terry Wilson * /, channels/chan_sip.c: Remove some extraneous debugging from registry memleak fix ........ Merged revisions 352551 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-25 17:16 +0000 [r352520] Richard Mudgett * channels/chan_sip.c, CHANGES, main/message.c, channels/sip/include/sip.h: Fixes for sending SIP MESSAGE outside of calls. * Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA function in the authorization attempt. * Pass up better From header contents for SIP to use. Now is in the "display-name" format expected by MessageSend. (Note that this is a behavior change that could concievably affect some people.) * Block user from adding standard headers that are added automatically. (To, From,...) * Allow the user to override the Content-Type header contents sent by MessageSend. * Decrement Max-Forwards header if the user transferred it from an incoming message. * Expand SIP short header names so the dialplan and other code only has to deal with the full names. * Documents what SIP expects in the MessageSend(from) parameter. (closes issue ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/1683/ 2012-01-25 16:54 +0000 [r352516] Kevin P. Fleming * main/format.c, main/format_cap.c, main/format_pref.c: Eliminate unnecessary rebuilds of main/format*.c. These files have no need to include "asterisk/version.h", and doing so forces them to be rebuilt each time a Subversion checkout moves between 'modified' and 'unmodified' states. 2012-01-25 16:49 +0000 [r352515] Terry Wilson * /, channels/chan_sip.c: Clean up some SIP registry-related memory leaks 1) Be sure and free at unload the epa_backend we allocate at startup 2) Do the same sip_registry cleanup at unload we do at reload Review: https://reviewboard.asterisk.org/r/1689/ ........ Merged revisions 352514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-25 16:41 +0000 [r352512] Jonathan Rose * /, configs/sip.conf.sample: Redocuments sip types peer, user, friend in sip.conf.sample There was faulty information in the sample config describing user as a synonym for friend so it has been changed to better elaborate on the differences between the three entity types. (closes issue ASTERISK-15537) Reported by: yarique ........ Merged revisions 352511 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-24 22:22 +0000 [r352430] Mark Michelson * /, channels/chan_sip.c: Don't do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured. (closes issue ASTERISK-16550) reported by: Olle Johansson ........ Merged revisions 352424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-24 20:35 +0000 [r352373] Jonathan Rose * /, sounds/Makefile: Set core sounds version to 1.4.22. Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22! (closes issue ASTERISK-18978) Reported by: Cameron Twomey Patches: confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002 uploaded by Cameron Twomey ........ Merged revisions 352367 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-24 17:02 +0000 [r352292] Richard Mudgett * /, funcs/func_odbc.c: Fix locking issues with channel datastores in func_odbc.c. * Fixed a potential memory leak when an existing datastore is manually destroyed by inline code instead of calling ast_datastore_free(). (closes issue ASTERISK-17948) Reported by: Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/ ........ Merged revisions 352291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-23 20:30 +0000 [r352228-352231] Mark Michelson * /, main/features.c: Fix grammar of comment. ........ Merged revisions 352230 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/features.c: Fix blind transfers from failing if an 'h' extension is present. This prevents the 'h' extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) Review: https://reviewboard.asterisk.org/r/1685 ........ Merged revisions 352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-23 19:18 +0000 [r352149] Matthew Jordan * /, res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17, V27, V29) before starting spandsp layer While the FAXOPT function could be used to set the modem capabilities, the input to that function was not being applied correctly to the spandsp layer. This patch applies the current model capabilities before starting the spandsp layer. (closes issue: ASTERISK-16409) Reported by: Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license 5081) spandsp-modems-10.diff uploaded by mnicholson (license 5081) ........ Merged revisions 352144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-23 17:34 +0000 [r352091] Richard Mudgett * /, channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the defined enum values. The invalid value used when notifycid was enabled was benign. As far as the code was concerned -1 and 1 are equivalent. (closes issue ASTERISK-19232) Reported by: Eike Kuiper ........ Merged revisions 352090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-21 00:21 +0000 [r352035] Richard Mudgett * /, funcs/func_timeout.c, main/app.c: Fix ast_app_dtget() time unit inconsistency. Note: Noone calls ast_app_dtget() with the timeout parameter of zero so the bad code normally will never get executed. * Fix unnecessary floating point division in func_timeout.c timeout_write() when all other values are integers. (closes issue ASTERISK-16817) Reported by: Dmitry Andrianov ........ Merged revisions 352029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-21 00:08 +0000 [r352015-352017] Mark Michelson * /, channels/chan_sip.c: Remove XXX comment that is not necessary. ........ Merged revisions 352016 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Fix RTP reference leak. If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference for the RTP instance of the transferer. This fixes the issue by merging two similar but slightly conflicting sections of code into a single area. It also adds a stop_media_flows() call in the case that the transferer's UA never sends a BYE to us like it is supposed to. (issue ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/ ........ Merged revisions 352014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-20 19:35 +0000 [r351816-351861] Kinsey Moore * /, codecs/ilbc/iLBC_test.c: More corrections for the ilbc code These changes are in a file that is not compiled by default, and so were missed on earlier checks. ........ Merged revisions 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /: Recorded merge of revisions 351858 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Allow ilbc code to build under dev mode GCC 4.6.3 found some set/unused variables in the ILBC code. * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Restore LSF_check function calls from set/unused variable removal These functions are not noops and modify the array that is passed in. Thanks for the catch Richard. * codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Remove more set, but unused variables in the ilbc codec GCC 4.6.3 caught these in dev mode as well. 2012-01-20 15:59 +0000 [r351762] Jonathan Rose * /, channels/chan_sip.c: Adds setting of mwi_from field to check_auth_result check_peer_ok (closes ASTERISK-19057) Reported By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license 5242) ........ Merged revisions 351759 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-20 15:54 +0000 [r351761] Matthew Jordan * codecs/ilbc/helpfun.c, /: Remove unused variable 'tmp' from helpfun in ilbc codec gcc version 4.6.2 caught an unused variable in the ilbc codec library. This would prevent compilation with --enable-dev-mode; variable removed. ........ Merged revisions 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-20 13:01 +0000 [r351708] Stefan Schmidt * /, contrib/asterisk-ng-doxygen: enable doxygen build for files in the channels/sip folder like reqresp_parser.c ........ Merged revisions 351707 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-19 23:25 +0000 [r351646] Richard Mudgett * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in get_calleridname() parsing and ensure that the output buffer is nul terminated. * Make get_calleridname() truncate the name it parses if the given buffer is too small rather than abandoning the parse and not returning anything for the name. Adjusted get_calleridname_test() unit test to handle the truncation change. * Fix get_in_brackets_test() unit test to check the results of get_in_brackets() correctly. * Fix parse_name_andor_addr() to not return the address of a local buffer. This function is currently not used. * Fix potential NULL pointer dereference in sip_sendtext(). * No need to memset(calleridname) in check_user_full() or tmp_name in get_name_and_number() because get_calleridname() ensures that it is nul terminated. * Reply with an accurate response if get_msg_text() fails in receive_message(). This is academic in v1.8 because get_msg_text() can never fail. ........ Merged revisions 351618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-19 22:43 +0000 [r351612] Kinsey Moore * res/res_rtp_asterisk.c, /: Correct output of RTCP jitter statistics in SR and RR reports Change the RTCP RR and SR generation code to convert Asterisk's internal jitter statistics to be represented in RTP timestamp units based on the rate of the codec in use instead of in seconds. (closes issue ASTERISK-14530) ........ Merged revisions 351611 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-19 21:47 +0000 [r351560] Jonathan Rose * /, channels/chan_sip.c, include/asterisk/netsock2.h: Eliminates doubling the :port part of SIP Notify Message-Account headers. This patch prevents the domain string from getting mangled during the initreqprep step by moving the initialization to before its immediate use. It also documents this pitfall for the ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported by: Yuri Review: https://reviewboard.asterisk.org/r/1678/ ........ Merged revisions 351559 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-19 21:12 +0000 [r351505] Joshua Colp * /, channels/chan_sip.c: Prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) Reported by: Catalin Sanda ........ Merged revisions 351504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-18 21:05 +0000 [r351451] Matthew Jordan * codecs/ilbc/helpfun.c (added), codecs/ilbc/LICENSE_ADDENDUM (added), codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c (added), codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c (added), codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h (added), codecs/ilbc/constants.c (added), codecs/ilbc/iLBC_decode.c (added), codecs/ilbc/createCB.h (added), codecs/ilbc/constants.h (added), codecs/ilbc/iLBC_decode.h (added), codecs/ilbc/iCBSearch.c (added), codecs/ilbc/filter.c (added), codecs/ilbc/hpInput.c (added), codecs/ilbc/gainquant.c (added), codecs/ilbc/iCBSearch.h (added), codecs/ilbc/hpOutput.c (added), codecs/ilbc/rfc3951.txt (added), codecs/ilbc/filter.h (added), codecs/ilbc/hpInput.h (added), codecs/ilbc/LPCencode.c (added), codecs/ilbc/gainquant.h (added), codecs/codec_ilbc.c, codecs/ilbc/hpOutput.h (added), codecs/ilbc/StateSearchW.c (added), codecs/ilbc/PATENTS (added), contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LPCencode.h (added), codecs/ilbc/LICENSE (added), codecs/ilbc/StateSearchW.h (added), codecs/ilbc/iCBConstruct.c (added), codecs/ilbc/syntFilter.c (added), /, codecs/ilbc/iCBConstruct.h (added), codecs/ilbc/iLBC_test.c (added), codecs/ilbc/syntFilter.h (added), codecs/ilbc/StateConstructW.c (added), codecs/ilbc/packing.c (added), codecs/ilbc/StateConstructW.h (added), codecs/ilbc/packing.h (added), codecs/ilbc/getCBvec.c (added), codecs/ilbc/LPCdecode.c (added), codecs/ilbc/enhancer.c (added), codecs/ilbc/lsf.c (added), codecs/ilbc/iLBC_encode.c (added), codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added), codecs/ilbc/enhancer.h (added), codecs/ilbc/FrameClassify.c (added), codecs/ilbc/iLBC_define.h (added), codecs/ilbc/lsf.h (added), codecs/ilbc/extract-cfile.awk (added), codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile, codecs/ilbc/FrameClassify.h (added): Include iLBC source code for distribution with Asterisk This patch includes the iLBC source code for distribution with Asterisk. Clarification regarding the iLBC source code was provided by Google, and the appropriate licenses have been included in the codecs/ilbc folder. Review: https://reviewboard.asterisk.org/r/1675 Review: https://reviewboard.asterisk.org/r/1649 (closes issue: ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan ........ Merged revisions 351450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-18 15:57 +0000 [r351408] Stefan Schmidt * /, channels/chan_sip.c: The get_pai function in chan_sip.c didn't recognized a proper callerid name and number from a P-Asserted-Identity cause the header parsing logic was wrong. Changing the parsing functions to the sip header parsing APIs in reqresp_parser.h solves this problem. Review: https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and Mark Michelson ........ Merged revisions 351396 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 17:22 +0000 [r351308] Mark Michelson * res/res_rtp_asterisk.c, /: Eliminate odd initialization of probation variable. ........ Merged revisions 351306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 17:08 +0000 [r351289] Jonathan Rose * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: Adds pjmedia probation concepts to res_rtp_asterisk's learning mode. In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ ........ Merged revisions 351287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 16:54 +0000 [r351286] Mark Michelson * /, channels/chan_sip.c: Use built-in parsing functions for Contact and Record-Route headers. If a Contact or a Record-Route header had a quoted string with an item in angle brackets, then we would mis-parse it. For instance, "Bob <1234>" <1234@example.org> would be misparsed as having the URI "1234" The fix for this is to use parsing functions from reqresp_parser.h since they are heavily tested and are awesome. (issue ASTERISK-18990) ........ Merged revisions 351284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 16:07 +0000 [r351234] Matthew Jordan * /, channels/chan_sip.c: Fix udptl issue with initial INVITE introduced by r351027 When an inital INVITE occurs that contains image media, a channel is not yet associated with the SIP dialog. The file descriptor associated with the udptl session needs to be set in initialize_udptl or in sip_new to account for this scenario. ........ Merged revisions 351233 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-17 01:43 +0000 [r351183] Russell Bryant * /, channels/chan_sip.c: Merged revisions 351182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) | 22 lines Add some missing locking in chan_sip. This patch adds some missing locking to the function send_provisional_keepalive_full(). This function is called from the scheduler, which is processed in the SIP monitor thread. The associated channel (or pbx) thread will also be using the same sip_pvt and ast_channel so locking must be used. The sip_pvt_lock_full() function is used to ensure proper locking order in a safe manner. In passing, document a suspected reference counting error in this function. The "fix" is left commented out because when the "fix" is present, crashes occur. My theory is that fixing it is exposing a reference counting error elsewhere, but I don't know where. (Or my analysis of this being a problem could have been completely wrong in the first place). Leave the comment in the code for so that someone may investigate it again in the future. Also add a bit of doxygen to transmit_provisional_response(). (closes issue ASTERISK-18979) Review: https://reviewboard.asterisk.org/r/1648 ........ 2012-01-16 21:17 +0000 [r351081-351131] Terry Wilson * /, channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200 response to INVITE When handling a non-2xx final response on an INVITE transaction, we have to keep the transaction around after we send an ACK in case we receive a retransmission of the response so we can re-transmit the ACK, but also tear down the ast_channel as soon as we transmit the ACK. Before this patch, we could fail at both of these things. Calling sip_alreadygone/needdestroy prevented us from keeping the transaction up and retransmitting the ACK, and queueing CONGESTION was not sufficient to cause the channel to be torn down when originating calls via the CLI, for example. This patch queues a hangup with CONGESTION instead of just queueing CONGESTION for these responses and removes the sip_alreadygone and sip_needdestroy calls from handle_response_invite on non-2xx responses. It relies on the hangup calling sip_scheddestroy. For more information, see section 17.1.1.1 of RFC 3261. (closes issue ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/ ........ Merged revisions 351130 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Don't prematurely stop SIP session timer When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry. (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: session_timer_fix.diff by Terry Wilson (License #5357) based on session_timer.patch by Thomas Arimont (License #5525) ........ Merged revisions 351080 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-16 19:12 +0000 [r351028] Matthew Jordan * /, channels/chan_sip.c: Create and initialize udptl only when dialog negotiates for image media Prior to this patch, the udptl struct was allocated and initialized when a dialog was associated with a peer that supported T.38, when a new SIP channel was allocated, or what an INVITE request was received. This resulted in any dialog associated with a peer that supported T.38 having udptl support assigned to it, including the UDP ports needed for communication. This occurred even in non-INVITE dialogs that would never send image media. This patch creates and initializes the udptl structure only when the SDP for a dialog specifies that image media is supported, or when Asterisk indicates through the appropriate control frame that a dialog is to support T.38. (closes issue ASTERISK-16698) Reported by: under Tested by: Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar Broad Tested by: Stefan Schmidt review: https://reviewboard.asterisk.org/r/1668/ ........ Merged revisions 351027 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-16 17:11 +0000 [r350978] Sean Bright * main/db.c: Sort the output of 'database showkey' as well. You can pass wildcards (%) to the database CLI commands, so this will sort the returned list of matches. 2012-01-16 17:06 +0000 [r350976] Joshua Colp * main/rtp_engine.c, /: Add missing code to set direct RTP setup information during dialing. ........ Merged revisions 350975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-16 14:27 +0000 [r350938] Sean Bright * main/db.c: Sort the output of 'database show' by key. This more closely mimics the behavior of 'database show' before the conversion to sqlite3. 2012-01-15 20:12 +0000 [r350886-350889] Walter Doekes * /, main/asterisk.c: Allow only one thread at a time to do asterisk cleanup/shutdown. Add locking around the really-really-quit part of the core stop/restart part. Previously more than one thread could be called to do cleanup, causing atexit handlers to be run multiple times, in turn causing segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1662/ Review: https://reviewboard.asterisk.org/r/1658/ ........ Merged revisions 350888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, utils/extconf.c: Fix -Werror=unused-but-set-variable compile error in utils/extconf.c. Note that I'm not confirming legitimacy of having that file in tree at all. Is anyone using aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged revisions 350885 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-14 16:41 +0000 [r350790-350838] Kevin P. Fleming * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac, autoconf/libcurl.m4: Ensure that all AC_LANG_PROGRAM calls in the configure script are properly quoted. Recent versions of autoconf (2.68 on my system) won't properly process the configure script unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in the script were, but many were not. This patch corrects the unquoted calls. ........ Merged revisions 350837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * contrib/scripts/install_prereq, /, channels/chan_h323.c, addons/chan_mobile.c, res/res_pktccops.c: Multiple revisions 350788-350789 ........ r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two prerequisites are properly installed on Debian-style distributions. * Don't specify a specific version of libgmime; newer versions are available now and acceptable. * Install libsrtp so that res_srtp can be built. ........ r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines Correct some 'set-but-not-used' variable warnings. ........ Merged revisions 350788-350789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 22:10 +0000 [r350737] Kinsey Moore * /, include/asterisk/autoconfig.h.in: Run bootstrap.sh for the for the ASTERISK-18929 fix configure and autoconfig.h.in were not regenerated when the fix was committed. ........ Merged revisions 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 21:51 +0000 [r350734] Richard Mudgett * /, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample: Correct eventtype names in cel_odbc and cel_pgsql sample files ........ Merged revisions 350733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 21:41 +0000 [r350731] Kinsey Moore * /, configure.ac, bootstrap.sh, main/asterisk.c: Make sure asterisk builds on OpenBSD OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not 'struct ucred', which causes compilation of main/asterisk.c to fail in read_credentials(). This allows configure to check for sockpeercred and asterisk to deal with it properly. (closes issue ASTERISK-18929) Reported-by: Barry Miller Patch-by: Barry Miller ........ Merged revisions 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 20:31 +0000 [r350680] Mark Michelson * /, channels/sip/config_parser.c: Set port to a default sane value if a bogus one is provided when parsing hostnames. ........ Merged revisions 350679 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 17:29 +0000 [r350585] Richard Mudgett * configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c, configs/cel.conf.sample, /, cel/cel_manager.c, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample, main/cel.c, configs/cel_custom.conf.sample: Add missing CEL logging fields to various CEL backends. Multiple revisions 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51 -0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging fields to various CEL backends. * Add missing eventextra to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and EventExtra to cel_manager.c. * Add missing userdeftype support for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample. (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13 Jan 2012) | 8 lines Use compatible names for event extra data for various CEL backends. * Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190) ........ Merged revisions 350555,350571 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-13 16:59 +0000 [r350550-350553] Matthew Jordan * /, apps/app_queue.c: Realtime queues failed to load queue information without queue member table Previously, realtime queues could be loaded without defining the queue member table. This allowed for queue members to be dynamic, while the realtime queue definitions could exist in some backing storage. Revision 342223 broke this when it changed the return value for realtime_multientry to return NULL when no results are returned. Previously, an empty ast_config object was expected. (closes issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 350552 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * bridges/bridge_builtin_features.c, channels/chan_bridge.c, include/asterisk/bridging.h, apps/app_confbridge.c, main/bridging.c: Fix crash from bridge channel hangup race condition in ConfBridge This patch addresses two issues in ConfBridge and the channel bridge layer: 1. It fixes a race condition wherein the bridge channel could be hung up 2. It removes the deadlock avoidance from the bridging layer and makes the bridge_pvt an ao2 ref counted object Patch by David Vossel (mjordan was merely the commit monkey) (issue ASTERISK-18988) (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628) (closes issue ASTERISK-19100) Reported by: Matt Jordan Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1654/ 2012-01-12 16:04 +0000 [r350502] Jonathan Rose * /, main/features.c: Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic Colledge Patches: features_18.patch uploaded by Nic Colledge (license 6245) ........ Merged revisions 350501 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-11 22:51 +0000 [r350312-350453] Richard Mudgett * /, main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a CEL dummy channel. (closes issue ASTERISK-19180) Reported by: Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_dial.c, /, CHANGES, apps/app_followme.c: Make FollowMe optionally update connected line information when the accepting endpoint is bridged. Like Dial and Queue, FollowMe needs to deal with AST_CONTROL_CONNECTED_LINE information so when the parties are initially bridged, the connected line information will be correct. * Added the 'I' option just like the app_dial and app_queue 'I' option. * Made 'N' option ignored if the call is already answered. (closes issue ASTERISK-18969) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1656/ ........ Merged revisions 350364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK function. The time passed by the LOCK function to an internal function was relative time when the function expected absolute time. * Don't use C++ keywords in get_lock(). (closes issue ASTERISK-16868) Reported by: Andrey Solovyev Patches: 20101102__issue18207.diff.txt (license #5003) patch uploaded by Andrey Solovyev (modified) ........ Merged revisions 350311 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-09 21:55 +0000 [r350221] Richard Mudgett * /, channels/chan_iax2.c: Fix joinable thread terminating without joiner memory leak in chan_iax.c. The iax2_process_thread() can exit without anyone waiting to join the thread. If noone is waiting to join the thread then a large memory leak occurs. * Made iax2_process_thread() deatach itself if nobody is waiting to join the thread. (closes issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch (license #5617) patch uploaded by Alex Villacis Lasso (modified) (closes issue ASTERISK-17825) Reported by: wangjin ........ Merged revisions 350220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-09 19:34 +0000 [r350180] Walter Doekes * main/db.c: Fix shutdown handling of sqlite3 astdb. If a db_sync was scheduled just before shutdown, the atexit code calling db_sync would have no effect, causing the astdb commit thread to stay alive. This caused the SIP/realtime_sipregs test to fail. (The fallback kill would run the atexit code again and that would wreak havoc.) This fixes that the atexit kill condition is picked up properly. (closes issue ASTERISK-18883) Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1659 2012-01-09 18:57 +0000 [r350076-350129] Richard Mudgett * /, contrib/scripts/valgrind_compare (added): Multiple revisions 350127-350128 ........ r350127 | rmudgett | 2012-01-09 12:40:33 -0600 (Mon, 09 Jan 2012) | 12 lines Update contrib script live_ast to invoke Asterisk with valgrind and suppression file. * Added valgrind_compare script to compare two valgrind log files for differences. (issue ASTERISK-17339) Reported by: Tzafrir Cohen Patches: valgrind_compare (license #5035) script uploaded by Tzafrir Cohen live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger ........ r350128 | rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 lines live_ast: valgrind: run asterisk under valgrind Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under valgrind. The extra command-line parameters are passed to Asterisk as usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review: https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 350127-350128 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/asterisk.c: Make Asterisk -x command line parameter imply -r parameter presence. The Asterisk -x command line parameter is documented inconsistently. * Made the -x documentation and behavior consistent. * Since this is also a new year, updated the copyright notices while here. (closes issue ASTERISK-19094) Reported by: Eugene Patches: issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified) Tested by: Eugene ........ Merged revisions 350075 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-09 15:39 +0000 [r350024] Kinsey Moore * /, apps/app_meetme.c: Prevent SLA settings from getting wiped out on reload If SLA was reloaded without the config file being changed, current settings got wiped out before the SLA reload code decided it wasn't going to reload the file since nothing was changed. Moving the settings reset later in the reload process fixes this. (closes issue AST-744) ........ Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-06 23:25 +0000 [r349977] Terry Wilson * /, channels/chan_sip.c: Don't leak CID in From header when presentation=unavailable When someone does Set(CALLERPRES()=unavailable) (or Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows "Anonymous" . When sendrpid=yes/pai, the From header will still display the callerid info, even though we supply an rpid header with the anonymous info. It seems like we shouldn't leak that info in any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems to indicate that one shouldn't send identifying info in the From in this case. This patch anonymizes the From header as well even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review: https://reviewboard.asterisk.org/r/1649/ ........ Merged revisions 349968 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-06 21:25 +0000 [r349928] Kinsey Moore * pbx/pbx_lua.c: Fix lua goto detection to prevent unexpected behavior with confbridge A bug in the pbx_lua goto detection was causing the dialplan to hangup unexpectedly after confbridge exited if it had called lua dialplan code during execution. Patch-by: Timo Teras Acked-by: Matt Nicholson (closes issue ASTERISK-18976) 2012-01-06 16:48 +0000 [r349873] Richard Mudgett * /, apps/app_followme.c: Fix memory leaks in app_followme find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt Jordan ........ Merged revisions 349872 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-05 23:56 +0000 [r349822] Matthew Jordan * res/res_fax.c: Fix premature free'ing of the frame committed in r349608 Even though we set the frame to the ast_null_frame and return that, the caller of the frame hook may still need the frame. This now is a bit more careful about when it frees the frame, i.e., only under the same conditions that applied when we duplicated it in the first place. 2012-01-05 23:46 +0000 [r349820] Richard Mudgett * /, cel/cel_sqlite3_custom.c: Make not assume that the cel_sqlite3_custom SQL table primary key is AcctId. If a table is created by some other application and the primary key is not named "AcctId", cel/cel_sqlite3_custom.c will always try to create the table and fail because it already exists. * Change the SQL table query to not require AcctId as the primary key. (closes issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch (license #6337) patch uploaded by socketpair ........ Merged revisions 349819 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-05 22:10 +0000 [r349732] Kinsey Moore * /, main/file.c: Allow playback of formats that don't support seeking ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. This regression was introduced in r158062. (closes issue ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-05 21:55 +0000 [r349673-349729] Jonathan Rose * main/dsp.c, /: Fix an issue where dsp.c would interpret multiple dtmf events from a single key press. When receiving calls from a mobile phone into a DISA system on a connection with significant interference, the reporter's Asterisk system would interpret DTMF incorrectly and replicate digits received. This patch resolves that by increasing the number of frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and adjusts dtmf_detect function to reset hits and misses only when an edge is detected. (closes issue ASTERISK-17493) Reported by: Alec Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546) Review: https://reviewboard.asterisk.org/r/1130/ ........ Merged revisions 349728 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, main/asterisk.c: Ensures Asterisk closes when receiving terminal signals in 'no fork' mode. When catching a signal, in no fork mode the console thread is identical to the thread responsible for catching the signal and closing Asterisk, which requires it to first dispense with the console thread. Prior to this patch, if these threads were identical, upon receiving a killing signal, the thread will send an URG signal to itself, which we also catch and then promptly do nothing with. Obviously this isn't useful behavior. (closes issue ASTERISK-19127) Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 349672 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-04 22:19 +0000 [r349608-349619] Matthew Jordan * apps/confbridge/conf_config_parser.c: Fix for ConfBridge config parser unlocking channel mutex too many times When looking up a ConfBridge profile, the config parser would, if it found a channel datastore on the channel requesting the bridge profile, unlock the channel mutex twice. Since that's a little aggressive, it now only unlocks it once. (closes issue ASTERISK-19042) Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042 uploaded by David Vossel (license 5628) * res/res_fax.c: Free successfully translated frame in fax_gateway_framehook A frame that is translated via ast_translate is also duplicated via ast_frdup. This will allocate a new frame on the heap, which needs to be free'd at the appropriate time. This issue reporter used valgrind to find that this occurred in res_fax's fax_gateway_framehook; a quick search through the code showed that only place this was currently not handling the translatted frame properly. (closes issue ASTERISK-19133) Reported by: Sylvain Rochet 2012-01-04 20:50 +0000 [r349559] Richard Mudgett * channels/chan_dahdi.c, /: Fix segfault in chan_dahdi for CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private pointer checks in the following chan_dahdi channel callbacks: dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by: Diego Aguirre Tested by: rmudgett ........ Merged revisions 349558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-04 20:23 +0000 [r349505-349532] Kinsey Moore * contrib/init.d/rc.debian.asterisk, /: Make debian init script conform to the LSB standard Previously, this init script would return 1 if Asterisk was already running. This is incorrect behavior according to the LSB standard and has been fixed by returning 0 instead. (closes issue ASTERISK-17958) Reported-by: johnc ........ Merged revisions 349529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8: Update autosupport script and man page Added information collection from the output of the utilities: top, free, uptime, ifconfig Added information collection from the output of the Asterisk command 'dahdi show status' Added option / flag '-n, --non-interactive' Updated man page to reflect new option / flag '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes issue AST-749) ........ Merged revisions 349504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-04 19:44 +0000 [r349451-349502] Jonathan Rose * /, channels/chan_sip.c: Adds Subscription-State header to notify with call completion. per RFC3265 (Closes issue ASTERISK-17953) Reported by: George Konopacki Patches: 19400.patch uploaded by mmichelson (license 5049) ........ Merged revisions 349482 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/pbx.c, /: Fix documentation for SayNumber to reflect the fact that language is changed in CHANNEL() (closes issue ASTERISK-18962) reported by: Nir Simionovich ........ Merged revisions 349450 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2012-01-27 Asterisk Development Team * Asterisk 10.1.0 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-4 2012-01-24 Asterisk Development Team * Asterisk 10.1.0-rc2 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-3 * main/file.c: Allow playback of formats that don't support seeking. ast_streamfile previously did unconditional seeking on files that broke playback of formats that don't support that functionality. This patch avoids the seek that was causing the problem. (closes issue ASTERISK-18994) Patch-by: Timo Teras * channels/chan_sip.c: AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) Reported by: Catalin Sanda * channels/chan_sip.c: Fix RTP reference leak. If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP instance of the transferer. (closes issue ASERISK-19192) Reported by: Tyuta Vitali * res/res_rtp_asterisk: Add pjmedia probation concepts to res_rtp_asterisk's learning mode. In order to better handle RTP sources with strictrtp enabled (which is the default setting in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. * main/features.c: Fix blind transfers from failing if an 'h' extension is present. This prevents the 'h' extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) * apps/app_queue.c: Realtime queues failed to load queue information without queue member table. Revision 342223 broke this when it changed the return value for realtime_multientry to return NULL when no results are returned. (closes issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene Mendoza 2011-12-30 Asterisk Development Team * Asterisk 10.1.0-rc1 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-1 2011-12-29 15:14 +0000 [r349340] Matthew Jordan * main/rtp_engine.c, /: Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary. (issue ASTERISK-19040) (issue ASTERISK-19128) (issue ASTERISK-17725) (issue ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1640/ ........ Merged revisions 349339 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-28 21:33 +0000 [r349290] Sean Bright * /, main/audiohook.c: Use ast_audiohook_write_list_empty to determine if our lists are empty instead of duplicating that logic. ........ Merged revisions 349289 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-28 19:00 +0000 [r349248-349250] Kevin P. Fleming * utils: Tell Subversion to gnore the 'astdb2bdb' binary file if it exists. * main/dsp.c, res/res_fax.c, include/asterisk/dsp.h, include/asterisk/res_fax.h, res/res_fax_spandsp.c: Improve T.38 gateway V.21 preamble detection. This commit removes the V.21 preamble detection code previously added to the generic DSP implementation in Asterisk, and instead enhances the res_fax module to be able to utilize V.21 preamble detection functionality made available by FAX technology modules. This commit also adds such support to res_fax_spandsp, which uses the Spandsp modem tone detection code to do the V.21 preamble detection. There should be no functional change here, other than much more reliable V.21 preamble detection (and thus T.38 gateway initiation). 2011-12-27 20:53 +0000 [r349195] Matthew Jordan * /, res/res_timing_pthread.c, include/asterisk/module.h, res/res_timing_dahdi.c, res/res_timing_timerfd.c, res/res_musiconhold.c: Fix timing source dependency issues with MOH Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patches: asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026) Review: https://reviewboard.asterisk.org/r/1578/ ........ Merged revisions 349194 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-27 17:17 +0000 [r349145] Sean Bright * /, main/audiohook.c: Once an audiohook is attached to a channel, we continue to transcode all of the frames, even after all of the hooks are detached. This patch short-cicuits us out before we transcode unnecessarily. ........ Merged revisions 349144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-23 17:32 +0000 [r349045] Sean Bright * /, apps/app_chanspy.c: Merged revisions 349044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec 2011) | 18 lines In ChanSpy, don't create audiohooks that will never be used. When ChanSpy is initialized it creates and attaches 3 audiohooks: 1) Read audio off of the channel that we are spying on 2) Write audio to the channel that we are spying on 3) Write audio to the channel that is bridged to the channel that we are spying on. The first is always necessary, but the others are used only when specific options are passed to the ChanSpy application (B, d, w, and W to be specific). When those flags are not passed, neither of those audiohooks are ever sent frames, but we still try to process the hooks for each voice frame that we recieve on the channel. So in short - only create and attach audiohooks that we actually need. ........ 2011-12-23 15:25 +0000 [r348993] Kinsey Moore * apps/app_dial.c, /: Fix missing doc tags found while fixing ASTERISK-18689 Add missing tags in app_dial documentation. ........ Merged revisions 348992 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-23 02:30 +0000 [r348952] Richard Mudgett * main/pbx.c, /, channels/chan_sip.c, include/asterisk/pbx.h: Fix extension state callback references in chan_sip. Chan_sip gives a dialog reference to the extension state callback and assumes that when ast_extension_state_del() returns, the callback cannot happen anymore. Chan_sip then reduces the dialog reference count associated with the callback. Recent changes (ASTERISK-17760) have resulted in the potential for the callback to happen after ast_extension_state_del() has returned. For chan_sip, this could be very bad because the dialog pointer could have already been destroyed. * Added ast_extension_state_add_destroy() so chan_sip can account for the sip_pvt reference given to the extension state callback when the extension state callback is deleted. * Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy() and handle_statechange() now that the struct ast_state_cb has a destructor to call. * Ensure that ast_extension_state_add_destroy() will never return -1 or 0 for a successful registration. * Fixed pbx.c statecbs_cmp() to compare the correct information. The passed in value to compare is a change_cb function pointer not an object pointer. * Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for deadlocking when those locks are held during the callback. * Removed unused lock declaration for the pbx.c store_hints list. (closes issue ASTERISK-18844) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/1635/ ........ Merged revisions 348940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-22 22:37 +0000 [r348846-348889] Matthew Jordan * cel/cel_pgsql.c, /: Fix for memory leaks / cleanup in cel_pgsql There were a number of issues in cel_pgsql's pgsql_log method: * If either sql or sql2 could not be allocated, the method would return while the pgsql_lock was still locked * If the execution of the log statement succeeded, the sql and sql2 structs were never free'd * Reconnection successes were logged as ERRORs. In general, the severity of several logging statements was reduced (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/ ........ Merged revisions 348888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/say.c, main/file.c, main/app.c, apps/app_confbridge.c, main/bridging.c: Add Asterisk TestSuite event hooks to support ConfBridge testing This patch adds initial testsuite event hooks so that ConfBridge tests can be executed in the Asterisk TestSuite. (issue ASTERISK-19059) 2011-12-22 20:17 +0000 [r348845] Terry Wilson * /, include/asterisk/format_pref.h: Allow packetization vaules > 127 According to the RTP packetization documentation, and the maximum values listed in AST_FORMAT_LIST, we should support values > that the signed char array that ast_codec_pref makes available to store the value. All places in the code treat the framing field as though it were an int array instaead of a char array anyway, so this just fixes the type of the array. (closes issue ASTERISK-18876) Review: https://reviewboard.asterisk.org/r/1639/ ........ Merged revisions 348833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-21 20:13 +0000 [r348736-348793] Richard Mudgett * codecs/speex: Make codecs/speex ignore *.i files also. * apps/confbridge: Make apps/confbridge ignore *.i files also. * /, channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS number if it is blank. Some ISDN switches complain or block the call if the RDNIS number is empty. * Made chan_iax2 not save a RDNIS number into the ast_channel if the string is blank. This is what other channel drivers do. (closes issue ASTERISK-17152) Reported by: rmudgett ........ Merged revisions 348735 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-19 21:37 +0000 [r348648] Richard Mudgett * /, configure, configure.ac: Fix crashes on other platforms caused by interference from Darwin weak symbol support. Support weak symbols on a platform specific basis. The Mac OS X (Darwin) support must be isolated from the other platforms because it has caused other platforms to crash. Several other platforms including Linux have GCC versions that define the weak attribute. However, this attribute is only setup for use in the code by Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged revisions 348647 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-19 19:48 +0000 [r348605] Leif Madsen * main/message.c: Update documentation for MESSAGE_SEND_STATUS variable. (Closes issue ASTERISK-19056) Reported by: Yuri Patches: 348360.diff uploaded by Yuri (license #5242) 2011-12-18 18:28 +0000 [r348517] 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 'force_rport' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. ........ Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 348516 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-16 23:56 +0000 [r348311-348465] Richard Mudgett * main/channel.c, /, main/features.c: Clean-up on isle five for __ast_request_and_dial() and ast_call_forward(). * Add locking when a channel inherits variables and datastores in __ast_request_and_dial() and ast_call_forward(). Note: The involved channels are not active so there was minimal potential for problems. * Remove calls to ast_set_callerid() in __ast_request_and_dial() and ast_call_forward() because the set information is for the wrong direction. * Don't use C++ keywords for variable names in ast_call_forward(). * Run the redirecting interception macro if defined when forwarding a call in ast_call_forward(). Note: Currently will never execute because the only callers that supply a calling channel supply a hungup or zombie channel. * Make feature_request_and_dial() put the transferee into autoservice when it calls ast_call_forward() in case a redirection interception macro is run. Note: Currently will never happen because the caller channel (Party B) is always hungup at this time. * Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame to silence a log message. ........ Merged revisions 348464 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/channel.c, /: Fix cut and past error in ast_call_forward(). (issue ASTERISK-18836) ........ Merged revisions 348401 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/channel.c, main/pbx.c, /, apps/app_authenticate.c, funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h, apps/app_followme.c, apps/app_queue.c, res/res_monitor.c: Fix crash during CDR update. The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ ........ Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_parkandannounce.c, /: Fix ParkAndAnnounce to pass the CallerID to the announcing channel. ParkAndAnnounce tried to pass the CallerID to the announcing channel but the ID was wiped out by the channel masquerade done when parking the call. * Save the CallerID before parking the channel to pass it to the announcing channel. * Fixed a minor memory leak in ParkAndAnnounce. * Updated some ParkAndAnnounce log messages. ........ Merged revisions 348310 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-14 22:34 +0000 [r348265] Matthew Jordan * apps/app_originate.c: Added support for all slin formats to app_originate Previously, app_originate could not originate a call into a non-8kHz conference bridge as the formats for non-8kHz slin codecs were not applied to the created channel. This patch adds all of the formats by default, such that if a created channel has a codec that supports a higher sampling rate, a translation path can be built between it and other channels. 2011-12-14 22:05 +0000 [r348213] Matthew Nicholson * /, res/res_fax.c: Don't clear LOCALSTATIONID before sending or receiving. The user may set that variable. ASTERISK-18921 ........ Merged revisions 348212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-14 21:58 +0000 [r348211] Matthew Jordan * apps/app_queue.c: Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input The function QUEUE_MEMBER has two required parameters (queuename, option). It was only checking for the presence of queuename. The patch checks for the existence of the option parameter and provides better error logging when invalid values are provided for the option parameter as well. 2011-12-14 20:35 +0000 [r348155-348158] Jonathan Rose * /, configs/features.conf.sample: Fix accidental use of tabs instead of spaces from previous features.conf.sample change ........ Merged revisions 348157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configs/features.conf.sample: Document PARKINGSLOT variable in features.conf.sample (issue ASTERISK-16239) ........ Merged revisions 348154 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-13 23:06 +0000 [r348102] Richard Mudgett * /, bridges/bridge_builtin_features.c, apps/app_followme.c: Fix FollowMe CallerID on outgoing calls. The addition of the Connected Line support changed how CallerID is passed to outgoing calls. The FollowMe application was not updated to pass CallerID to the outgoing calls. * Fix FollowMe CallerID on outgoing calls. * Restructured findmeexec() to fix several memory leaks and eliminate some duplicated code. * Made check the return value of create_followme_number(). Putting a NULL into the numbers list is bad if create_followme_number() fails. * Fixed a couple uses of ast_strdupa() inside loops. * The changes to bridge_builtin_features.c fix a similar CallerID issue with the bridging API attended and blind transfers. (Not used at this time.) (closes issue ASTERISK-17557) Reported by: hamlet505a Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1612/ ........ Merged revisions 348101 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-13 15:20 +0000 [r348056] Stefan Schmidt * channels/chan_sip.c: Fix possible misshandling of an incoming SIP response as a peer poke response. Also make sure peer has even qualify enabled when handle a peer poke response. (closes issue ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed by: David Vossel ........ Merged revisions 348048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-12 19:24 +0000 [r347996] Terry Wilson * res/res_srtp.c, /: Add a separate buffer for SRTCP packets The function ast_srtp_protect used a common buffer for both SRTP and SRTCP packets. Since this function can be called from multiple threads for the same SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the packets to become corrupted as the buffer was used by both threads simultaneously. This patch adds a separate buffer for SRTCP packets to avoid the problem. (closes issue ASTERISK-18889, Reported/patch by Daniel Collins) ........ Merged revisions 347995 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-12 18:13 +0000 [r347953-347955] Richard Mudgett * configs/extensions.conf.sample, configs/iax.conf.sample, configs/chan_dahdi.conf.sample, configs/chan_ooh323.conf.sample, configs/vpb.conf.sample, configs/extensions.lua.sample, configs/sip.conf.sample: Reverting -r347953 for ASTERISK-14122 * configs/extensions.conf.sample, configs/iax.conf.sample, configs/chan_dahdi.conf.sample, configs/chan_ooh323.conf.sample, configs/vpb.conf.sample, configs/extensions.lua.sample, configs/sip.conf.sample: Update sample configs to put incoming calls into context public. * Add warning about the SIP allowguest option in context public. (closes issue ASTERISK-14122) Reported by: Alec Davis Review: https://reviewboard.asterisk.org/r/719/ 2011-12-09 01:29 +0000 [r347812] Richard Mudgett * main/pbx.c, /: Fix some parsing issues in add_exten_to_pattern_tree(). * Simplify compare_char() and avoid potential sign extension issue. * Fix infinite loop in add_exten_to_pattern_tree() handling of character set escape handling. * Added buffer overflow checks in add_exten_to_pattern_tree() character set collection. * Made ignore empty character sets. * Added escape character handling to end-of-range character in character sets. This has a slight change in behavior if the end-of-range character is an escape character. You must now escape it. * Fix potential sign extension issue when expanding character set ranges. * Made remove duplicated characters from character sets. The duplicate characters lower extension matching priority and prevent duplicate extension detection. * Fix escape character handling when the escape character is trying to escape the end-of-string. We could have continued processing characters after the end of the exten string. We could have added the previous character to the pattern matching tree incorrectly. (closes issue ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions 347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-08 21:31 +0000 [r347727] Walter Doekes * /, channels/chan_sip.c: Fix regression when using tcpenable=no and tlsenable=yes. The tlsenable settings are tucked away in main/tcptls.c, so I missed them when resolving ASTERISK-18837. This should resolve the test suite breakage of the sip tls tests. Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt Jordan ........ Merged revisions 347718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-08 20:43 +0000 [r347656] Jonathan Rose * apps/app_queue.c: Fix regressed behavior of queue set penalty to work without specifying 'in ' r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when attempting to set penalty on an interface without specifying a queue in the queue set penalty CLI command. In addition, no attempt would be made whatsoever to perform the penalty setting on all the queues in the core list with either the cli command or the non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to document and rename some functions required by this command to better represent what they actually do. Oh yeah, and the use of this command without specifying a specific queue actually works now. Review: https://reviewboard.asterisk.org/r/1609/ 2011-12-08 17:53 +0000 [r347600] Richard Mudgett * /, main/features.c: Mark channel running the h exten with the soft-hangup flag. When a bridge is broken, ast_bridge_call() might execute the h exten on the calling channel. However, that channel may not have been the channel that broke the bridge by hanging up. The channel executing the h exten must be in a hung up state so things like AGI run in the correct mode. * Make sure ast_bridge_call() marks the channel it is executing the h exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to match the pbx.c main dialplan execution loop when it executes the h exten.) (closes issue ASTERISK-18811) Reported by: David Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: David Hajek, rmudgett ........ Merged revisions 347595 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-08 16:20 +0000 [r347532] 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) ........ Merged revisions 347530 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 347531 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-07 21:39 +0000 [r347439] Richard Mudgett * main/manager.c, /: Update AMI Getvar and Setvar documentation about supplying a channel name. (closes issue ASTERISK-18958) Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-07 20:27 +0000 [r347383] Jonathan Rose * /, apps/app_meetme.c: Fix: Meetme recording variables from realtime DB use null entries over channel variables Meetme would attempt to substitute the realtime values of RECORDING_FILE and RECORDING_FORMAT from the meetme db entry instead of using the channel variable set for those variables in spite of those database entries being NULL or even lacking a column to represent them. (closes issue ASTERISK-18873) Reported by: Byron Clark Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157) ........ Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-07 20:12 +0000 [r347344] Terry Wilson * Makefile, include/asterisk/paths.h, configs/asterisk.conf.sample, build_tools/make_defaults_h, main/asterisk.c, main/db.c: Add ASTSBINDIR to the list of configurable paths This patch also makes astdb2sqlite3 and astcanary use the configured directory instead of relying on $PATH. (closes issue ASTERISK-18959) Review: https://reviewboard.asterisk.org/r/1613/ 2011-12-06 23:56 +0000 [r347293] Richard Mudgett * /, channels/chan_sip.c: Make SIP INFO messages for dtmf-relay signals case insensitive. (closes issue ASTERISK-18924) Reported by: Kevin Taylor ........ Merged revisions 347292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-06 21:53 +0000 [r347240] Jonathan Rose * main/pbx.c, /: Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten If waitExten specifies a music class to use with its music on hold option, it will use CHANNEL(musicclass) instead if that channel variable has been set on the initiating channel. This documents that behavior in the waitExten app so that this can be known without checking the documentation of the code in function local_ast_moh_start. (closes issue ASTERISK-18804) ........ Merged revisions 347239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-06 19:42 +0000 [r347124-347167] Walter Doekes * /, channels/chan_sip.c: Don't allow transport=tcp when tcpenable=no. When tcpenable=no, sending to transport=tcp hosts was still allowed. Resolving the source address wasn't possible and yielded the string "(null)" in SIP messages. Fixed that and a couple of not-so-correct log messages. (closes issue ASTERISK-18837) Reported by: Andreas Topp Review: https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan ........ Merged revisions 347166 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: Add regression tests for issue ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572 Reviewed by: Matt Jordan ........ Merged revisions 347131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: Move setting of voicemail zonetag and locale up a bit. The voicemail [general] zonetag and locale variables weren't loaded until after the mailboxes were initialized. This caused the settings to be unset for those mailboxes until a reload was performed. (closes issue ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570 Reviewed by: Matt Jordan ........ Merged revisions 347111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-06 17:24 +0000 [r347068] Matthew Jordan * /, channels/chan_sip.c: Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. When a peer is removed, either by pruning realtime SIP peers or by unloading / loading chan_sip, the MWI subscriptions that were orphaned would still be on the event engine list of valid subscriptions but have a pointer to a peer that no longer was valid. When an MWI event would occur, this would cause a seg fault. (closes issue ASTERISK-18663) Reported by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches: blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283) Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged revisions 347058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-05 17:42 +0000 [r347007] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_analog.h: Restore call progress code for analog ports. Extracting sig_analog from chan_dahdi lost call progress detection functionality. * Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patches: chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me) sig_analog.h.diff (license #5685) patch uploaded by Richard Miller ........ Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-05 15:02 +0000 [r346955] Jonathan Rose * main/pbx.c, /: Resolve duplicate label used in multiple priorities for the same extension. Prior to this patch, if labels with the same name were used for different priorities in the same extension, the new label would be accepted, but it would be unusable since attempts to reach that label would just go to the first one. Now pbx.c detects this, generates a warning in logs, and culls the label before adding it to the dialplan. (closes issue ASTERISK-18807) Reported by: Kenneth Shumard Patches: pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........ Merged revisions 346954 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-05 14:46 +0000 [r346952] Kinsey Moore * res/res_jabber.exports.in, /: Fix chan_jingle/gtalk load regression introduced in r346087 Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy for usage outside res_jabber. Testing of these changes focused on res_jabber itself, so this problem was missed. Reported-by: Michael Spiceland ........ Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-04 10:03 +0000 [r346900] Walter Doekes * /, channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and domain ACL bypass. The code that allowed admins to create users with domain-only uri's had stopped to work in 1.8 because of the reqresp parser rewrites. This is fixed now: if you have a [mydomain.com] sip user, you can register with useraddr sip:mydomain.com. Note that in that case -- if you're using domain ACLs (a configured domain list) -- mydomain.com must be in the allow list as well. Reviewboard r1606 shows a list of registration combinations and which SIP response codes are returned. Review: https://reviewboard.asterisk.org/r/1533/ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes issue ASTERISK-18741) ........ Merged revisions 346899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-02 23:27 +0000 [r346856] Matthew Jordan * channels/chan_sip.c: Update SIP MESSAGE To parsing to correctly handle URI The previous patch (r346040) incorrectly parsed the URI in the presence of a port, e.g., user@hostname:port would fail as the port would be double appended to the SIP message. This patch uses the parse_uri function to correctly parse the URI into its username and hostname parts, and places them in the correct fields in the sip_pvt structure. (issue ASTERISK-18903) Review: https://reviewboard.asterisk.org/r/1597/ 2011-12-02 16:42 +0000 [r346763] Alexandr Anikin * addons/chan_ooh323.c, /, channels/chan_h323.c: Merged revisions 346762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines process null frame pointer returned by ast_rtp_instance_read correctly (closes issue ASTERISK-16697) Reported by: under Patches: segfault.diff (License #5871) patch uploaded by under ........ 2011-12-01 21:14 +0000 [r346701] Richard Mudgett * main/stun.c, /, res/res_stun_monitor.c, configs/res_stun_monitor.conf.sample, include/asterisk/stun.h: Re-resolve the STUN address if a STUN poll fails for res_stun_monitor. The STUN socket must remain open between polls or the external address seen by the STUN server is likely to change. However, if the STUN request poll fails then the STUN server address needs to be re-resolved and the STUN socket needs to be closed and reopened. * Re-resolve the STUN server address and create a new socket if the STUN request poll fails. * Fix ast_stun_request() return value consistency. * Fix ast_stun_request() to check the received packet for expected message type and transaction ID. * Fix ast_stun_request() to read packets until timeout or an associated response packet is found. The stun_purge_socket() hack is no longer required. * Reduce ast_stun_request() error messages to debug output. * No longer pass in the destination address to ast_stun_request() if the socket is already bound or connected to the destination. (closes issue ASTERISK-18327) Reported by: Wolfram Joost Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1595/ ........ Merged revisions 346700 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-12-01 20:37 +0000 [r346565-346698] Jonathan Rose * /, channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing. 183 is actually a session progress message. (closes issue ASTERISK-18925) Reported by: Sebastian Denz Tested by: jrose Patches: asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian Denz (License #6139) ........ Merged revisions 346697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h: r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines Cleaning up chan_sip/tcptls file descriptor closing. This patch attempts to eliminate various possible instances of undefined behavior caused by invoking close/fclose in situations where fclose may have already been issued on a tcptls_session_instance and/or closing file descriptors that don't have a valid index for fd (-1). Thanks for more than a little help from wdoekes. (closes issue ASTERISK-18700) Reported by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged revisions 346564 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-30 19:37 +0000 [r346473] Leif Madsen * /, configs/queues.conf.sample: Update queues.conf.sample documentation. Update the documentation surrounding the use of MONITOR_EXEC to make it more clear that it can be used for both Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413) Reported by: David Woolley Patches: issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026) ........ Merged revisions 346472 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-29 00:00 +0000 [r346349] David Vossel * include/asterisk/message.h, main/message.c: Fixes memory leak in message API. The ast_msg_get_var function did not properly decrement the ref count of the var it retrieves. The way this is implemented is a bit tricky, as we must decrement the var and then return the var's value. As long as the documentation for the function is followed, this will not result in a dangling pointer as the ast_msg structure owns its own reference to the var while it exists in the var container. 2011-11-28 14:32 +0000 [r346293] Stefan Schmidt * res/res_rtp_asterisk.c, /: Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified. (closes issue ASTERISK-18693) Reported by: Davide Dal Fra Review: https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter Doekes ........ Merged revisions 346292 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 22:58 +0000 [r346240] Richard Mudgett * include/asterisk/acl.h, /, channels/chan_skinny.c, channels/chan_h323.c, main/acl.c, channels/chan_iax2.c: Fix calls to ast_get_ip() not initializing the address family. ........ Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 20:45 +0000 [r346145-346198] Walter Doekes * /, channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text() function. In r116240, get_msg_text() got an extra parameter to fix the unwanted addition of trailing newlines to SIP MESSAGE bodies. This caused all linefeeds to be trimmed, which isn't right either. This is a stop-gap; the right fix is to return the original SIP request body. Review: https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan ........ Merged revisions 346147 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, include/asterisk/strings.h: Fix ast_str_truncate signedness warning and documentation. Review: https://reviewboard.asterisk.org/r/1594 ........ Merged revisions 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 17:14 +0000 [r346087] Kinsey Moore * channels/chan_jingle.c, /, include/asterisk/jabber.h, channels/chan_gtalk.c, res/res_jabber.c: Fix res_jabber resource leaks This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 16:19 +0000 [r346040] Matthew Jordan * channels/chan_sip.c: Fixed SendMessage stripping extension from To: header in SIP MESSAGE When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip attempted to validate the hostname or IP Address. In the process, it stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ 2011-11-23 16:10 +0000 [r346031] Terry Wilson * /, res/res_musiconhold.c: Resume playing existing hold music for cached realtime MOH As a result of the fix for ASTERISK-18039, realtime caching MOH no longer properly resumes playing back a file between different holds in the same call. This is because scanning for new files causes the existing file array to be emptied and we were just comparing that the saved pointer to the filename matched the pointer to the filename in a particular position in the array. An easy fix is to save the filename instead of a pointer to it and then do a strcmp instead of comparing the addresses. (closes issue ASTERISK-18912) Review: https://reviewboard.asterisk.org/r/1596/ ........ Merged revisions 346030 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-23 16:06 +0000 [r346029] Paul Belanger * res/res_format_attr_celt.c, res/res_format_attr_silk.c: Added support level for new modules 2011-11-22 23:00 +0000 [r345977] Richard Mudgett * main/dnsmgr.c, /, include/asterisk/dnsmgr.h: Fix dnsmgr entries to ask for the same address family each time. The dnsmgr refresh would always get the first address found regardless of the original address family requested. So if you asked for only IPv4 addresses originally, you might get an IPv6 address on refresh. * Saved the original address family requested by ast_dnsmgr_lookup() to be used when the address is refreshed. ........ Merged revisions 345976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-22 20:31 +0000 [r345924] Walter Doekes * include/asterisk/logger.h, /: Clarify why the AST_LOG_* macros exist next to the LOG_* macros. (issue ASTERISK-17973) ........ Merged revisions 345923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-22 16:40 +0000 [r345882] Paul Belanger * apps/confbridge/conf_config_parser.c: Add missing sound_only_one config variable (closes issue ASTERISK-18895) Reported by: zvision Patches: conf_config_parser.diff (license #5755) patch uploaded by zvision 2011-11-21 21:07 +0000 [r345830] Terry Wilson * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: 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 ........ Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-19 15:10 +0000 [r345640-345683] Tilghman Lesher * /, main/db.c: Update the documentation to better clarify how the existing commands work. Review: https://reviewboard.asterisk.org/r/1593/ ........ Merged revisions 345682 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/db.c: Fix a change in behavior in 'database show' from 1.8. In 1.8 and previous versions, one could use any fullword portion of the key name, including the full key, to obtain the record. Until this patch, this did not work for the full key. Closes issue ASTERISK-18886 Patch: by tilghman Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev 2011-11-17 17:29 +0000 [r345558] Richard Mudgett * /, channels/sig_pri.c: Remove dead code since pri_grab() can never fail. Dead code makes programmers sick. I am sick of looking at it. ........ Merged revisions 345546 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-16 14:49 +0000 [r345488] Jonathan Rose * apps/app_voicemail.c, /: Guarantee messages go into the right folders with multiple recipients Before, using the U flag in Voicemail with multiple recipients would put urgent messages in the INBOX folder for all users past the first thanks to a bug with the message copying function. This would also cause messages to fail to be sent if the INBOX directory hadn't been created for that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged revisions 345487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-15 20:10 +0000 [r345220-345432] Richard Mudgett * /, res/res_agi.c: Make FastAGI HANGUP show up in AGI debug output. * Change from using send() to ast_agi_send() so the HANGUP shows up in the AGI debug output. (closes issue ASTERISK-18723) Reported by: James Van Vleet Patches: jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 345431 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/sig_pri.c: Fix typo in sig_pri using wrong structure name. It is fortunate that the typo does not alter generated code since the e->restart.channel and e->ring.channel members are in the same position. (closes issue ASTERISK-18868) Reported by: zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by zvision ........ Merged revisions 345370 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Make queue log indicate if ADDMEMBER is paused for AMI and realtime. * Add parameter to queue log ADDMEMBER to indicate if the member is paused. (closes issue ASTERISK-18645) Reported by: garlew Patches: paused.diff (License #5337) patch uploaded by garlew Tested by: rmudgett, garlew Review: https://reviewboard.asterisk.org/r/1469/ ........ Merged revisions 345285 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c, configs/sip.conf.sample, UPGRADE-1.8.txt, channels/sip/include/sip.h: Restore SIP DTMF overlap dialing method. The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ ........ Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/pbx.c, /: Fix Progress spelling error in main/pbx.c. (closes issue ASTERISK-18857) Reported by: David M Patches: mainpbx-trivial.patch (License #6326) patch uploaded by David M ........ Merged revisions 345219 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-14 19:10 +0000 [r345164] Terry Wilson * main/channel.c, /: Don't read past end of input when calling write() int blah = 1; ... write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is only valid when new_frames == 1. Otherwise we start reading into adjacent variables declared on the stack. The read end discards what is read, so the values don't matter but it's not a good idea to read past where we want even though new_frames is almost always 1 and should never be large. This patch is basically taken out of kpfleming's eventfd branch, as he mentioned that he remembered fixing it there when I talked to him about this issue. Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged revisions 345163 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-14 19:01 +0000 [r345161] Walter Doekes * channels/sip/include/reqresp_parser.h, /: Update reqresp_parser parse_uri doxygen comments. The issue mentioned in the bug report had been fixed recently by twilson. The reporter included this documentation fix. (closes issue ASTERISK-18572) Reported by: Richard Miller Patch by: Richard Miller (modified) ........ Merged revisions 345160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-14 16:12 +0000 [r345117] Jonathan Rose * apps/app_voicemail.c, /: Moves voicemail setup password entry to the end of the setup process. This change was made because forcegreeting and forcename settings in voicemail could be circumvented by hanging up after entering a password, because the only way voicemail currently observes whether a mailbox is new or not is by checking to see if the password is the same as the mailbox number or not. (closes issue ASTERISK-18282) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/ ........ Merged revisions 345062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-14 15:10 +0000 [r345064] Kinsey Moore * /, channels/chan_sip.c: Ensure that a null vmexten does not cause a segfault When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten was not expected to be null. This change handles that situation to avoid a segfault. ........ Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-12 16:17 +0000 [r344966] Gregory Nietsky * channels/chan_misdn.c, /: mISDN Round Robin break when no channel is available Prevent channels been parsed repetitively. ........ Merged revisions 344965 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-12 00:34 +0000 [r344900] Terry Wilson * /, res/res_musiconhold.c: Don't forget to rescan MOH files for cached realtime classes Realtime MOH class caching was implemented because without it, you would build a completely new MOH class and would start the music over at the beginning each time hold was pressed in a conversation. Unfortunately, this broke re-scanning for file changes for realtime MOH classes. This patch corrects that issue. (closes issue ASTERISK-18039) Review: https://reviewboard.asterisk.org/r/1579/ ........ Merged revisions 344899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 21:58 +0000 [r344845] Walter Doekes * include/asterisk/stringfields.h, include/asterisk/utils.h, /, main/utils.c: Use __alignof__ instead of sizeof for stringfield length storage. Kevin P Fleming suggested that r343157 should use __alignof__ instead of sizeof. For most systems this won't be an issue, but better fix it now while it's still fresh. Review: https://reviewboard.asterisk.org/r/1573 ........ Merged revisions 344843 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 21:50 +0000 [r344842] Matthew Jordan * /, main/file.c: Video format was treated as audio when removed from the file playback scheduler This patch fixes the format type check in ast_closestream and filestream_destructor. Previously a comparison operator was used, but since audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats that have a value greater than the video formats), a bitwise AND operation is used instead. Duplicated code was also moved to filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo Bedrij Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1580/ ........ Merged revisions 344823 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 21:36 +0000 [r344836-344839] Walter Doekes * /, channels/sip/reqresp_parser.c: Remove unneeded if(params) checks in reqresp_parser. Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent reason. There is no way that params could become NULL in that piece of code, so I removed these excess checks again. ........ Merged revisions 344837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/manager.c, /: Fix bad quoting of multiline mxml opaque_data that caused invalid xml. The opaque_data was added and enclosed in single quotes, assuming it would be only a single line. The rest of the lines were appended after the closing quote. (closes issue ASTERISK-18852) Reported by: peep_ on IRC Review: https://reviewboard.asterisk.org/r/1577 ........ Merged revisions 344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 20:14 +0000 [r344770] Kinsey Moore * /, channels/chan_sip.c: Fix regression introduced by SDP fixups If capability is adjusted when switching to UDPTL during fax transmission, fax teardown fails. Make sure capability is only touched if RTP is active. This regression was introduced in R344385. ........ Merged revisions 344769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 18:36 +0000 [r344662-344716] Richard Mudgett * /, channels/chan_sip.c: Check sip.conf maxforwards parameter for range 1 <= x <= 255. JIRA AST-710 ........ Merged revisions 344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/cli.c, /: Make CLI "core show channel" not hold the channel lock during console output. Holding the channel lock while the CLI "core show channel" command is executing can slow down the system. It could block the system if the console output is halted or paused. * Made capture the CLI "core show channel" output into a buffer to be output after the channel is unlocked. * Removed use of C++ keyword as a variable name. out renamed to obuf. * Checked allocation of obuf for failure so will not crash. (closes issue ASTERISK-18571) Reported by: Pavel Troller Tested by: rmudgett ........ Merged revisions 344661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-11 15:33 +0000 [r344609] Jonathan Rose * main/pbx.c, /: Fix a segmentation fault when using an extension with CID matching and no CID. Attempting to call an extension which used Caller ID matching with a channel that has an empty caller id string would result in a segmentation fault. (closes issue ASTERISK-18392 Reported By: Ales Zelenik ........ Merged revisions 344608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-10 23:21 +0000 [r344537-344557] Richard Mudgett * apps/app_macro.c: Fix app_macro.c MODULEINFO section termination. (closes issue ASTERISK-18848) Reported by: Tony Mountifield * /, apps/app_queue.c: Fix potential deadlock calling ast_call() with channel locks held. Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks held. Chan_local attempts to do deadlock avoidance in its ast_call() callback and could deadlock if a channel lock is already held. ........ Merged revisions 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel. It was strange that the AgentCalled AMI event would get most of its information from the incoming channel but then get the CallerID information from the outgoing channel. Before connected line support was added, this information was always the same at this point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham Tested by: rmudgett ........ Merged revisions 344536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-10 21:54 +0000 [r344493] David Vossel * main/bridging.c: Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever. When a conference user enters the DTMF menu they are suspended from the bridge while the channel is handed off to the DTMF feature code. If a user entered this state and hungup, there existed a race condition where the channel could not exit the conference because it was waiting on a signal that would never arrive. This patch fixes that, because it would stupid for me to talk about the problem and commit a patch for something else. (closes issue ASTERISK-18829) Reported by: zvision 2011-11-10 21:14 +0000 [r344386-344440] Kinsey Moore * /, apps/app_meetme.c: Fix another incorrect case with meetme's PIN logic and add documentation This fixes an issue where a user of a dynamic conference was asked for a PIN twice. This also adds documentation to assist in future modifications to the piece of code responsible for PIN checking. (closes issue AST-670) ........ Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c, channels/sip/include/sip.h: Fix several bugs with SDP parsing and well-formedness of responses Fix bug ASTERISK-16558 which dealt with the order of responses to incoming streams defined by SDP. Fix unreported bug where offering multiple same-type streams would cause Asterisk to reply with an incorrect SDP response missing one or more streams without a proper declination. Fix bugs related to a single non-audio stream being offered with responses requesting codecs that were not offered in the initial invite along with an additional audio stream that was not in the initial invite. Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-10 16:25 +0000 [r344334] Matthew Nicholson * res/res_rtp_asterisk.c, /: only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny (modified) ASTERISK-18490 ........ Merged revisions 344330 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 20:53 +0000 [r344271] Richard Mudgett * /, channels/chan_sip.c: Fix deadlock during dialplan reload. Another deadlock between the conlock/hints and channels/channel locking orders. * Don't hold the channel and private lock in sip_new() when calling ast_exists_extension(). (closes issue ASTERISK-18740) Reported by: Byron Clark Patches: sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark Tested by: Byron Clark ........ Merged revisions 344268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 20:07 +0000 [r344175-344216] Terry Wilson * channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c, channels/sip/reqresp_parser.c, channels/sip/include/sip.h: Don't treat a host:port string as a domain The domain matching code prior to 1.8 used to manually remove the port from the host:port string when determining if an incoming request matched the list of domains. When switching to the new parsing functions, the documentation implied that the "domain" was being returned by these functions, when instead it was returning the "hostport" as defined by RFC 3261. This led to confusion and resulted in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch renames the "domain" variables in the parsing functions to "hostport" to more accurately describe what it is that they are returning and also properly truncates the resulting hostport strings when dealing with domain matching. Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged revisions 344215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, tests/test_netsock2.c: Add a unit test for ast_sockaddr_split_hostport Review: https://reviewboard.asterisk.org/r/1575/ ........ Merged revisions 344157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 19:02 +0000 [r344159-344160] Alexandr Anikin * /: delete svn:mergeinfo * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooh323.c, /, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooq931.h, addons/ooh323c/src/ootypes.h, addons/ooh323c/src/oochannels.c: Generate response to Status Enquiry message with Status q.931 message. Some PBXes require this for call status checking (closes issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches: ASTERISK-18748-5.patch (License #5415) patch uploaded by may213 Tested by: Fabrizio Lazzaretti ........ Merged revisions 344158 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 17:14 +0000 [r344103] Kinsey Moore * /, apps/app_meetme.c: Fix pin parameter behavior regression in MeetMe The last time this code was touched (by me), a subtlety was missed based on the difference between needing to check a pin's validity and the need to prompt for a pin. (closes issue ASTERISK-18488) ........ Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-09 15:26 +0000 [r344049] Matthew Nicholson * /, formats/format_wav.c: don't call ltohl() twice on the same value ASTERISK-18739 Patch by: pawel (modified) ........ Merged revisions 344048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-08 21:59 +0000 [r344004] Richard Mudgett * channels/chan_sip.c: Residual changes for Asterisk v10 branch from ASTERISK-18747. Residual changes for Asterisk v10 branch from ASTERISK-18747 after https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs callid hash key change fix. * Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from dialogs_rtpcheck. This is an optimization to avoid an unneeded lock/unlock and object search when using ao2_unlink. * Prevent crash in check_rtp_timeout() if dialog->rtp is NULL. Review: https://reviewboard.asterisk.org/r/1557/ 2011-12-15 Asterisk Development Team * Asterisk 10.0.0 Released. 2011-12-08 Asterisk Development Team * Asterisk 10.0.0-rc3 Released. * Add ASTSBINDIR to the list of configurable paths This patch also makes astdb2sqlite3 and astcanary use the configured directory instead of relying on $PATH. (closes issue ASTERISK-18959) Review: https://reviewboard.asterisk.org/r/1613/ * 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) * Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. When a peer is removed, either by pruning realtime SIP peers or by unloading / loading chan_sip, the MWI subscriptions that were orphaned would still be on the event engine list of valid subscriptions but have a pointer to a peer that no longer was valid. When an MWI event would occur, this would cause a seg fault. (closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1610/ * Fix a change in behavior in 'database show' from 1.8. In 1.8 and previous versions, one could use any fullword portion of the key name, including the full key, to obtain the record. Until this patch, this did not work for the full key. (closes issue ASTERISK-18886) * Default to nat=yes; warn when nat in general and peer differ AST-2011-013. 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/ * Fixed SendMessage stripping extension from To: header in SIP MESSAGE When using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip attempted to validate the hostname or IP Address. In the process, it stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure. (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ 2011-11-15 Asterisk Development Team * Asterisk 10.0.0-rc2 Released. * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK10RCS-3 * Ensure that a null vmexten does not cause a segfault Ensure that a null vmexten does not cause a segfault. When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten was not expected to be null. This change handles that situation to avoid a segfault Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever. When a conference user enters the DTMF menu they are suspended from the bridge while the channel is handed off to the DTMF feature code. If a user entered this state and hungup, there existed a race condition where the channel could not exit the conference because it was waiting on a signal that would never arrive. This patch fixes that, because it would stupid for me to talk about the problem and commit a patch for something else. (closes issue ASTERISK-18829) Reported by: zvision * Fix app_macro.c MODULEINFO section termination. (closes issue ASTERISK-18848) Reported by: Tony Mountifield 2011-11-08 Asterisk Development Team * Asterisk 10.0.0-rc1 Released. * Test results: http://bamboo.asterisk.org/browse/AST10-LUCID-317 2011-11-08 19:27 +0000 [r343944] wdoekes : * /, pbx/pbx_config.c: Fix crash when dialplan remove include is called with too few arguments. "dialplan remove include x from y" crashed when the amount of arguments was less than 6. (closes issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by: Andrey Solovyev ........ Merged revisions 343936 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-08 18:29 +0000 [r343900] David Vossel * channels/chan_sip.c: Fixes regression caused by r343635 There was a missing unlock for a function return that is only present in Asterisk 10 and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by: Michael L. Young Patches: asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young 2011-11-08 18:01 +0000 [r343852] Richard Mudgett * /, channels/chan_sip.c, main/acl.c: Fixed reference to incorrect variable if unknown host configured crash. * Fixed a LOG_ERROR message referencing the config variable list v that had previously been processed and became NULL. * Added error return value set that was missing in an ast_append_ha() error return path. (closes issue ASTERISK-18743) Reported by: Michele Patches: issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes Tested by: Michele ........ Merged revisions 343851 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-08 13:26 +0000 [r343789-343792] Leif Madsen * /: Recorded merge of revisions 343791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Fix boo-boo in prep_tarball script. A hardcoded a branch number was in the prep_tarball which could not work. Changed it to the variable. * build_tools/prep_tarball: Fix boo-boo in prep_tarball script. A hardcoded a branch number was in the prep_tarball which could not work. Changed it to the variable. 2011-11-07 22:37 +0000 [r343743] Kinsey Moore * channels/chan_sip.c: Make "sip show settings" CLI command get RPID flags from the right global page The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command pulled the flags from the incorrect global flags page. These are now read from sip global flags page 0. (closes issue AST-711) 2011-11-07 21:42 +0000 [r343691] Matthew Nicholson * /, channels/chan_sip.c: respect case changes in peer names on sip reload ASTERISK-18669 ........ Merged revisions 343690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-07 21:27 +0000 [r343677] Richard Mudgett * /, channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly changing dialogs hash key callid. Changing an object value used as a container key requires removing the object from the container and reinserting it. * Created change_callid_pvt() to call instead of build_callid_pvt(). The change_callid_pvt() will correctly change the dialog callid so the ao2 conainter can explicitly unlink it. ........ Merged revisions 343637 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-07 20:31 +0000 [r343635] Kinsey Moore * /, channels/chan_sip.c: Prevent BLF subscriptions from causing deadlocks Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks. This function now requires that both the peer and associated pvt be unlocked before it is called for cases where peer and peer->mwipvt form a circular reference. (closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1563/ ........ Merged revisions 343621 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-07 19:55 +0000 [r343580] wdoekes : * main/udptl.c, UPGRADE.txt: Correct the default udptl port range. The udptl port range was defined as 4000-4999 in the udptl.conf.sample, as 4500-4599 if you didn't have a config and 4500-4999 if your config was broken. Default is now 4000-4999. (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher Review: https://reviewboard.asterisk.org/r/1565 2011-11-07 19:51 +0000 [r343578] Richard Mudgett * /, channels/chan_sip.c: Fix deadlock if peer is destroyed while sending MWI notice. A dialog cannot be destroyed by the ao2_callback dialog_needdestroy because of a deadlock between the dialogs container lock and the RWLOCK of the events subscription list. * Create dialogs_to_destroy container to hold dialogs that will be destroyed. * Ensure that the event subscription callback will never happen with an invalid peer pointer by making the event callback removal the first thing in the peer destructor callback. NOTE: This particular deadlock will not happen with Asterisk 10, but some of the changes still apply. (closes issue ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/ ........ Merged revisions 343577 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-07 18:39 +0000 [r343533] Matthew Nicholson * main/format.c: list all of the codecs associated with a particular format id for CLI command "core show codec" AST-699 2011-11-04 15:11 +0000 [r343445] Alexandr Anikin * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/dlist.c, /, addons/ooh323c/src/dlist.h, addons/ooh323c/src/printHandler.c: Final fix memleaks in GkClient codes, same for Timer codes. (these memleaks stop development of gk codes, now i can continue) Fix printHandler 'Unbalanced Structure' issues with locking printHandler data for single thread. ........ Merged revisions 343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-03 20:31 +0000 [r343393] wdoekes : * /, res/res_config_sqlite.c: Fix sqlite config driver segfault and broken queries The sqlite realtime handler assumed you had a static config configured as well. The realtime multientry handler assumed that you weren't using dynamic realtime. (closes issue ASTERISK-18354) (closes issue ASTERISK-18355) Review: https://reviewboard.asterisk.org/r/1561 ........ Merged revisions 343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-03 19:57 +0000 [r343337] Richard Mudgett * /, funcs/func_dialgroup.c: Remove invalid flag given to iterator in func_dialgroup.c ........ Merged revisions 343336 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-03 15:39 +0000 [r343221-343277] Terry Wilson * /, channels/sip/include/sip.h: Make room for the fax detect flags The original REGISTERTRYING flag, in addition to being impossible to check, also encroached on the space for the flag above it. This patch moves the flags that were below REGISTERTRYING back to where they were as though we had just removed the REGISTERTRYING option. ........ Merged revisions 343276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c, channels/sip/include/sip.h: Remove registertrying option in chan_sip This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ ........ Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-02 22:24 +0000 [r343158-343192] wdoekes : * /, channels/chan_sip.c: Fix improper warning introduced by r342927 and more tweaks Changeset r342927 introduced a warning which was only supposed to be emitted when a found realtime peer had an empty (or no) name. It turned out that there were some inconsistencies left. Now found peers with an empty name are explicitly ignored like before r342927 but better. Reviewed by: Stefan Schmidts, Terry Wilson Review: https://reviewboard.asterisk.org/r/1560 ........ Merged revisions 343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * include/asterisk/stringfields.h, include/asterisk/utils.h, /, main/utils.c: Ensure that string field lengths are properly aligned Integers should always be aligned. For some platforms (ARM, SPARC) this is more important than for others. This changeset ensures that the string field string lengths are aligned on *all* platforms, not just on the SPARC for which there was a workaround. It also fixes that the length integer can be resized to 32 bits without problems if needed. (closes issue ASTERISK-17310) Reported by: radael, S Adrian Reviewed by: Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review: https://reviewboard.asterisk.org/r/1549 ........ Merged revisions 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-02 19:33 +0000 [r343048-343103] Leif Madsen * /, apps/app_authenticate.c: Add note about how Authenticate() application with option 'd' works. (closes issue ASTERISK-17422) Reported by: Leif Madsen ........ Merged revisions 343102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, configs/queues.conf.sample: Update documentation for leastrecent strategy. In queues.conf.sample the leastrecent strategy was incorrectly described. Now updated to reflect how the strategy actually checks peers. (closes issue ASTERISK-17854) Reported by: Sebastian Denz Patches: queues.conf-doc_issue.patch (License #6139) ........ Merged revisions 343047 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-02 13:45 +0000 [r342991] Kevin P. Fleming * /, apps/app_meetme.c: Modify comments in MeetMe application documentation about DAHDI. The MeetMe application documentation has some comments about usage of DAHDI, and they were a bit outdated relative to modern DAHDI releases. This patch changes the comment to just tell the user that a functional DAHDI timing source is required, and no longer mention 'dahdi_dummy', since that module does not exist in current DAHDI releases. ........ Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-11-01 20:58 +0000 [r342870-342929] wdoekes : * /, channels/chan_sip.c, configs/extconfig.conf.sample, include/asterisk/config.h, main/config.c: Several fixes to the chan_sip dynamic realtime peer/user lookup There were several problems with the dynamic realtime peer/user lookup code. The lookup logic had become rather hard to read due to lots of incremental changes to the realtime_peer function. And, during the addition of the sipregs functionality, several possibilities for memory leaks had been introduced. The insecure=port matching has always been broken for anyone using the sipregs family. And, related, the broken implementation forced those using sipregs to *still* have an ipaddr column on their sippeers table. Thanks Terry Wilson for comprehensive testing and finding and fixing unexpected behaviour from the multientry realtime call which caused the realtime_peer to have a completely unused code path. This changeset fixes the leaks, the lookup inconsistenties and that you won't need an ipaddr column on your sippeers table anymore (when you're using sipregs). Beware that when you're using sipregs, peers with insecure=port will now start matching! (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1395 ........ Merged revisions 342927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * contrib/realtime/mysql/sipfriends.sql (removed), contrib/realtime/mysql/sippeers.sql (added), configs/res_config_mysql.conf.sample, /, configs/extconfig.conf.sample, configs/res_ldap.conf.sample, res/res_realtime.c, UPGRADE-1.8.txt, configs/dbsep.conf.sample, main/config.c: Cleanup references to sipusers and sipfriends dynamic realtime families Somewhere between 1.4 and 1.8 the sipusers family has become completely unused. Before that, the sipfriends family had been obsoleted in favor of separate sipusers and sippeers families. Apparently, they have been merged back again into a single family which is now called "sippeers". Reviewed by: irroot, oej, pabelanger Review: https://reviewboard.asterisk.org/r/1523 ........ Merged revisions 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-31 17:46 +0000 [r342824] Richard Mudgett * main/format.c, main/format_cap.c: Misc format capability fixes. * Fixed typo in format_cap.c:joint_copy_helper() using the wrong variable. * Fix potential race between checking if an interface exists and adding it to the container in format.c:ast_format_attr_reg_interface(). * Fixed double rwlock destroy in format.c:ast_format_attr_init() error exit path. * Simplified format.c:find_interface() and format.c:has_interface(). 2011-10-31 16:04 +0000 [r342770] Matthew Jordan * main/pbx.c, /, channels/chan_iax2.c: Fixed invalid memory access when adding extension to pattern match tree When an extension is removed from a context, its entry in the pattern match tree is not deleted. Instead, the extension is marked as deleted. When an extension is removed and re-added, if that extension is also a prefix of another extension, several log messages would report an error and did not check whether or not the extension was deleted before accessing the memory. Additionally, if the extension was already in the tree but previously deleted, and the pattern was at the end of a match, the findonly flag was not honored and the extension would be erroneously undeleted. Additionaly, it was discovered that an IAX2 peer could be unregistered via the CLI, while at the same time it could be scheduled for unregistration by Asterisk. The unregistration method now checks to see if the peer was already unregistered before continuing with an unregistration. (closes issue ASTERISK-18135) Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1526 ........ Merged revisions 342769 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-30 02:21 +0000 [r342715] Terry Wilson * res/res_calendar.c: Don't crash on empty notify channel 2011-10-29 04:26 +0000 [r342662] Richard Mudgett * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c: Fix AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable. AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an iteration or before AST_LIST_REMOVE_CURRENT() without corrupting the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the list if AST_LIST_INSERT_BEFORE_CURRENT() or AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed cut and paste error using the wrong variable in AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-27 20:10 +0000 [r342605] Matthew Nicholson * main/dsp.c: tweak the v21 detector to detect an additional pattern of hits and misses 2011-10-27 19:41 +0000 [r342546-342603] Jonathan Rose * res/res_rtp_multicast.c, /: Fix sequence number overflow over 16 bits causing codec change in RTP packets. Sequence number was handled as an unsigned integer (usually 32 bits I think, more depending on the architecture) and was put into the rtp packet which is basically just a bunch of bits using an or operation. Sequence number only has 16 bits allocated to it in an RTP packet anyway, so it would add to the next field which just happened to be the codec. This makes sure the sequence number is set to be a 16 bit integer regardless of architecture (hopefully) and also makes it so the incrementing of the sequence number does bitwise or at the peak of a 16 bit number so that the value will be set back to 0 when going beyond 65535 anyway. (closes issue ASTERISK-18291) Reported by: Will Schick Review: https://reviewboard.asterisk.org/r/1542/ ........ Merged revisions 342602 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, res/res_jabber.c: Cleanup reference leaks in res_jabber res_jabber.c had a number of places where astobjs would be referenced and have their reference counts bumped without having a dereference made before the object lost scope. This patch adds a number of ASTOBJ_UNREFs to resolve that. Review: https://reviewboard.asterisk.org/r/1478/ ........ Merged revisions 342545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-25 22:05 +0000 [r342485-342488] Richard Mudgett * /, main/astobj2.c: Check fopen return value for ao2 reference debug output. Reported by: wdoekes Patched by: wdoekes Review: https://reviewboard.asterisk.org/r/1539/ ........ Merged revisions 342487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/sig_pri.c: Change D-channel warning to be less confusing on non-NFAS setups. The "No D-channels available! Using Primary channel as D-channel anyway!" WARNING message has been confusing on non-NFAS setups. The message refers to things that are NFAS specific. * Changed the warning to several different warnings to be more accurate for the situation and less confusing as a result: "No D-channels up! Switching selected D-channel from X to Y.", "No D-channels up!", and "D-channel is down!". ........ Merged revisions 342484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-25 21:10 +0000 [r342381-342436] Terry Wilson * /, apps/app_queue.c: Use int for storing ao2_container_count instad of size_t AST-676 ........ Merged revisions 342435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Simplify queue membercount code Despite an ominous sounding comment stating that membercount was for "logged in" members only and thus we couldn't use ao2_container_count(), I could not find a single place in the code where that seemed to be accurate. The only time we decremented membercount was when we were marking something dead or actually removing it. The only places we incremented it were either after ao2_link(), or trying to correct for having set it to 0 during a reload. In every case where we were correcting the value, it seemed that we were trying to make the count actually match what ao2_container_count() would return. The only place I could find where we made a determination about something being "logged in" or not, we didn't trust the membercount, but instead looked at devicestate, paused, etc. This patch removes membercount, replaces its use with ao2_container_count, and manually adds the results of ao2_container_count to a "membercount" field for ast_data queue query results. This patch also would fix AST-676, but as it is slightly riskier than the previously committed fix, the two commits have been made separately. Reivew: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_queue.c: Properly update membercount for reloaded members Since q->membercount is set to 0 before reloading, it is important to increment it again for reloaded members as well as added. (closes issue AST-676) Review: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-25 19:08 +0000 [r342277-342329] Kinsey Moore * pbx/pbx_spool.c, /: Fix compilation on Snow Leopard/FreeBSD for pbx_spool.c One of the changes in the recent spool handling of hardlinks patch was just outside a HAVE_INOTIFY block and caused compilation to fail in some build environments. This has been corrected. ........ Merged revisions 342328 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * pbx/pbx_spool.c, /: Merged revisions 342276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) | 18 lines Fix spool handling to allow call files to be hardlinked into place This fixes the inotify code to handle call files being hardlinked into the spool directory. The smsq utility does this, instead of rename(), to ensure that it cannot accidentally overwrite an existing spool file. A rename() might do that, but link() will definitely not. The inotify code had broken this, because it would wait for an IN_CLOSE_WRITE event on the file... which was never forthcoming, since it was never opened. Now we look for IN_OPEN events following the IN_CREATE event, and only wait for an IN_CLOSE_WRITE if the file was actually opened. Patch-by: dwmw2 (closes issue ASTERISK-18331) Review: https://reviewboard.asterisk.org/r/1391/ ........ 2011-10-25 01:25 +0000 [r342224] Terry Wilson * /, include/asterisk/config.h, main/config.c: Return NULL when no results returned for realtime_multientry It was not documented what the return value should be when no entries were returned with the multientry realtime callback. This change forces consistent behavior even if the backends return an empty ast_config. Review: https://reviewboard.asterisk.org/r/1521/ ........ Merged revisions 342223 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-24 22:32 +0000 [r342183] Richard Mudgett * include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add missing link/unlink nolock debug defines. 2011-10-24 19:51 +0000 [r342062] Jonathan Rose * /, channels/chan_sip.c: Outbound SIP OPTIONS messages will now include fromuser of related peer. This behavior matches up more closely with the way invite/register/etc are handled. This patch also modifies some adjacent code for code style compliance. Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded by Jeremy Kister (license #6232) ........ Merged revisions 342061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-24 07:31 +0000 [r341920-342017] Gregory Nietsky * apps/app_queue.c: queues container needs locking when using the OBJ_NOLOCK flag * apps/app_queue.c: Remove some ref leaks and a return without unlock. There some resource leaks introduced in asterisk 10 make sure that locks are not held on return and we release ref's held. * /, apps/app_queue.c: Revert Janitor patch 341920 For now * /, apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial patch is related to work on RB1538 ........ Merged revisions 341906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-21 16:42 +0000 [r341807-341810] Matthew Nicholson * /, pbx/pbx_lua.c: only process args that exist ASTERISK-18395 ........ Merged revisions 341809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, pbx/pbx_lua.c: don't limit the length of app and function arguments ASTERISK-18395 ........ Merged revisions 341806 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-20 21:58 +0000 [r341718] Richard Mudgett * include/asterisk/features.h, /, main/features.c, res/res_agi.c: Fix AGI exec Park to honor the Park application parameters. The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-20 21:27 +0000 [r341665-341707] Paul Belanger * /, funcs/func_callerid.c: Fixed typo from previous commit ........ Merged revisions 341704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, funcs/func_callerid.c: Updated documentation for the optional CID parameter with CALLERID ........ Merged revisions 341664 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-20 18:20 +0000 [r341580-341599] Gregory Nietsky * configs/queues.conf.sample: add documentation for check_state_unknown in configs/queues.conf.sample app_queue allows calls to members in a "Unknown" state to be treated as available setting check_state_unknown = yes will cause app_queue to query the channel driver to better determine the state this only applies to queues with ringinuse or ignorebusy set appropriately. * CHANGES, apps/app_queue.c: Add option to check state when state is unknown r341486 reverts r325483 this is a rework of the patch. optimize to minimize load. add option check_state_unknown to control whether a member with unknown device state is checked there is a small % chance that calls will be sent to the member when they on a call. app_queue will see a device with unknown state as available and does not try verify the state without this option enabled. Review: https://reviewboard.asterisk.org/r/1535/ 2011-10-20 15:14 +0000 [r341530] Terry Wilson * /, include/asterisk/strings.h: Clean up ast_check_digits The code was originally copied from the is_int() function in the AEL code. wdoekes pointed out that the function should take a const char* and that their was an unneeded variable. This is now fixed. ........ Merged revisions 341529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-19 21:23 +0000 [r341486] Matthew Nicholson * apps/app_queue.c: Fix a performance regression introduced in r325483. The regression was caused by a call to ast_parse_device_state() in app_queue's ring_entry() function. The ast_parse_device_state() function eventually calls ast_channel_get_full() with a channel name prefix which causes it to walk the channel list causing massive lock contention and slow downs. This patch fixes the regression by removing the call to ast_parase_device_state() which should be unnecessary. Queue member device state should be maintained by device state events. Some users have seen instances where busy agents were called when they shouldn't have, which is the reason the call to ast_parse_device_state() was added. That change appears to have resolved that issue but also causes this performance regression. There may still be issues with queue member status, and if so, alternative methods should be investigated to resolve them. AST-695 2011-10-19 19:01 +0000 [r341436] Paul Belanger * /, channels/chan_gtalk.c: Outgoing calls with Google Voice Google has recently make some changes (again) to their protocol. Rather then patching asterisk to flip between the two different methods, we now allow both. Lets hope this keeps Google Voice happy for a while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311) ........ Merged revisions 341435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-19 07:42 +0000 [r341380] Terry Wilson * /, channels/chan_sip.c, include/asterisk/strings.h: Don't use is_int() since it doesn't link well on all platforms Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-19 07:23 +0000 [r341377] Stefan Schmidt * /, channels/chan_sip.c: Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS ........ Merged revisions 341366 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-18 23:42 +0000 [r341315] Terry Wilson * /, channels/chan_sip.c: Don't resolve numeric hosts or contact unresolved hosts If a SIP dial string contains a numeric hostname that is not a peer name, don't try to resolve it as it is unlikely that someone really means Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that create_addr returns -1 if an address isn't resolved so that we don't attempt to send SIP requests to an address that doesn't resolve. (closes issue ASTERISK-17146, ASTERISK-17716) Review: https://reviewboard.asterisk.org/r/1532/ ........ Merged revisions 341314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-18 23:33 +0000 [r341313] Alexandr Anikin * addons/chan_ooh323.c, /: Merged revisions 341312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3 lines fix issue on channel numbering (calls could have same channel number on heavy loaded system) ........ 2011-10-18 21:11 +0000 [r341255] Richard Mudgett * channels/chan_mgcp.c, include/asterisk/features.h, channels/chan_dahdi.c, channels/sig_analog.c, /, channels/chan_sip.c, main/features.c, channels/chan_iax2.c, channels/sip/include/sip.h: More parking issues. * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-17 17:36 +0000 [r341190] Terry Wilson * /, channels/chan_sip.c: Initialize variables before calling parse_uri If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) ........ Merged revisions 341189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-17 16:53 +0000 [r341148] Tzafrir Cohen * /, pbx/pbx_realtime.c: Remove an unused include of md5.h Unused include of asterisk/md5.h in pbx_realtime.c . A commit needed to test the commit message. Merged-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@341074 2011-10-17 16:38 +0000 [r341122-341146] Paul Belanger * tests/test_format_api.c: Set 'core' support level for test_format_api.c * apps/app_voicemail.c, /: Multiple revisions 341108,341112 ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core' support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged revisions 341108,341112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-17 16:18 +0000 [r341094] Jason Parker * CHANGES: Add information about limitations of new codec support in channel drivers. (issue ASTERISK-18680) 2011-10-17 15:39 +0000 [r341089] Terry Wilson * /, channels/chan_sip.c: Don't try to remove peers without IPs from peers_by_ip (closes issue ASTERISK-18696) ........ Merged revisions 341088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-14 21:36 +0000 [r341023] Kevin P. Fleming * /, build_tools/embed_modules.xml, Makefile.moddir_rules: Change the internal name of the menuselect options that are used to control whether modules are embedded or not; using just the bare category name led to accidentally enabling these options when users used the wrong "--enable" operation on the menuselect command line. Now the internal option names are prefixed with "EMBED_", so they won't be the same as the name of the category containing the modules they control the embedding of. ........ Merged revisions 341022 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-14 20:50 +0000 [r340971] Kinsey Moore * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions 340970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ 2011-10-14 18:23 +0000 [r340931] Jonathan Rose * utils/utils.xml, funcs/func_jitterbuffer.c: Some additional module documentation changes for 10 for the menuselect change. (issue ASTERISK-18268) 2011-10-14 16:39 +0000 [r340879] Terry Wilson * main/channel.c, /: Avoid unnecessary WARNING message Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid displaying a WARNING message. (closes issue ASTERISK-18610) Patch by: Kristijan_Vrban ........ Merged revisions 340878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-14 16:18 +0000 [r340868] Jonathan Rose * funcs/func_realtime.c, build_tools/cflags.xml, utils/utils.xml, /, res/res_fax.c, apps/app_celgenuserevent.c, codecs/codec_dahdi.c, apps/app_system.c, res/res_curl.c: Fixes some support level info so that it can be read by menuselect. (issue ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/ ........ Merged revisions 340863 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-13 22:54 +0000 [r340810] Richard Mudgett * /, main/features.c: Fix DTMF blind transfer continuing to execute dialplan after transfer. Party A calls Party B. Party A DTMF blind transfers Party B to Party C. Party A channel continues to execute dialplan. * Fixed the return value of builtin_blindtransfer() to return the correct value after a transfer so the dialplan will not keep executing. * Removed unnecessary connected line update that did not really do anything. * Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer(). * Fixed leak of xferchan for failure cases in check_goto_on_transfer(). * Updated debug messages in builtin_blindtransfer() and check_goto_on_transfer(). (closes issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 340809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-13 08:46 +0000 [r340770] Gregory Nietsky * channels/chan_sip.c: Only send MWI Notify on register if the registration is successful. lastmsgssent was removed from chan_sip and the old behavior of sending a mwi notify on register [except when subscribemwi is set] was restored but this must only happen when registration succeeds. leaking information for unsuccessful registrations is not secure. 2011-10-13 06:59 +0000 [r340718] Stefan Schmidt * channels/chan_sip.c: Merged revisions 340717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) | 3 lines storing the route-set also on a 181 response not only on 180,182 or 183. ........ 2011-10-13 06:56 +0000 [r340578-340716] Terry Wilson * /, channels/chan_sip.c: Initialize ast_sockaddr before calling ast_sockaddr_resolve Avoid possible jump based on unitialized value ........ Merged revisions 340715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, res/res_config_sqlite.c: Don't skip the query field on a realtime multi query There is no documented reason to not add the query field to the varlist returned by a realtime multi query, despite the config category being set to its value. Of course, there is no documentation that the category should be set to the value either. There is lots of no documentation when it comes to realtime. But, other engines do not skip this field so I am forcing this backend to follow the convention, because not doing so is very silly. ........ Merged revisions 340662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, channels/chan_sip.c: Merged revisions 340534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) | 9 lines Update SIP realtime fullcontact regardless of caching We should update the fullcontact field in the realtime table whether or not rtcachefriends is set. There is no reason to treat a non-cached realtime entity differently than a cached in this regard. (closes issue ASTERISK-18446) Reported by: wdoekes ........ 2011-10-12 20:33 +0000 [r340577] Stefan Schmidt * /, channels/chan_sip.c: Merged revisions 340576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) | 3 lines Store route-set from provisional SIP responses so early-dialog requests can be routed properly ........ 2011-10-12 20:08 +0000 [r340471-340523] Richard Mudgett * channels/chan_dahdi.c, /: Initialize the PRI channel alarms properly on startup. The PRI channel alarms were initialized with an inverted sense. (closes issue ASTERISK-18710) Reported by: Tzafrir Cohen ........ Merged revisions 340522 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * /, apps/app_meetme.c: Update MeetMe p and X option documentation when interacting with the s option. ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 ........ Merged revisions 340470 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-12 16:28 +0000 [r340419] Paul Belanger * /, channels/chan_sip.c: Fix verbose messages when IPv6 logic was added (closes issue ASTERISK-18612) Reported by: Tim Osman ........ Merged revisions 340418 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-11 21:05 +0000 [r340281-340366] Richard Mudgett * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c: Add protection for SS7 channel allocation and better glare handling. * Added a CLI "ss7 show channels" command that might prove useful for future debugging. * Made the incoming SS7 channel event check and gripe message uniform. * Made sure that the DNID string for an incoming call is always initialized. (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 340365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * channels/sip/include/dialog.h, /, channels/chan_sip.c: Fix some potential deadlocks pointed out by helgrind. * Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * main/manager.c, /, include/asterisk/manager.h: Convert registered AMI actions to ao2 objects. * Fixed race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential memory leak if an AMI action failed to get registered because is already was registered. Part of the ao2 conversion. * Fixed AMI ListCommands action not walking the actions list with a lock held. * Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage. * Fix AMI Originate action Variable header requiring a space after the header colon. Reported by Yaroslav Panych on the asterisk-dev list. * Increased the number of listed variables allowed per AMI Originate action Variable header to 64. * Fixed AMI GetConfigJSON action output format. * Fixed usage of res contents outside of scope in append_channel_vars(). * Fixed inconsistency of config file channelvars option. The values no longer accumulate with every channelvars option in the config file. Only the last value is kept to be consistent with the CLI "manager show settings" command. (closes issue ASTERISK-18479) Reported by: Jaco Kroon ........ Merged revisions 340279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-10-11 18:41 +0000 [r340280] Tzafrir Cohen * main/channel.c, /, main/sha1.c, include/asterisk/sha1.h: Update SHA1 code to RFC 6234 RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 2011-10-10 22:55 +0000 [r340219-340222] Terry Wilson * main/db.c: On astdb conversion, also warn about permissions requirements The user running Asterisk must have permission to the directory the Asterisk database resides in since SQLite 3 needs to be able to create a journal file. (closes issue ASTERISK-18174) * utils/astdb2bdb.c (added): Add a missing file for the astdb2bdb conversion utility * utils/Makefile, utils/utils.xml, UPGRADE.txt: Add astdb conversion utility for Berkeley to SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10 they can use the astdb2bdb utility to convert the database back to the Berkeley format that Asterisk 1.8 uses. Review: https://reviewboard.asterisk.org/r/1502/ 2011-10-10 20:30 +0000 [r340165] Matthew Jordan * /, channels/chan_sip.c: Merged revisions 340164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) | 13 lines Updated chan_sip to place calls on hold if SDP address in INVITE is ANY This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::. In this case, the call should be placed on hold. Previously, we checked for the address being null; this patch keeps that behavior but also checks for the ANY IP addresses. Review: https://reviewboard.asterisk.org/r/1504/ (closes issue ASTERISK-18086) Reported by: James Bottomley Tested by: Matt Jordan ........ 2011-10-10 14:15 +0000 [r340109] Matthew Nicholson * main/loader.c, main/xmldoc.c, main/pbx.c, main/manager.c, /, res/res_fax.c, apps/app_fax.c, include/asterisk/module.h, res/res_agi.c, include/asterisk/xmldoc.h, doc/appdocsxml.dtd: Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ 2011-10-09 22:18 +0000 [r339992-340031] Damien Wedhorn * channels/chan_skinny.c: Return -1 to skinny_session if register rejected. If device registration is rejected, return -1 so that the session is destroyed immediately. Previously, a segfault would occur on a graceful shutdown if a register is rejected and the skinny_session has not yet timed out. * channels/chan_skinny.c: Remove log message on traverse session list. On destroying a session, a list of sessions is traversed to find the matching session. For each session not matching, skinny erroneously logged that the session was not matched. While technically correct the message was misleading, and tended to indicate errors that were not there. 2011-10-09 01:18 +0000 [r339831-339942] igorg : * channels/chan_unistim.c, /: Merged revisions 339938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) | 6 lines Fix compilation issue, caused by missed session structure (closes issue ASTERISK-18694) Reported by: alex70 ........ * channels/chan_unistim.c, /: Merged revisions 339884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | 7 lines Fix segfault in Unistim channel (closes issue ASTERISK-18638) Reported by: jonnt ........ * channels/chan_unistim.c, /: Merged revisions 339830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | 8 lines Fix char array cast as short array in send_client() function (for ARM platform) (closes issue ASTERISK-17314) Reported by: jjoshua ........ 2011-10-07 19:36 +0000 [r339777] Richard Mudgett * /, apps/app_url.c: Merged revisions 339776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011) | 5 lines Initialize option flags for SendURL application. (closes issue ASTERISK-18574) Reported by: marcelloceschia ........ 2011-10-06 23:08 +0000 [r339722] Damien Wedhorn * channels/chan_skinny.c: Reject v17 skinny devices in Asterisk10 Small fix for Asterisk10 to reject skinny devices with skinny firmware version17 and above. Review: https://reviewboard.asterisk.org/r/1497/ 2011-10-06 22:58 +0000 [r339720] Richard Mudgett * /, configure, include/asterisk/autoconfig.h.in, configure.ac, autoconf/ast_ext_lib.m4: Merged revisions 339719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines Fix regression in configure script for libpri capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2 persistence issues with some telcos. ASTERISK-18535 attempted to fix the unexpected requirement that libpri *must* have that feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI optional features required. Unfortunately, I thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and deleted those lines for libpri. The result was the HAVE_PRI_xxx defines that control the ability to use optional libpri features were also deleted. * Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional features in a library that the source code could take advantage of if the code supports the feature. (closes issue ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........ 2011-10-06 20:47 +0000 [r339681] Damien Wedhorn * channels/chan_skinny.c: Fixed segfault on core stop gracefully. There was an issue that the cap and confcap pointers for each line and device were being memcpy'd so they all pointed to the same ast_format_cap. On destroying, a segfault occured on the second call to the same struct. skinny reload now works again as well. Tested by snuff (in trunk) and myself. 2011-10-06 17:53 +0000 [r339626] Richard Mudgett * main/udptl.c, /, channels/chan_sip.c: Merged revisions 339625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) | 18 lines Fix debugging messages generated by 'udptl debug'. * Makes chan_sip set the tag to the channel name. * Fixes received debug message sequence number. * Removed tx/rx debug message type since it was hard coded to 0. * Made udptl.c logged message header consistent if possible: "UDPTL (%s): ". * Removed unused rx_expected_seq_no from struct ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P. Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Matthew Nicholson ........ 2011-10-06 13:43 +0000 [r339586] Leif Madsen * /, build_tools/prep_tarball: Merged revisions 339566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011) | 8 lines Update prep_tarball script to download pre-exported documentation. I've updated the prep_tarball script to now download the pre-exported documentation from the Asterisk wiki. This will give us more control over what is being included in the tarball releases, and will make both the PDF and HTML exported documentation look much better (especially when viewing from a console). (Closes issue ASTERISK-18677) ........ 2011-10-05 17:01 +0000 [r339508-339512] Richard Mudgett * apps/app_dial.c, /: Merged revisions 339511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05 Oct 2011) | 1 line Fix Dial F option notes formatting. ........ * main/manager.c, /: Merged revisions 339504,339506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011) | 7 lines Add missing documentation of required AMI action Challenge AuthType header. (closes issue ASTERISK-18554) Reported by: Vlad Povorozniuc Patches: __20110919-manager-challenge-docs.patch.txt (license #4999) patch uploaded by Leif Madsen ........ r339506 | rmudgett | 2011-10-05 11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI action Challenge. ........ 2011-10-05 16:32 +0000 [r339507] Matthew Nicholson * /, res/res_fax.c: Merged revisions 339505 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct 2011) | 3 lines The app name in the documentation must match what we register the application as. ........ 2011-10-05 06:28 +0000 [r339463] Gregory Nietsky * res/res_fax.c: Only change the capabilities on the gateway when the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. 2011-10-04 22:56 +0000 [r339407] Richard Mudgett * Makefile, /: Merged revisions 339406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011) | 8 lines Make always create the MOH directory (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael Keuter ........ 2011-10-04 19:44 +0000 [r339298-339353] Jonathan Rose * /, main/say.c: Merged revisions 339352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) | 12 lines Removes improper use of sound 'and' in German language mode from application saynumber Asterisk would say 'Five hundert und sechs und zwanzig' instead of 'Five hundert sechs und zwanzig'... which is both weird sounding and wrong. This patch makes sure Asterisk will only say the 'and' word between the single digit and double digit places. (closes issue ASTERISK-18212) Reported By: Lionel Elie Mamane Patches: upstream_germand_no_and.diff (License #5402) uploaded by Lionel Elie Mamane ........ * /, res/res_jabber.c: Merged revisions 339297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) | 13 lines Reverting revision 333265 due to component connection problems it introduces. I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this problem, but first it seems prudent to remove this rather broad attempt to fix it and instead approach this problem either from the same angle but looking only at canceling (or possibly rescheduling) the send when we absolutely know it will cause a segfault or, if that can't be easily accomplished, strictly from the devstate side of things. Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe. (issue ASTERISK-18626) (issue ASTERISK-18078) ........ 2011-10-04 11:49 +0000 [r339245] Alexandr Anikin * addons/ooh323c/src/memheap.c, /: Merged revisions 339244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2 lines fix forget declaration in previous change ........ 2011-10-03 20:13 +0000 [r339145-339148] Leif Madsen * /, channels/chan_sip.c: Merged revisions 339147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011) | 6 lines Remove duplicated Maxforwards line in AMI output. (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by Jacek Konieczny ........ * apps/app_dial.c, /: Merged revisions 339144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011) | 6 lines Make documentation for Dial() options 'F' and 'F()' more clear. (Closes issue ASTERISK-18646) Reported by: Physis Heckman Tested by: Richard Mudgett ........ 2011-10-03 18:52 +0000 [r339089] Alexandr Anikin * addons/ooh323c/src/memheap.c, /: Merged revisions 339087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4 lines destroy memheap mutex properly before memheap deleted (fix memory leak occured after r304950 changes with DEBUG_THREAD compile option) ........ 2011-10-03 18:44 +0000 [r339088] Terry Wilson * /, channels/chan_sip.c, main/file.c: Merged revisions 339086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) | 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite happens. If we receive a re-invite from a device the waitstream_core was not aware of the new control frame and would drop the call. (closes issue ASTERISK-18610) Reported by: Kristijan_Vrban ........ 2011-10-03 15:54 +0000 [r339011-339045] Matthew Nicholson * res/res_fax.c: Ported ast_fax_caps_to_str() to 10, not sure why it wasn't already here. This function prints a list of caps instead of a hex bitfield. * res/res_fax.c: Don't clear the AST_FAX_TECH_MULTI_DOC flag right after we set it. * res/res_fax.c: properly remove the AST_FAX_TECH_GATEWAY flag (instead of setting all of the other flags) 2011-10-03 14:38 +0000 [r338904-338997] Gregory Nietsky * CHANGES: Documentation noting the extension of CHANNEL() for chan_ooh323 * addons/chan_ooh323.c, funcs/func_channel.c: Remove the channel function OOH323() and place its options into CHANNEL() channel drivers should not have there own dialplan functions. * res/res_fax.c: Fixup a race condition in res_fax.c where FAXOPT(gateway)=no will turn off the gateway but the framehook is not destroyed. this problem happens when a gateway is attempted in the dialplan and the device is not available i may want to do fax to mail in the server it will not be allowed. instead of checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts 338904 Fix some white space. * res/res_fax.c: Remove T38 Gateway capability when detaching framehook. SET(FAXOPT(gateway)=no) does not remove the capability when detaching the framehook. small patch to fix this problem. 2011-09-30 22:06 +0000 [r338801] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 338800 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) | 12 lines Fix segfault in analog_ss_thread() not checking ast_read() for NULL. NOTE: The problem was reported against v1.6.2. It is unlikely to ever happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The version in sig_analog.c has largely replaced it. (closes issue ASTERISK-18648) Reported by: Stephan Bosch Patches: jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Stephan Bosch ........ 2011-09-30 18:55 +0000 [r338719] Jonathan Rose * /, configs/queues.conf.sample: Merged revisions 338718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep 2011) | 1 line Adds documentation for QueueMemberStatus event generation ........ 2011-09-30 16:35 +0000 [r338664] Richard Mudgett * /, channels/chan_sip.c: Fix formatting of AMI header for SIP show peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes issue ASTERISK-18649) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny ........ Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 2011-09-29 21:14 +0000 [r338556] Paul Belanger * tests/test_amihooks.c, tests/test_security_events.c, /, tests/test_locale.c, tests/test_logger.c, tests/test_dlinklists.c, tests/test_linkedlists.c: Merged revisions 338555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu, 29 Sep 2011) | 2 lines Test modules should depend on the TEST_FRAMEWORK flag ........ 2011-09-29 20:54 +0000 [r338552] Jason Parker * /, tests/test_db.c, tests/test_netsock2.c: Merged revisions 338551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep 2011) | 1 line Test modules have a support level of core. ........ 2011-09-29 18:32 +0000 [r338493] Leif Madsen * /, channels/chan_sip.c: Merged revisions 338492 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338492 | lmadsen | 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) | 6 lines Update documentation for SIP_HEADER. The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated in trunk, but not in 1.8 or 10, so I'm making them match. (Closes issue ASTERISK-18640) ........ 2011-09-29 12:16 +0000 [r338417] Gregory Nietsky * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 338416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | 12 lines The rtptimeout setting is ignored on a per peer basis. Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 ........ 2011-09-28 22:36 +0000 [r338253-338323] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 338322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) | 5 lines Make duplicate call ptr warning message more helpful. * Adds the value of the call ptr to the duplicate call ptr message to help trace why there is a duplicate call ptr. ........ * include/asterisk/logger.h, /: Merged revisions 338235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011) | 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch (license #6278) patch uploaded by Luke H ........ 2011-09-28 20:54 +0000 [r338228] Jason Parker * build_tools/cflags.xml, channels/chan_usbradio.c, build_tools/cflags-devmode.xml, agi/agi.xml, utils/utils.xml, /, build_tools/embed_modules.xml, tests/test_db.c, tests/test_netsock2.c: Merged revisions 338227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) | 1 line Add support levels to non-module sections of menuselect (cflags, utils, etc). ........ 2011-09-28 20:26 +0000 [r338225] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 338224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) | 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present. (closes issue ASTERISK-18357) Reported by: Matthew Nicholson ........ 2011-09-27 20:13 +0000 [r338085] Paul Belanger * /, apps/app_macro.c: Merged revisions 338084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue, 27 Sep 2011) | 2 lines Upgrade app_macro to core ........ 2011-09-26 19:35 +0000 [r337974] Richard Mudgett * cdr/cdr_manager.c, cdr/cdr_custom.c, apps/app_voicemail.c, apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /, include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c, include/asterisk/channel.h, main/cel.c, main/manager.c, funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c, main/logger.c, cel/cel_sqlite3_custom.c: Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ 2011-09-27 Asterisk Development Team * Asterisk 10.0.0-beta2 Released. * Based on revision that passed automated testing (http://bamboo.asterisk.org/browse/AST10-LUCID-178) 2011-09-26 19:35 +0000 [r337974] Richard Mudgett * cdr/cdr_manager.c, cdr/cdr_custom.c, apps/app_voicemail.c, apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /, include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c, include/asterisk/channel.h, main/cel.c, main/manager.c, funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c, main/logger.c, cel/cel_sqlite3_custom.c: Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ 2011-09-23 19:18 +0000 [r337840-337902] Gregory Nietsky * /, contrib/init.d/rc.archlinux.asterisk: Merged revisions 337898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) | 4 lines Spelling fix ........ * /, apps/app_queue.c: Merged revisions 337839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) | 11 lines Make sure a CDR is on the stack for call in the Queue. Only let update_cdr act on the last CDR in the stack. In some circumstances [Attended transfer to queue] a CDR record is not inserted for this call where it should. (closes issue ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266 ........ 2011-09-23 00:45 +0000 [r337775] Russell Bryant * configs/res_pktccops.conf.sample, /: Merged revisions 337774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011) | 11 lines Comment out entries in sample res_pktccops.conf. With these options enabled, they can cause Asterisk to freak out by SYN flooding a network and eating the CPU. Obviously it would be good to fix the code so that this can't happen, but we can at least change the default configuration so it doesn't happen. This was reported downstream to the Fedora issue tracker: https://bugzilla.redhat.com/show_bug.cgi?id=658431 ........ 2011-09-22 21:37 +0000 [r337721] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 337720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) | 18 lines Made ISDN not add numbering plan prefix strings to empty numbers. 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 Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw ........ 2011-09-22 18:43 +0000 [r337640] Paul Belanger * CREDITS, apps/app_meetme.c, CHANGES: Revert previous commit New feature should be added into trunk, unfortunately it is too late for the Asterisk 10 branch. 2011-09-22 15:47 +0000 [r337595-337597] Jonathan Rose * channels/sip/security_events.c (added), channels/sip/include/security_events.h (added): Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ * configs/logger.conf.sample, channels/chan_sip.c, include/asterisk/event_defs.h, main/security_events.c, main/event.c, CHANGES, channels/sip/include/sip.h, include/asterisk/security_events_defs.h: Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ 2011-09-22 11:44 +0000 [r337431-337542] Gregory Nietsky * res/res_srtp.c, /: Merged revisions 337541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) | 8 lines Add warned to ast_srtp to prevent errors on each frame from libsrtp The first 9 frames are not reported as some devices dont use srtp from first frame these are suppresed. the warning is then output only once every 100 frames. ........ * /, channels/chan_h323.c: Merged revisions 337486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | 10 lines If IP address is used in chan_h323 host parameter of peer configuration. module tries to resolve IP address to IP address and fails. Simple fix to set family of socket this is a hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500) ........ * apps/app_originate.c, CHANGES: Revert commit r337261 This commit is for trunk not version 10 ----- Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. ----- * main/channel.c, /: Merged revisions 337430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) | 19 lines Its possible to loose audio on ast_write when the channel is not transcoded correctly. in the case of DAHDI the channel is hungup. This patch tries to "fix" the problem and make the channel compatiable and warn the user of this problem. Please note there is a underlying problem with codec negotion this does not fix the problem it does try to rectify it and prevent loss of service. Review: https://reviewboard.asterisk.org/r/1442/ (closes issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325) (issue ASTERISK-18422) ........ 2011-09-21 21:25 +0000 [r337342-337380] Tilghman Lesher * apps/app_voicemail.c, /: More silly spacing changes ..... Merged revisions 337353 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * apps/app_voicemail.c, /: ........ Dumb little spacing fix. ........ Merged revisions 337344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * funcs/func_curl.c, /: ........ Escape commas in keys and values, when keys and values are enumerated by commas. Review: https://reviewboard.asterisk.org/r/1433 ........ Merged revisions 337325 from https://origsvn.digium.com/svn/asterisk/branches/1.8 2011-09-21 11:15 +0000 [r337261-337263] Gregory Nietsky * configs/sip.conf.sample: Whitespace fixup from SRTP patch * apps/app_originate.c, CHANGES: Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. Contributed by: jacco (thank you for the work) Review: https://reviewboard.asterisk.org/r/1310/ 2011-09-21 09:32 +0000 [r337178-337219] Olle Johansson * configs/extensions.conf.sample, main/pbx.c, CHANGES: Make ast_pbx_run() not default to s@default if extension is not found Review: https://reviewboard.asterisk.org/r/1446/ This is a bug - or architecture mistake - that has been in Asterisk for a very long time. It was exposed by the AMI originate action and possibly some other applications. Most channel drivers checks if an extension exists BEFORE starting a pbx on an inbound call, so most calls will not depend on this issue. Thanks everyone involved in the review and on IRC and the mailing list for a quick review and all the feedback. (closes issue ASTERISK-18578) * res/res_rtp_asterisk.c, configs/rtp.conf.sample, CHANGES: Change strictrtp option to default to yes in the RTP module Suggested by Kapejod on Facebook Review: https://reviewboard.asterisk.org/r/1448/ (closes issue ASTERISK-18587) Thanks for quick feedback to kpfleming and Tilghman --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M res/res_rtp_asterisk.c 2011-09-20 22:49 +0000 [r337120] Matthew Jordan * apps/app_voicemail.c, apps/app_dial.c, include/asterisk/app.h, /, apps/app_meetme.c, apps/app_minivm.c, main/app.c, apps/app_confbridge.c, apps/app_followme.c: Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ 2011-09-20 22:47 +0000 [r337119] Richard Mudgett * funcs/func_strings.c: Fix crash with STRREPLACE function. The ast_func_read() function calls the .read2 callback with the len parameter set to zero indicating no size restrictions on the supplied ast_str buffer. The value was used to dimension a local starts[] array with the array subsequently used. * Reworked the strreplace() function to perform the string replacement in a straight forward manner. Eliminated the need for the starts[] array. (closes issue ASTERISK-18545) Reported by: Federico Alves Patches: jira_asterisk_18545_v10.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Federico Alves 2011-09-20 22:19 +0000 [r337116] Leif Madsen * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 337115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011) | 7 lines Update RedHat Init script to work with Heartbeat. The current RedHat init script was not LSB compatible. This change will make it LSB compatible so that it can work correctly with Heartbeat. (Closes issue ASTERISK-18253) Reported by: c0rnoTa ........ 2011-09-20 21:05 +0000 [r337062] Kinsey Moore * tests/test_pbx.c, main/pbx.c, /: Merged revisions 337061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) | 11 lines Make CANMATCH with the new pattern match engine behave more like the old one When checking an extension for E_CANMATCH using the new extension matching algorithm, an exact match was not returned as a possible match resulting in the queue failing to allow a caller to exit on DTMF. This removes the requirement that an extension be longer than acquired digits for an E_CANMATCH operation to succeed. (closes issue ASTERISK-18044) Review: https://reviewboard.asterisk.org/r/1367/ ........ 2011-09-20 19:12 +0000 [r336978-337008] Richard Mudgett * /, channels/sig_ss7.c: Merged revisions 337007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) | 15 lines Check if a channel was created before using the pointer in sig_ss7_new_ast_channel(). Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing libss7 access lock protection. * Prevent cancelling the ss7_linkset() thread at inoportune times just like the pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett (attached to related ASTERISK-17966) ........ * /, channels/sig_ss7.c: Merged revisions 336977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) | 21 lines Fix deadlock from not releasing SS7 linkset lock. sig_ss7_hangup() failed to release the SS7 linkset lock if the call had the alreadyhungup flag set. * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the alreadyhungup flag is set. * Made ss7_start_call() not hold any locks while creating the channel for an incoming call to prevent deadlock. * Made ss7_grab() a void function, since it could never fail, to simplify calling code. * Made obtain the channel lock to do softhangup in some places. Patches: jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett JIRA AST-668 ........ 2011-09-20 16:51 +0000 [r336936] Gregory Nietsky * channels/sip/sdp_crypto.c, channels/chan_sip.c, channels/sip/include/sdp_crypto.h, channels/sip/include/srtp.h, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Allow Setting Auth Tag Bit length Based on invite or config option Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ 2011-09-20 01:03 +0000 [r336878] Russell Bryant * res/res_rtp_asterisk.c, /: Merged revisions 336877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) | 36 lines Fix crashes in ast_rtcp_write(). This patch addresses crashes related to RTCP handling. The backtraces just show a crash in ast_rtcp_write() where it appears that the RTP instance is no longer valid. There is a race condition with scheduled RTCP transmissions and the destruction of the RTP instance. This patch utilizes the fact that ast_rtp_instance is a reference counted object and ensures that it will not get destroyed while a reference is still around due to scheduled RTCP transmissions. RTCP transmissions are scheduled and executed from the chan_sip scheduler context. This scheduler context is processed in the SIP monitor thread. The destruction of an RTP instance occurs when the associated sip_pvt gets destroyed (which happens when the sip_pvt reference count reaches 0). However, the SIP monitor thread is not the only thread that can cause a sip_pvt to get destroyed. The sip_hangup function, executed from a channel thread, also decrements the reference count on a sip_pvt and could cause it to get destroyed. While this is being changed anyway, the patch also removes calling ast_sched_del() from within the RTCP scheduler callback. It's not helpful. Simply returning 0 prevents the callback from being rescheduled. (closes issue ASTERISK-18570) Related issues that look like they are the same problem: (issue ASTERISK-17560) (issue ASTERISK-15406) (issue ASTERISK-15257) (issue ASTERISK-13334) (issue ASTERISK-9977) (issue ASTERISK-9716) Review: https://reviewboard.asterisk.org/r/1444/ ........ 2011-09-19 22:13 +0000 [r336792] Terry Wilson * /, channels/chan_sip.c: Merged revisions 336791 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 ........ 2011-09-19 21:41 +0000 [r336734-336789] Tilghman Lesher * funcs/func_strings.c: Ensure substring will not be found in the previous match. * include/asterisk/optional_api.h, Makefile, /, configure, include/asterisk/autoconfig.h.in, main/Makefile, codecs/gsm/Makefile, configure.ac, Makefile.rules: Merged revisions 336733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) | 11 lines Various changes to allow 1.8 to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6 extended to work on 10.7 and later. * Now uses the 'weak' symbol for Lion systems, which no longer support 'weak_import' Closes ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej. ........ 2011-09-19 20:16 +0000 [r336717] Jonathan Rose * /, apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c, apps/app_morsecode.c, res/res_musiconhold.c, apps/app_queue.c, apps/app_mixmonitor.c: Merged revisions 336716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines Document applications that play audio and do not answer unanswered calls. This patch is part of an effort to document early media and its usage. If you are interested in contributing to this documentation effort, there are probably other applications worth documenting as well as an Asterisk wiki article at https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application ........ 2011-09-19 18:51 +0000 [r336659] Richard Mudgett * apps/app_dial.c, /, UPGRADE-1.8.txt: Merged revisions 336658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines Made Dial d and H options no longer immediately auto-answer the calling leg. The Dial d and H options break DTMF attended transfer atxferdropcall option. 1) Party A calls party B. 2) Party B does a DTMF attended transfer to Party C. If the dialplan uses the Dial d or H options to call Party C then the Dial application answers the call immediately before initiating the call leg to Party C. The premature answer causes the transfer code to not invoke the atxferdropcall=no behavior for a blonde transfer since Party C has "answered". The transfer code thinks that Party B has "consulted" with Party C when Party B hangs up and completes the transfer to Party A. Party A now hears ringback until Party C actually answers. ASTERISK-13294 Dial d option. ASTERISK-11067 Dial H option to disconnect before answer. The referenced issues made Dial answer with the d and H options because many SIP and ISDN phones cannot send DTMF before the call is connected. * Made require the dialplan to control when or if the call needs to be answered to use the Dial application d and H options. (The call is no longer surprise answered when using the Dial d or H options.) Review: https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA AST-666 ........ 2011-09-19 15:42 +0000 [r336573] Leif Madsen * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 336572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7 lines Update get_ilbc_source.sh script to work again. Recently iLBC support in Asterisk has changed after the acquisition of GIPS by Google. More information about how this may affect you is available in a blog post at: http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/ ........ 2011-09-19 15:32 +0000 [r336570] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 336569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA AST-675 ........ 2011-09-19 13:48 +0000 [r336502-336504] Olle Johansson * Makefile: Revert accidental change * Makefile, /, channels/chan_sip.c: Merged revisions 336501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 lines Add diversion header to a 302 redirect response if we have diversion data (closes issue ASTERISK-18143) patch by oej ........ 2011-09-19 13:31 +0000 [r336500] Gregory Nietsky * /, channels/chan_h323.c: Merged revisions 336499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines A long time ago in a galaxy far far away a IPv6 update was made, chan_h323 was not updated causeing all to flee to chan_ooh323. the brave Jedi [asterisk developers] pondered this miscarrige of justice and restored order to the force for the sake of closing out 2 old issues. (closes issue ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread, sybasesql Tested by: irroot Reviewed by: IRC (russellb, kpfleming) ........ 2011-09-19 12:15 +0000 [r336381-336441] Olle Johansson * main/manager.c, /: Merged revisions 336440 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2 lines Make sure manager_debug option is reset at reload ........ * /, channels/chan_sip.c: Merged revisions 336378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 lines Add missing unlock at MWI message sending time (closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! ........ 2011-09-16 22:11 +0000 [r336313-336316] Terry Wilson * /, funcs/func_frame_trace.c: Merged revisions 336314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16 Sep 2011) | 2 lines Whitespace fix ........ * /, funcs/func_frame_trace.c: Merged revisions 336312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011) | 5 lines Add missing frame types to func_frame_trace Also casts control frames to the proper enum so that the compile will catch new additions. ........ 2011-09-16 21:09 +0000 [r336307] Jonathan Rose * main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/frame.h: Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ 2011-09-16 19:10 +0000 [r336235] Sean Bright * /, UPGRADE-1.8.txt: Merged revisions 336234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep 2011) | 2 lines Make a note that inotify won't work with an NFS mounted spooler directory. ........ 2011-09-16 10:12 +0000 [r336094-336167] Gregory Nietsky * channels/chan_misdn.c, /: Merged revisions 336166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | 16 lines The round robin routing routine in chan_misdn.c is broken. it rotates between ports but never checks the channels in the ports. i have extensivly tested it and verified it works on 1 upto 4 ports. before the patch only 1 out of each port was used now all are used as expected. (closes issue ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed by: irroot Review: https://reviewboard.asterisk.org/r/1410/ ........ * /, apps/app_queue.c: Merged revisions 336093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | 20 lines Locking order in app_queue.c causes deadlocks. a channel lock must never be held with the queues container lock held. the deadlock occured on masquerade. the queues container lock is a relic of the past the old queue module lock. with ao2 there is no need to hold this lock when dealing with members this patch removes unneeded locks. (closes issue ASTERISK-18101) (closes issue ASTERISK-18487) Reported by: Paul Rolfe, Jason Legault Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by: Matthew Nicholson Review: https://reviewboard.asterisk.org/r/1402/ ........ 2011-09-15 15:19 +0000 [r336091] David Vossel * main/format_cap.c: Removes some no-op code found in format_cap.c. 2011-09-15 12:46 +0000 [r336042] Olle Johansson * CREDITS, apps/app_meetme.c, CHANGES: Meetme: Introducing a new option "k" to kill a conference if there's only a single member left. When using Meetme as a modular call bridge from third party applications, it's handy to make it behave like a normal call bridge. When the second to last person exists, the last person will be kicked out of the conference when this option is enabled. (closes issue ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored by ClearIT, Solna, Sweden 2011-09-15 08:29 +0000 [r335991] Gregory Nietsky * /, channels/chan_agent.c: Merged revisions 335978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | 11 lines lock the channel before calling ast_bridged_channel() to prevent a seg fault. AMI agents list called on shutdown causes a segfault, introducing proper locking will prevent this. (closes issue ASTERISK-18092) Reported by: agustina Patches: chan_agent.patch (License #5041) patch uploaded by irroot ........ 2011-09-14 18:31 +0000 [r335852-335912] Richard Mudgett * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 335911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines Remove unnecessary libpri dependency checks in the configure script. Using the --with-pri option with the configure script generated an error about not having PRI_L2_PERSISTENCE if you did not have the absolute latest libpri SVN checkout installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to be for libraries that are dependent upon other libraries and not necessarily for optional/added features within a library. (closes issue ASTERISK-18535) Reported by: Michael Keuter ........ * channels/chan_dahdi.c, /: Merged revisions 335851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) | 11 lines Fixed cut-n-paste regression using the wrong variable. Fixes the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration. (closes issue ASTERISK-18496) Reported by: Sean Darcy Patches: jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Sean Darcy, rmudgett ........ 2011-09-14 13:28 +0000 [r335791] Matthew Nicholson * main/manager.c, /: Merged revisions 335790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep 2011) | 4 lines The tech and data members of fast_originate_helper are not string fields. ASTERISK-17709 ........ 2011-09-13 22:10 +0000 [r335721] Richard Mudgett * /, apps/app_directed_pickup.c: Merged revisions 335720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011) | 1 line Remove obsolete todo comment about PICKUPRESULT. ........ 2011-09-13 21:37 +0000 [r335717] Tzafrir Cohen * main/asterisk.c: do parse defaultlanguage from asterisk.conf Do parse the option "defaultlanguage" from the [options] section of asterisk.conf, as in the sample config file. Otherwise the build-time default language (normally "en") is always the default one. Review: https://reviewboard.asterisk.org/r/1342/ Signed-off-by: Tzafrir Cohen (License #5035) Original-Commit: http://svn.digium.com/svn/asterisk/branches/1.8@335716 2011-09-13 18:55 +0000 [r335656] Tilghman Lesher * /, configure, configure.ac: Merged revisions 335655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011) | 4 lines Move mandatory checks closer to the beginning of the file. If these are going to fail, they should fail as quickly as possible. ........ 2011-09-13 18:47 +0000 [r335653] Matthew Nicholson * main/pbx.c, main/manager.c, /: Merged revisions 335618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep 2011) | 5 lines Don't limit the size of appdata for manager originate actions. ASTERISK-17709 Patch by: tilghman (with modifications) ........ 2011-09-13 07:24 +0000 [r335510] Russell Bryant * include/asterisk/event.h, /, res/ais/evt.c, main/event.c: Merged revisions 335497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines Fix a crash in res_ais. This patch resolves a crash observed in a load testing environment that involved the use of the res_ais module. I observed some crashes where the event delivery callback would get called, but the length parameter incidcating how much data there was to read was 0. The code assumed (with good reason I would think) that if this callback got called, there was an event available to read. However, if the rare case that there's nothing there, catch it and return instead of blowing up. More specifically, the change always ensure that the size of the received event in the cluster is always big enough to be a real ast_event. Review: https://reviewboard.asterisk.org/r/1423/ ........ 2011-09-12 15:55 +0000 [r335434] Matthew Nicholson * main/channel.c, /: Merged revisions 335433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep 2011) | 6 lines Properly set caller_warning and callee_warning before we try to use them. ASTERISK-18199 Patch by: elguero Testing by: rtang ........ 2011-09-12 14:22 +0000 [r335346] Kinsey Moore * apps/app_dial.c, /: Merged revisions 335341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines Ensure frames are not written to dialed channel if ringback is requested When a single channel was dialed and there was media to be forwarded to the calling channel, the media was written without regard for ringback causing silence to be heard in some circumstances. This regression was introduced when the meaning of "single" changed to mean only the number of channels dialed. (closes issue ASTERISK-18083) ........ 2011-09-12 13:47 +0000 [r335323] Olle Johansson * /, channels/chan_sip.c: Merged revisions 335319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines Lock the peer->mvipvt to avoid crashes with SIP history enabled After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt, which cause issues with SIP history additions in combination with the max limit for number of history entries. Review: https://reviewboard.asterisk.org/r/1373/ (closes issue ASTERISK-18288) Thanks to irrot for peer review. Work with this bug funded by IPvision AS ........ 2011-09-12 13:27 +0000 [r335321] Kinsey Moore * /, channels/chan_iax2.c: Merged revisions 335320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6 addresses via DNS queries. (closes issue ASTERISK-18090) ........ 2011-09-12 11:11 +0000 [r335260] Stefan Schmidt * /, channels/chan_sip.c: Merged revisions 335259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value. adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ ........ 2011-09-09 16:27 +0000 [r335078] Matthew Jordan * channels/chan_mgcp.c, channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, addons/chan_ooh323.c, /, channels/chan_sip.c, channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c, main/channel.c, channels/chan_usbradio.c, main/dial.c, channels/chan_dahdi.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_frame_trace.c, main/features.c, channels/chan_h323.c, channels/chan_alsa.c, include/asterisk/frame.h, channels/sig_ss7.c: Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ 2011-09-09 07:23 +0000 [r335014] Gregory Nietsky * funcs/func_dialplan.c, apps/app_readexten.c, CHANGES: Move code for VALID_EXTEN from app_readexten to func_dialplan Mark VALID_EXTEN deprecated. Review: https://reviewboard.asterisk.org/r/1396/ 2011-09-08 22:28 +0000 [r334954] Richard Mudgett * /, main/logger.c: Merged revisions 334953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) | 10 lines Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to unregister its logger level. * Make ast_logger_unregister_level() use ast_free() instead of free(). When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call to free(). Therefore, if you allocated memory with a form of ast_malloc you must free it with ast_free. ........ 2011-09-07 19:37 +0000 [r334844] Paul Belanger * /, channels/chan_iax2.c: Merged revisions 334843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) | 4 lines Cleanup chan_iax2.c log messages Review: https://code.asterisk.org/code/cru/CR-AST-11 ........ 2011-09-07 19:33 +0000 [r334841] Richard Mudgett * /, main/features.c: Merged revisions 334840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines Fix AMI action Park crash. * Made AMI action Park not say anything to the parker channel (AMI header Channel2) since the AMI action is a third party parking the call. (This is a change in behavior that cannot be preserved without a lot of effort.) * Made not play pbx-parkingfailed if the Park 's' option is used. JIRA AST-660 ........ 2011-09-07 15:10 +0000 [r334682-334747] Stefan Schmidt * /, main/features.c: Merged revisions 334682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. ........ * main/features.c: another clean up * main/features.c: Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. 2011-09-07 08:14 +0000 [r334617-334621] Alec L Davis * /, CHANGES, apps/app_queue.c: Merged revisions 334620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) | 2 lines peroid typo ........ * main/pbx.c, /: Merged revisions 334616 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep 2011) | 10 lines Prevent segfault if call arrives before Asterisk is fully booted. Prevent ast_pbx_start and ast_run_start from starting a new thread unless asterisk is fully booted. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1407/ ........ 2011-09-06 15:47 +0000 [r334514] Paul Belanger * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: authdebug is now disabled by default To enable this functionaility again set authdebug = yes in iax.conf Review: https://reviewboard.asterisk.org/r/1414/ 2011-09-06 13:58 +0000 [r334455] Gregory Nietsky * apps/app_voicemail.c, /: Merged revisions 334453 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines Make SQL query in app_voicemail.c portable LIMIT is not portable. Regression from r312212 (closes issue ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1415/ ........ 2011-09-02 21:08 +0000 [r334297-334357] Richard Mudgett * /, res/res_musiconhold.c: Merged revisions 334355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) | 19 lines MusicOnHold has extra unref which may lead to memory corruption and crash. The problem happens when a call is disconnected and you had started a MOH class that does not use the files mode. If you define REF_DEBUG and recreate the problem, it will announce itself with the following warning: Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained, and class is still in a container! * Fixed moh_alloc() and moh_release() functions not handling the state->class reference consistently. (closes issue ASTERISK-18346) Reported by: Mark Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski Review: https://reviewboard.asterisk.org/r/1404/ ........ * /, include/asterisk/config.h, main/config.c: Merged revisions 334296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) | 39 lines Fix potential memory allocation failure crashes in config.c. * Added required checks to the returned memory allocation pointers to prevent crashes. * Made ast_include_rename() create a replacement ast_variable list node if the new filename is longer than the available space. Fixes potential crash and memory leak. * Factored out ast_variable_move() from ast_variable_update() so ast_include_rename() can also use it when creating a replacement ast_variable list node. * Made the filename stuffed at the end of the struct a minimum allocated size in ast_variable_new() in case ast_include_rename() changes the stored filename. * Constify struct char pointers pointing to strings stuffed at the end of the struct for: ast_variable, cache_file_mtime, and ast_config_map. * Factored out cfmtime_new() to remove inlined code and allow some struct pointers to become const. * Removed the list lock from struct cache_file_mtime that was never used. * Added doxygen comments to several structure elements and better documented what strings are stuffed at the struct end char array. * Reworked ast_config_text_file_save() and set_fn() to handle allocation failure of the include file scratch pad object tracking blank lines. * Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure it is long enough for any filename with path. Also reduced the number of container fileset buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review: https://reviewboard.asterisk.org/r/1378/ ........ 2011-09-01 17:39 +0000 [r334235] Tilghman Lesher * main/pbx.c, /: Merged revisions 334234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01 Sep 2011) | 2 lines Remove 1.6 compatibility documentation from 1.8, as it no longer applies. ........ 2011-09-01 17:36 +0000 [r334233] Matthew Nicholson * CHANGES: fixed a typo 2011-09-01 17:30 +0000 [r334230] Tilghman Lesher * res/res_config_odbc.c, /: Merged revisions 334229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011) | 18 lines Create a local alias for ast_odbc_clear_cache. As a function pointer, the reference has to be resolved at load time irrespective of the RTLD_LAZY flag. Creating a local alias solves this problem, because the structure is initialized with that local function pointer, while the actual function can remain lazily linked until runtime. The reason why this is important is because we lazily load function references during the module loading process, in order to obtain priority values for each module, ensuring that modules are loaded in the correct order. Previous to this change, when this module was initially loaded, the module loader would emit a symbol resolution error, because of the above requirement. Closes ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by Walter Doekes, patch by me) ........ 2011-08-31 18:53 +0000 [r334064-334157] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 334156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) | 4 lines Disable T.38 when we get a invite with image media port set to 0 ASTERISK-17678 ........ * res/res_fax.c: only alter the gateway_timeout when attching the gateway to a channel ASTERISK-18219 2011-08-31 16:00 +0000 [r334010-334013] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 334012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines No DAHDI channel available for conference, user introduction disabled. The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards Patches: jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ * main/channel.c, /, channels/chan_agent.c: Merged revisions 334009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines Call pickup race leaves orphaned channels or crashes. Multiple users attempting to pickup a call that has been forked to multiple extensions either crashes or fails a masquerade with a "bad things may happen" message. This is the scenario that is causing all the grief: 1) Pickup target is selected 2) target is marked as being picked up in ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app dial or queue gets a chance to hang up losing calls and calls ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with ast_channel_masquerade(), ast_hangup() completes successfully and the channel is no longer in the channels container. 6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the masquerade on the dead channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel 8) bad things happen while doing the masquerade and in the process ast_do_masquerade() puts the dead channel back into the channels container 9) The "orphaned" channel is visible in the channels list if a crash does not happen. This patch does the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel and not release the channel lock until that has happened. * Made __ast_channel_masquerade() not setup a masquerade if either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work. (closes issue ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer Review: https://reviewboard.asterisk.org/r/1400/ ........ 2011-08-31 15:19 +0000 [r334007] Kinsey Moore * /, channels/chan_sip.c: Merged revisions 334006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines Correct an AMI protocol violation with SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are ended by using \r\n this confuses any interfacing script. (closes issue ASTERISK-17486) ........ 2011-08-30 21:53 +0000 [r333961-333962] Alexandr Anikin * addons/ooh323c/src/ooh323.c: security fix. really drop call if signalling addr is not same as socket addr * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, /, addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/oochannels.c: Merged revisions 333947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines cleanups in ACF/ARJ GK replies processing fixed long (24 sec) pause if acf/arj proccessed before ast_cond_wait called to wait this ........ 2011-08-30 14:01 +0000 [r333895] Matthew Nicholson * res/res_fax.c: Replaced FAXOPT(gwtimeout) with a second parameter to FAXOPT(gateway). Patch by: irroot Review: https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219 2011-08-29 21:41 +0000 [r333837] Terry Wilson * /, channels/chan_sip.c: Merged revisions 333836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) | 15 lines Refresh peer address if DNS unavailable at peer creation If Asterisk starts and no DNS is available, outbound registrations will fail indefinitely. This patch copies the address from the sip_registry struct, which will be updated, to the peer->addr when necessary. If dnsmgr is enabled, the registration fails without the patch because even though the address on the registry is updated via dnsmgr, the address is just copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update the address that is copied to the sip_pvt or peers. Closes issue ASTERISK-18000 Review: https://reviewboard.asterisk.org/r/1335/ ........ 2011-08-29 21:12 +0000 [r333786] Richard Mudgett * /, include/asterisk/channel.h, addons/chan_mobile.c: Merged revisions 333784-333785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011) | 2 lines Fix deadlock potential of chan_mobile.c:mbl_ast_hangup(). ........ r333785 | rmudgett | 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line Add some do not hold locks notes to channel.h ........ 2011-08-29 18:22 +0000 [r333716] Matthew Nicholson * res/res_fax_spandsp.c: It is possible for the gateway to be attached when the channel is still negotiating T.38. This change handles that case. ASTERISK-18329 2011-08-29 17:28 +0000 [r333681] Terry Wilson * main/channel.c, CHANGES: Use realtime text when it is negotiated This patch make use of wirte_text() realtime text instead of send_text() if T.140 is in native formats. ASTERISK-17937 Review: https://reviewboard.asterisk.org/r/1356/ 2011-08-29 17:12 +0000 [r333631] Matthew Jordan * apps/app_voicemail.c, /: Merged revisions 333630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29 Aug 2011) | 1 line Fixed improperly formatted TestEvent AMI message in app_voicemail ........ 2011-08-29 15:56 +0000 [r333570] Jonathan Rose * /, res/res_jabber.c: Merged revisions 333569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) | 4 lines Accidental use of variable client->status instead of client->state in from ASTERISK-18078 (issue ASTERISK-18078) ........ 2011-08-28 09:55 +0000 [r333508] Tzafrir Cohen * channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6) GCC 4.6 detects variables that get assined to, but never used later. Also removes some remmed-out lines that become invalid. (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen (License #5035) , 2011-08-26 16:28 +0000 [r333410] Jonathan Rose * /, res/res_jabber.c: Merged revisions 333378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) | 13 lines [patch] Buddies are always auto-registered when processing the roster Reporter said autoregister flag was ignored for registering 'buddies' which had a subscription to us. Verified that this was the case and observed how the patch addressed this and made sure it didn't break anything. (closes issue ASTERISK-14233) Reported by: Simon Arlott Patches: asterisk-0015229.patch (license #5756) patch uploaded by Simon Arlott Tested by: Jonathan Rose ........ 2011-08-26 15:58 +0000 [r333370] Matthew Jordan * apps/app_voicemail.c, /: Merged revisions 333339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011) | 20 lines Bug fixes for voicemail user emailsubject / emailbody. This code change fixes a few issues with the voicemail user override of emailbody and emailsubject, including escaping the strings, potential memory leaks, and not overriding the voicemail defaults. Revision 325877 fixed this for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent check-in prevented 325877 from being applied to 10. This check-in resolves both issues, and applies the changes to 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by: Sebastien Couture Tested by: mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1374 ........ 2011-08-25 19:01 +0000 [r333268] Jason Parker * Makefile, /: Merged revisions 333267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug 2011) | 2 lines Fix for DESTDIR spaces patch. ........ 2011-08-25 19:00 +0000 [r333266] Jonathan Rose * /, res/res_jabber.c: Merged revisions 333265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) | 14 lines Segfault when publishing device states via XMPP and not connected When using publishing device state with res_jabber, Asterisk will attempt to send a device state using the unconnected client using iks_send_raw and crash. This patch checks the validity of the connection before attempting to send the device state. (closes issue ASTERISK-18078) Reported by: Michael L. Young Patches: res_jabber-segfault-pubsub-not-connected2.patch (license #5026) patch uploaded by Michael L. Young Tested by: Jonathan Rose ........ 2011-08-25 15:29 +0000 [r333203] Jason Parker * Makefile, build_tools/mkpkgconfig, /, configure, configure.ac, makeopts.in, sounds/Makefile: Merged revisions 333201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines Fix installation into directories containing spaces. This also vastly simplifies the logic in sounds/Makefile (Closes issue ASTERISK-18290) Reported by: Paul Belanger Review: https://reviewboard.asterisk.org/r/1379/ ........ 2011-08-24 16:51 +0000 [r333115] Matthew Nicholson * res/res_fax.c: Changed the "timeout" option to "gwtimeout". ASTERISK-18219 2011-08-23 18:15 +0000 [r332878-333011] Richard Mudgett * /, apps/app_queue.c: Merged revisions 333010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011) | 12 lines Memory Leak in app_queue The patch that was committed in the 1.6.x versions of Asterisk for ASTERISK-15862 actually fixed two issues. One was not applicable to 1.8 but the other is. queue_leak.patch fixes the portion applicable to 1.8. (closes issue ASTERISK-18265) Reported by: Fred Schroeder Patches: queue_leak.patch (license #5049) patch uploaded by mmichelson Tested by: Thomas Arimont ........ * /, main/config.c: Merged revisions 332939 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011) | 7 lines Minor code optimizations. * Simplify ast_category_browse() logic for easier understanding. * Remove dead code in ast_variable_delete() and simplify some of its logic. ........ * /, apps/app_queue.c: Merged revisions 332874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011) | 18 lines Reference leaks in app_queue. * Fixed load_realtime_queue() leaking a queue reference when it overwrites q when processing a realtime queue. (issue ASTERISK-18265) * Make join_queue() unreference the queue returned by load_realtime_queue() when it is done with the pointer. The load_realtime_queue() returns a reference to the just loaded realtime queue. * Fixed queues container reference leak in queues_data_provider_get(). * queue_unref() should not return q that was just unreferenced. * Made logic in __queues_show() and queues_data_provider_get() when calling load_realtime_queue() easier to understand. ........ 2011-08-22 19:43 +0000 [r332877] Paul Belanger * /, channels/chan_gtalk.c: Merged revisions 332876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug 2011) | 6 lines Revert previous commit It seems google is still making changes to the protocol. (issue ASTERISK-18301) ........ 2011-08-22 19:41 +0000 [r332875] Richard Mudgett * /: Fix merge property. 2011-08-22 18:40 +0000 [r332832] Matthew Jordan * apps/app_voicemail.c, include/asterisk/test.h, main/manager.c, /, main/file.c, main/test.c, main/app.c, configs/manager.conf.sample, include/asterisk/manager.h: Merged revisions 332817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) | 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. ........ 2011-08-22 18:32 +0000 [r332761-332830] Richard Mudgett * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged revisions 332816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011) | 8 lines Memory leaks in realtime_multi_xxx() when database access returns error. * Fix realtime_multi_pgsql() configuration memory leak when the database access returns an error. * Fix realtime_multi_odbc() configuration category use after free when the database access returns an error. ........ * /, main/config.c: Merged revisions 332759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011) | 15 lines Memory leak reading realtime database variable list. Calling ast_load_realtime() can leak the last list node if the read list only contains empty variable value items. * Fixed list filter loop in ast_load_realtime() to delete the list node immediately instead of the next time through the loop. The next time through the loop may not happen if the node to delete is the last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265) Patches: jira_asterisk_18265_v1.8_config.patch (license #5621) patch uploaded by rmudgett ........ 2011-08-22 16:29 +0000 [r332756] Matthew Nicholson * res/res_fax.c, include/asterisk/res_fax.h: add a way to disable and/or modify the gateway timeout ASTERISK-18219 2011-08-21 14:33 +0000 [r332700] Paul Belanger * /, channels/chan_gtalk.c: Merged revisions 332699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug 2011) | 5 lines Fix outgoing calls in chan_gtalk (closes issue ASTERISK-18301) Reported by: az1324 ........ 2011-08-19 19:59 +0000 [r332654] Kinsey Moore * apps/app_confbridge.c: Make CONFBRIDGE_INFO behave more nicely CONFBRIDGE_INFO doesn't behave as well in edge cases as MEETME_INFO. With this patch, CONFBRIDGE_INFO should behave in a much more reasonable manner when presented with invalid conferences and keywords. Review: https://reviewboard.asterisk.org/r/1359/ 2011-08-18 21:34 +0000 [r332560] Terry Wilson * main/netsock2.c, /: Merged revisions 332559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 5 lines Fix possible error on stringification of IPv4-mapped addrs The FreeBSD netsock2 test has been failing for a while. We were pasing sa->len to getnameinfo instead of sa_tmp->len. ASTERISK-18289 ........ 2011-08-18 19:29 +0000 [r332504] Kinsey Moore * channels/chan_dahdi.c, /: Merged revisions 332503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) | 8 lines CRC4 in "dahdi show status" gives wrong impression to T1 users Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in more situations without confusing users, especially since T1 lines use CRC6 instead of CRC4. (closes issue AST-471) ........ 2011-08-18 14:48 +0000 [r332369-332447] Tilghman Lesher * build_tools/cflags.xml, build_tools/cflags-devmode.xml, /: Merged revisions 332446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18 Aug 2011) | 2 lines Move BETTER_BACKTRACES out of development mode, as it's useful when DEBUG_THREADS is enabled. ........ * Makefile, agi/Makefile, utils/Makefile, /, configure, include/asterisk/autoconfig.h.in, configure.ac, Makefile.moddir_rules, makeopts.in, sounds/Makefile: Merged revisions 332355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 10 lines Re-add support for spaces in pathnames, including now spaces in DESTDIR. This was initially added to 1.8 prior to release, primarily to support the standard paths on Mac OS X, but was partially reverted recently in Subversion, due to the lack of support for spaces in DESTDIR. This commit restores support for the standard paths on Mac OS X, and also includes support for spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by: pabelanger Review: https://reviewboard.asterisk.org/r/1326/ ........ 2011-08-17 18:09 +0000 [r332321] Terry Wilson * /, res/res_timing_timerfd.c: Merged revisions 332320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011) | 10 lines Don't read from a disarmed or invalid timerfd Numerous isues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. This patch adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18166, ASTERISK-18197, AST-486, AST-495, AST-507 and possibly others. Review: https://reviewboard.asterisk.org/r/1361/ ........ 2011-08-17 16:01 +0000 [r332265] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c: Merged revisions 332264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) | 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 ........ 2011-08-16 20:11 +0000 [r332177] Paul Belanger * tests/test_amihooks.c, tests/test_substitution.c, tests/test_heap.c, /, tests/test_expr.c, tests/test_ast_format_str_reduce.c, tests/test_logger.c, tests/test_gosub.c, tests/test_app.c, tests/test_dlinklists.c, tests/test_event.c, tests/test_db.c, tests/test_linkedlists.c, tests/test_sched.c, tests/test_netsock2.c, tests/test_pbx.c, tests/test_strings.c, tests/test_func_file.c, tests/test_security_events.c, tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c, tests/test_acl.c, tests/test_locale.c, tests/test_utils.c, tests/test_devicestate.c, tests/test_aoc.c, tests/test_astobj2.c, tests/test_poll.c: Merged revisions 332176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug 2011) | 4 lines Flag test modules as 'core' Review: https://reviewboard.asterisk.org/r/1369/ ........ 2011-08-16 17:45 +0000 [r332119] Jonathan Rose * /, channels/chan_sip.c: Merged revisions 332118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) | 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox setting in sip.conf would only result in updates being sent on whichever mailbox triggered the mwi event. Now all of them get counted regardless. Also fixes a bug involving parsing of the mailbox option in sip.conf so that trailing and leading spaces before/after commas are trimmed. (closes issue ASTERISK-18067) Reported by: aragon (closes issue ASTERISK-15479) Reported by: Ben Winslow Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow ........ 2011-08-16 17:17 +0000 [r332101] Richard Mudgett * /, main/features.c, CHANGES, configs/features.conf.sample, main/asterisk.c: Merged revisions 332100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ ........ 2011-08-16 15:20 +0000 [r332022-332042] Matthew Nicholson * channels/sip/include/sip.h: fix a code comment AST-580 * UPGRADE.txt, CHANGES: Moved notes about 'storesipcause' to UPGRADE.txt from CHANGES AST-580 * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 332026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option AST-580 ........ * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: In 10 and trunk this option is disabled by default. Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ 2011-08-15 17:35 +0000 [r331956] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 331955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) | 13 lines Fix some minor chan_dahdi config load issues. * Address chan_dahdi.conf dahdichan option todo item about needing line number. * Make ignore_failed_channels option also apply to dahdichan option. * Don't attempt to create a default pseudo channel if the chan_dahdi.conf channel/channels option is not allowed. * Add a similar check for dahdichan in normal chan_dahdi.conf sections as is done in users.conf. ........ 2011-08-15 15:22 +0000 [r331894] Paul Belanger * main/rtp_engine.c, /: Merged revisions 331886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug 2011) | 5 lines Fix noisy message when briding channels (closes issue ASTERISK-18270) Reported by: Federico Alves ........ 2011-08-15 15:14 +0000 [r331868] David Vossel * /, channels/chan_sip.c: Merged revisions 331867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) | 6 lines Fixes locking inversion issues present in the handling of the sip REFER method. (closes issue ASTERISK-18082) Reported by: James Van Vleet ........ 2011-08-12 19:03 +0000 [r331775] Matthew Nicholson * /, apps/app_queue.c: Merged revisions 331774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug 2011) | 11 lines Unlock the channel before calling update_queue. Holding the channel lock when calling update_queue which attempts to lock the queue lock can cause a deadlock. This deadlock involves the following chain: 1. hold chan lock -> wait queue lock 2. hold queue lock -> wait agent list lock 3. hold agent list lock -> wait chan list lock 4. hold chan list lock -> wait chan lock ........ 2011-08-12 18:59 +0000 [r331715-331772] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 331771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) | 8 lines Suppress warning message when using DAHDITransfer or DAHDIHangup. * The fake event should only be processed by the channel that currently owns the private and not the associated call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 ........ * channels/chan_dahdi.c, /: Merged revisions 331714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) | 22 lines AMI actions DAHDIHangup and DAHDITransfer have no effect. The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI channel. These two AMI actions are highly specialized to analog channels and appear to make the channel behave like a jack port for headsets. * Made the faked DAHDI event get processed before a normal media stream read in dahdi_read() instead of trying to trigger an exception read by setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long ago that changed how AST_FLAG_EXCEPTION is processed in the core. Unfortunately, the faked DAHDI events no longer worked when that happened. * Updated the DAHDI AMI action documentation for the following actions: DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and DAHDIRestart. * Made use sscanf() instead of atoi() for better error checking of the DAHDIChannel header string. JIRA AST-620 JIRA SWP-3616 ........ 2011-08-12 16:31 +0000 [r331659] Terry Wilson * /, tests/test_netsock2.c: Merged revisions 331658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12 Aug 2011) | 4 lines Fix netsock2 multiple zero-expansion test Remove erroneous single bracket. ........ 2011-08-12 16:21 +0000 [r331654] Kinsey Moore * /, main/logger.c: Merged revisions 331649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) | 12 lines Logger does not warn of failure to open logging channels Currently, logger only prints an error message to stderr when it fails to open a logger channel where many users will not see it because the logger lock is held. The alternative provided by this patch is to log the error to all attached consoles in the hopes that it will be easier to see. Additionally, this patch prevents the failed logger channel from being added to the list where it would silently fail on each call to the Asterisk logger. (closes issue ASTERISK-16231) Review: https://reviewboard.asterisk.org/r/1338 ........ 2011-08-12 16:18 +0000 [r331644] Jonathan Rose * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) | 1 line Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme ........ 2011-08-11 21:54 +0000 [r331579] Jason Parker * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) | 6 lines Use proper values for 64-bit option flags. Also, reusing bits es no bueno, so change the value of a duplicate. (issue ASTERISK-18239) ........ 2011-08-11 21:42 +0000 [r331576] Richard Mudgett * /, funcs/func_shell.c: Merged revisions 331575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines Segfault in shell_helper in func_shell.c. The return value of popen() was not checked for failure to open. (closes issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael Myles Tested by: rmudgett ........ 2011-08-10 22:23 +0000 [r331518] Kinsey Moore * /, channels/chan_sip.c: Merged revisions 331517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed. (closes issue ASTERISK-18091) Review: https://reviewboard.asterisk.org/r/1332/ ........ 2011-08-10 20:41 +0000 [r331418-331462] Richard Mudgett * /, main/logger.c: Merged revisions 331461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011) | 30 lines Output of queue log not started until logger reloaded. ASTERISK-15863 caused a regression with queue logging. The output of the queue log is not started until the logger configuration is reloaded. * Queue log initialization is completely delayed until the first message is posted to the queue log system. Including the initial opening of the queue log file. * Fixed rotate_file() ROTATE strategy to give the file just rotated out to the configured exec function after rotate. Just like the other strategies. * Fixed logger reload to always post the queue reload entry instead of just if there is a queue log file. * Refactored some code to eliminate some redundancy and to reduce stack utilization. (closes issue ASTERISK-17036) JIRA SWP-2952 Reported by: Juan Carlos Valero Patches: jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett (closes issue ASTERISK-18208) Reported by: Christian Pinedo Review: https://reviewboard.asterisk.org/r/1333/ ........ * main/features.c: Make sure feature_request_and_dial() initializes outstate if passed in. * main/features.c, CHANGES: Revert -r318141. It was a band-aid that only partially fixed parking. A better fix is on reviewboard review 1358. (issue ASTERISK-17374) 2011-08-10 13:48 +0000 [r331316] Kinsey Moore * main/manager.c, /: Merged revisions 331315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) | 8 lines AMI action ModuleReload returns Error if Module: missing or empty An empty string was not being checked for properly causing identification of the module to be reloaded to fail and return an Error with message "No such module." (closes issue AST-616) ........ 2011-08-09 23:12 +0000 [r331265] Richard Mudgett * apps/app_parkandannounce.c, main/pbx.c, /, channels/chan_sip.c, main/features.c, channels/chan_iax2.c: Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ 2011-08-09 16:36 +0000 [r331147-331200] Alexandr Anikin * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooGkClient.c: Setup IP proto version for call in GK mode Added additional check for IP semantics before parse destination by ast_parse_args due to it can parse numeric as IP. (closes issue ASTERISK-18218) Reported by: slesru Patch: ASTERISK-18218.patch * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c, /, addons/ooh323c/src/ooLogChan.c: Merged revisions 331146 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4 lines move ast_cond_signal for admitted call after all data filled/freed clear all log channels by pointed number not only first free allocated callToken in ooh323_answer ........ 2011-08-09 15:59 +0000 [r331138-331143] Jason Parker * /, doc/asterisk.8: Merged revisions 331142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug 2011) | 1 line Regenerate asterisk man page from sgml. ........ * doc/asterisk.sgml, /, doc/asterisk.8, configs/asterisk.conf.sample, configs/voicemail.conf.sample: Merged revisions 306999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) | 12 lines Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir ........ * doc/asterisk.sgml, /, doc/asterisk.8, configs/asterisk.conf.sample, configs/voicemail.conf.sample: Revert merge of r306999, due to merge conflict. 2011-08-08 22:59 +0000 [r331041-331097] Terry Wilson * UPGRADE.txt, CHANGES, include/asterisk/manager.h: Bump the AMI protocol version to 1.2 As a result of converting Unlink events that were missed in the AMI 1.1 update to Bridge events, the AMI protocol version is being incremented. * main/channel.c, CHANGES: Replace AMI Unlink events with Bridge events A previous update converted some of the Link and Unlink events to Bridge events, but a couple of Unlink events were missed. This patch rectifies the situation. (closes issue ASTERISK-17455) 2011-08-08 20:53 +0000 [r331039] Kinsey Moore * /, res/res_musiconhold.c: Merged revisions 331038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) | 11 lines In-queue MOH stops after a periodic announcement If the seek value is past the end of file when resuming G.722 MOH, MOH will cease to function for the duration of the MOH session through all starts and stops until saved state is cleared. Adjusting the code to guarantee a single valid read (which is already assumed) fixes the bug. (closes issue ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/ Tested-by: Jonathan Rose ........ 2011-08-05 15:53 +0000 [r330940] David Vossel * codecs/codec_resample.c: The slin resampler is no longer dependent on an external library, but the dependency was not removed correctly. 2011-08-05 07:38 +0000 [r330899] Alexandr Anikin * addons/ooh323c/src/ooGkClient.c, /, addons/ooh323c/src/ooCmdChannel.c: Merged revisions 330827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4 lines change gk client behaivour on rrq/grq failures to setup timers and next tries after timeout instead of complete failure in the ooh323 stack ........ 2011-08-04 20:51 +0000 [r330844] Terry Wilson * /, configure, configure.ac: Merged revisions 330843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330843 | twilson | 2011-08-04 15:29:19 -0500 (Thu, 04 Aug 2011) | 4 lines Make libsrtp instructions more explicit when linking fails (closes issue ASTERISK-18139) ........ 2011-08-03 15:15 +0000 [r330706-330763] Kinsey Moore * /, main/Makefile: Merged revisions 330762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) | 9 lines editing files in main/editline does not ensure rebuild of libedit.a When editing a source file in main/editline, the build system does not rebuild libedit.a and uses the already existing one instead. Adding a PHONY to CHECK_SUBDIR fixes this problem. (closes issue ASTERISK-16221) Patch-by: Walter Doekes ........ * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions 330705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) | 10 lines Call pickup broken for DAHDI channels when beginning with # The call pickup feature did not work on DAHDI devices for anything other than feature codes beginning with * since all feature codes in chan_dahdi were originally hard-coded to begin with *. This patch is also applied to chan_dahdi.c to fix this bug with radio modes. (closes issue AST-621) Review: https://reviewboard.asterisk.org/r/1336/ ........ 2011-08-02 20:52 +0000 [r330649] Kevin P. Fleming * /, res/res_jabber.c: Merged revisions 330648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02 Aug 2011) | 2 lines Convert an error message to actually be helpful. ........ 2011-08-02 16:17 +0000 [r330576-330586] David Vossel * /, channels/chan_iax2.c: Merged revisions 330581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) | 8 lines Fixes crash in chan_iax2. Fixes crash in chan_iax2 resulting from an edge case in the way control frames are queued during calltoken negotiation is complete. (closes issue ASTERISK-17610) Reported by: mgrobecker ........ * /, channels/chan_sip.c: Merged revisions 330578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) | 2 lines Optimization to buffer initialization fix. ........ * /, channels/chan_sip.c: Merged revisions 330575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) | 5 lines Fixes uninitialized string buffer in log message. (closes issue ASTERISK-17200) Reported by: lmadsen ........ 2011-08-01 15:23 +0000 [r330434] Kinsey Moore * /, main/say.c: Merged revisions 330433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) | 9 lines Incorrect playback for Spanish in some circumstances When you say the time in spanish and it is 01:00 - 01:59 or 13:00 - 13:59 you must use female pronunciation "1F". The function "say_date_with_format_es" does not take this in account. (closes ASTERISK-15016) Patch-by: Luis Jimenez ........ 2011-07-30 23:57 +0000 [r330369] Richard Mudgett * main/channel.c, /: Merged revisions 330368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011) | 4 lines Remove some redundant locking code in ast_do_masquerade(). Also updated some comments. ........ 2011-07-30 15:34 +0000 [r330312] Gregory Nietsky * main/channel.c, /: Merged revisions 330311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) | 9 lines prevent double masqurading channels when one is been hung up and deadlock avoidance is used. There is a race condition in ast_do_masquerade / ast_hangup (at least) Reported by me signed off by schmidts with input from David Vossel Review: https://reviewboard.asterisk.org/r/1323/ ........ 2011-07-29 17:19 +0000 [r330204-330217] Sean Bright * /, formats/format_wav.c: Merged revisions 330213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri, 29 Jul 2011) | 2 lines Correct the check for O_RDONLY. ........ * /, formats/format_wav.c: Merged revisions 330203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri, 29 Jul 2011) | 2 lines Only write to wav files that were opened to be written to. ........ 2011-07-29 05:25 +0000 [r330162] Paul Belanger * apps/app_confbridge.c: Fix typo pointed out on #asterisk Thanks notten 2011-07-28 21:44 +0000 [r330108] Terry Wilson * main/term.c, /: Merged revisions 330107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28 Jul 2011) | 2 lines Make console colors work for TERM=xterm-256color ........ 2011-07-28 17:10 +0000 [r330051] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 330050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r330050 | rmudgett | 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) | 22 lines Merged revisions 330033 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and outgoing call legs of a data call are using different formats: a-law, u-law. When the call is bridged, the media stream is run through translation to convert the media formats. The translation is bad for data calls. * Make incoming call that does not explicitly specify u-law or a-law use the DAHDI channel's default law. The outgoing call always uses the default law from the DAHDI channel. (closes issue ABE-2800) Patches: jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett .......... ................ 2011-07-28 15:45 +0000 [r329995] Jason Parker * /, channels/chan_sip.c: Merged revisions 329994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) | 6 lines Fix a SIP transfer deadlock. The locking in this function is very scary. There are like 6 structs involved. (closes issue AST-470) ........ 2011-07-28 15:28 +0000 [r329992] Matthew Nicholson * /, res/res_fax.c: Merged revisions 329991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul 2011) | 6 lines check for CONFIG_STATUS_FILE_INVALID when loading the res_fax config file Patch by: tzafrir Reported by: tzafrir (closes issue ASTERISK-18161) ........ 2011-07-28 13:03 +0000 [r329771-329952] Sean Bright * configs/confbridge.conf.sample: The default conf-usermenu says that '8' can be used to leave the conference, so put that in the sample user menu. '5' is supposed to extend the conference, but there doesn't appear to be a concept of that in the menu actions. * apps/app_confbridge.c: Correct the spelling of 'conference.' * /, channels/chan_sip.c: Merged revisions 329895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) | 2 lines Make the output of Externhost in 'sip show settings' more consistent. ........ * /, Makefile.moddir_rules: Merged revisions 329767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed, 27 Jul 2011) | 8 lines Explicitly sort the module list so that the menuselect lists are sorted. (closes issue ASTERISK-18141) Reported by: Richard Miller Patches: sort-order.diff uploaded by seanbright (License #5060) Tested by: leifmadsen ........ 2011-07-27 18:11 +0000 [r329710] Jonathan Rose * /, configs/indications.conf.sample: Merged revisions 329709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) | 8 lines Fix New Zealand indications profile based on http://www.telepermit.co.nz/TNA102.pdf (closes issue ASTERISK-16263) Reported by: richardf Patches: nz-indications.patch uploaded by richardf (License #6015) ........ 2011-07-27 15:25 +0000 [r329670] Sean Bright * main/loader.c: Sort the module list so that 'module show' is alphabetical. 2011-07-27 04:25 +0000 [r329614] Tilghman Lesher * /, cdr/cdr_odbc.c: Merged revisions 329613 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011) | 6 lines Duration and billsec are swapped in high resolution time. Closes ASTERISK-18024 Patches: 20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003) ........ 2011-07-26 14:19 +0000 [r329528-329538] Jonathan Rose * apps/app_voicemail.c, /: Merged revisions 329529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) | 5 lines Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more appropriate anyway. ........ * apps/app_voicemail.c, include/asterisk/app.h, /, configs/voicemail.conf.sample, main/app.c: Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ 2011-07-25 19:55 +0000 [r329472] Paul Belanger * /, main/enum.c: Merged revisions 329471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon, 25 Jul 2011) | 2 lines Decrease verbose messages to debug, to help clean up CLI. ........ 2011-07-25 14:06 +0000 [r329430-329431] Gregory Nietsky * main/dsp.c, include/asterisk/dsp.h: dsp_process was enhanced to work with alaw and ulaw in addition to slin. noticed that some functions could be refactored here it is. Reported by: irroot Tested by: irroot, mnicholson Review: https://reviewboard.asterisk.org/r/1304/ * channels/chan_sip.c, channels/sip/include/sip.h: Remove lastmsgssent from sip it has not been working since 1.6 Clean up the return values to be consistant not currently used Add doxygen returns MWI Event is sent on Register (closes issue ASTERISK-17866) Reported by: one47 Tested by: irroot, mvanbaak Review: https://reviewboard.asterisk.org/r/1272/ 2011-07-22 21:14 +0000 [r329331-329334] Richard Mudgett * main/pbx.c, /: Make use less redundant loop construct for iterating over hints. * main/pbx.c, /: Merged revisions 329299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011) | 48 lines Deadlocks dealing with dialplan hints during reload. There are two remaining different deadlocks reported dealing with dialplan hints. The deadlock in ASTERISK-17666 is caused by invalid locking order in ast_remove_hint(). The hints container must be locked before the hint object. The deadlock in ASTERISK-17760 is caused by a catch-22 situation in handle_statechange(). The deadlock is caused by not having the conlock before calling the watcher callbacks. Unfortunately, having that lock causes a different deadlock as reported in ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made handle_statechange() no longer call the watcher callbacks holding any locks that matter. * Made hint ao2 destructor do the watcher callbacks for extension deactivation to guarantee that they get called. * Fixed hint reference leak in ast_add_hint() if the callback container constructor failed. * Fixed hint reference leak in complete_core_show_hint() for every hint it found for CLI tab completion. * Adjusted locking in ast_merge_contexts_and_delete() for safety. * Added context_merge_lock to prevent ast_merge_contexts_and_delete() and handle_statechange() from interfering with each other. * Fixed ast_change_hint() not taking into account that the extension is used for the hash key. (closes issue ASTERISK-17666) Reported by: irroot Tested by: irroot JIRA SWP-3318 (closes issue ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/ ........ 2011-07-22 Leif Madsen * Asterisk 10.0.0-beta1 Released. 2011-07-21 20:22 +0000 [r329257] Russell Bryant * channels/chan_dahdi.c, main/features.c, include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c, include/asterisk/rtp_engine.h: s/1.10/10.0/ 2011-07-21 18:05 +0000 [r329200-329204] Richard Mudgett * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 329203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. ........ * /, apps/app_directed_pickup.c: Merged revisions 329199 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines Update PickupChan documentation. The PickupChan uses the ampersand as the argument separator. Was documented as: PickupChan(channel[,channel2[,...][,options]]) Fixed documentation to: PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) This is a continuation of ASTERISK-17494 for v1.8 and later. (closes issue ASTERISK-18144) Reported by: Erik Smith Patches: pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith Tested by: Erik Smith ........ 2011-07-21 17:27 +0000 [r329188] Jason Parker * UPGRADE.txt: Fix version number in UPGRADE.txt. 2011-07-21 16:52 +0000 [r329145] Richard Mudgett * /, main/features.c: Merged revisions 329144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked! This appears to be a leftover from when ast_channel was converted to ao2 objects. Simply removed the extraneous unlock. (closes issue ASTERISK-17772) ........ 2011-07-21 16:04 +0000 [r329104] Russell Bryant * / (added): Change Asterisk 2.0 to 2.0 in binary 2011-07-20 21:31 +0000 [r329056] Paul Belanger * /, UPGRADE-1.8.txt: Merged revisions 329055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r329055 | pabelanger | 2011-07-20 17:27:50 -0400 (Wed, 20 Jul 2011) | 9 lines Merged revisions 329027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, 20 Jul 2011) | 2 lines Asterisk now requires libpri 1.4.11+ for PRI support. ........ ................ 2011-07-20 20:19 +0000 [r328996] Terry Wilson * /, tests/test_netsock2.c: Merged revisions 328992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r328992 | twilson | 2011-07-20 15:18:25 -0500 (Wed, 20 Jul 2011) | 12 lines Merged revisions 328987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) | 5 lines We can't guarantee an eth0 is present FreeBSD test fails on this case presumably because there is no eth0 on the test machine. Better to just remove this test for now. ........ ................ 2011-07-20 19:03 +0000 [r328937] Kinsey Moore * /, channels/chan_sip.c: Merged revisions 328936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines Merged revisions 328935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines Inband DTMF regression The functionality of inband DTMF in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF, never inband. This fixes the regression introduced in revision 328823. ........ ................ 2011-07-19 21:32 +0000 [r328880-328881] Kevin P. Fleming * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged revisions 328879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r328879 | kpfleming | 2011-07-19 16:31:16 -0500 (Tue, 19 Jul 2011) | 23 lines Merged revisions 328878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines Revert partial attempt at handling pathnames with spaces. Revision 299794 attempted to improve the build system to be able to handle pathnames (primarily DESTDIR) with spaces in them, since this is common on some platforms (including Mac OSX). Unfortunately, the changes were incomplete and did not actually provide the desired behavior, and as a side effect the functionality that ensured that stale headers in the Asterisk 'include' directory were removed got broken. In addition, the check for stale (and possibly incompatible) modules in the Asterisk 'modules' directory also got broken, and would never report any stale modules. Users upgrading to this version or later versions would then see unexpected module load errors. Since there are few users who actually want to install Asterisk into paths that contain spaces, and a proper fix for the build system would take many hours, the best solution for now is to just revert the partial solution. ........ ................ * /: Edit the merge properties to match their names. 2011-07-19 21:21 +0000 [r328877] Russell Bryant * /: Fix properties after twilson's change so merges still work 2011-07-19 18:07 +0000 [r328772-328825] Kinsey Moore * res/res_rtp_asterisk.c, main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged revisions 328824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines Merged revisions 328823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ ........ ................ * /, apps/app_meetme.c: Merged revisions 328771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines Merged revisions 328770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines MeetMe requests a PIN twice in some circumstances If a call to MeetMe includes both the dynamic(D) and always request PIN(P) options, MeetMe will ask for the PIN two times: once for creating the conference and once for entering the conference. This behavior was introduced in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN entry for joining a conference. (closes AST-601) Review: https://reviewboard.asterisk.org/r/1305/ ........ ................ 2011-07-19 02:00 +0000 [r328718] Terry Wilson * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c (added): Merged revisions 328717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328717 | twilson | 2011-07-18 20:55:32 -0500 (Mon, 18 Jul 2011) | 14 lines Merged revisions 328716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) | 7 lines Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This commit also adds linked list unit tests. Review: https://reviewboard.asterisk.org/r/1321/ ........ ................ 2011-07-18 20:51 +0000 [r328610-328665] Mark Murawki * apps/app_dial.c, /: Merged revisions 328664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines Merged revisions 328663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines app_dial may double free a channel datastore When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it. (closes issue ASTERISK-17917) Reported by: Mark Murawski Tested by: Mark Murawski ........ ................ * /, channels/chan_sip.c: Merged revisions 328611 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines Merged revisions 328608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines If the sip private structure is null, sip_setoption() will defref the null pointer and crash. Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash. (closes issue ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark Murawski ........ ................ * /, main/asterisk.c: Merged revisions 328609 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328609 | markm | 2011-07-18 08:37:53 -0400 (Mon, 18 Jul 2011) | 15 lines Merged revisions 328593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | 8 lines Fixed invalid read and null pointer deref on asterisk shutdown. In some cases when starting asterisk with -c and hitting control-c to shutdown, there will be an invalid read and null pointer deref causing a crash. (closes issue ASTERISK-17927) Reported by: Mark Murawski Tested by: Mark Murawski, Kinsey Moore, Tilghman Lesher ........ ................ 2011-07-18 07:12 +0000 [r328542] Tilghman Lesher * /, funcs/func_odbc.c: Merged revisions 328541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328541 | tilghman | 2011-07-18 02:11:26 -0500 (Mon, 18 Jul 2011) | 9 lines Merged revisions 328540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 Jul 2011) | 2 lines Typo ........ ................ 2011-07-15 21:41 +0000 [r328502] Alexandr Anikin * addons/ooh323c/src/ooGkClient.c, /: Merged revisions 328428-328429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328428 | may | 2011-07-15 23:31:09 +0400 (Fri, 15 Jul 2011) | 13 lines Merged revisions 328427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 lines small gk processing fixes: - decrease for 1 second registration ttl for very low expirations (some providers expire few earlier than TTL) - delete rrq and registration expire timers on URQ received as we make new registration. ........ ................ r328429 | may | 2011-07-15 23:35:50 +0400 (Fri, 15 Jul 2011) | 2 lines delete unproperly changed svn props ................ 2011-07-15 21:19 +0000 [r328449-328459] Leif Madsen * /, apps/app_macro.c: Merged revisions 328451 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) | 1 line Build app_macro by default because things depend on it. ........ * /, UPGRADE-1.10.txt, UPGRADE.txt, CHANGES: Merged revisions 328448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) | 2 lines Update UPGRADE.txt and CHANGES files. Update documentation files stating that deprecated modules are no longer built by default. ........ 2011-07-15 08:19 +0000 [r328381] Damien Wedhorn * channels/chan_skinny.c: Add SLA to skinny. Adds sublines to skinny lines. Each subline can be attached to an SLA station/trunk combo. Includes the following functionality: Callid is persistent for both in/out calls on all skinny devices. Can join, hold, resume. All sublines appear under a single line button. See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc. (closes issue ASTERISK-17947) Review: https://reviewboard.asterisk.org/r/1239/ 2011-07-15 00:23 +0000 [r328318-328344] Richard Mudgett * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c, include/asterisk/extconf.h, include/asterisk/pbx.h, apps/app_queue.c: Merged revisions 328329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines Make hint watcher callback take const strings for context and exten parameters. ........ * /, channels/chan_sip.c: Merged revisions 328317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328317 | rmudgett | 2011-07-14 18:28:49 -0500 (Thu, 14 Jul 2011) | 13 lines Merged revisions 328302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) | 6 lines Missing SIP pvt and channel unlock in sip_set_rtp_peer(). Regression introduced by -r326144. Add missing SIP pvt and channel unlock in sip_set_rtp_peer(). ........ ................ 2011-07-14 20:28 +0000 [r328259] Leif Madsen * funcs/func_speex.c, apps/app_playtones.c, bridges/bridge_softmix.c, apps/app_alarmreceiver.c, res/res_calendar_caldav.c, apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c, res/res_pktccops.c, channels/chan_skinny.c, pbx/pbx_ael.c, formats/format_h263.c, cdr/cdr_odbc.c, cdr/cdr_manager.c, utils/refcounter.c, funcs/func_timeout.c, formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c, bridges/bridge_simple.c, funcs/func_cut.c, apps/app_talkdetect.c, apps/app_db.c, funcs/func_callcompletion.c, funcs/func_channel.c, funcs/func_iconv.c, pbx/pbx_config.c, res/res_odbc.c, apps/app_voicemail.c, formats/format_sln.c, apps/app_authenticate.c, apps/app_readexten.c, res/res_phoneprov.c, apps/app_userevent.c, codecs/codec_gsm.c, tests/test_func_file.c, apps/app_setcallerid.c, res/res_config_odbc.c, funcs/func_audiohookinherit.c, apps/app_osplookup.c, funcs/func_odbc.c, cel/cel_custom.c, tests/test_utils.c, apps/app_mp3.c, res/res_timing_timerfd.c, codecs/codec_resample.c, formats/format_h264.c, apps/app_directory.c, formats/format_siren14.c, tests/test_amihooks.c, res/res_config_pgsql.c, funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c, res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c, apps/app_forkcdr.c, funcs/func_blacklist.c, apps/app_sms.c, formats/format_g723.c, utils/extconf.c, tests/test_poll.c, apps/app_stack.c, apps/app_verbose.c, utils/check_expr.c, funcs/func_module.c, codecs/codec_adpcm.c, tests/test_event.c, cdr/cdr_adaptive_odbc.c, apps/app_image.c, formats/format_wav_gsm.c, utils/stereorize.c, pbx/pbx_loopback.c, tests/test_time.c, funcs/func_shell.c, apps/app_skel.c, channels/chan_alsa.c, apps/app_externalivr.c, apps/app_milliwatt.c, formats/format_gsm.c, res/res_speech.c, apps/app_dial.c, apps/app_page.c, apps/app_fax.c, utils/astman.c, apps/app_disa.c, res/res_monitor.c, apps/app_waitforring.c, addons/cdr_mysql.c, res/res_fax_spandsp.c, res/res_timing_kqueue.c, apps/app_chanspy.c, apps/app_cdr.c, channels/chan_unistim.c, funcs/func_base64.c, channels/chan_multicast_rtp.c, funcs/func_md5.c, apps/app_meetme.c, tests/test_gosub.c, funcs/func_sysinfo.c, funcs/func_vmcount.c, res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c, res/res_config_sqlite.c, apps/app_controlplayback.c, cdr/cdr_csv.c, formats/format_ilbc.c, channels/chan_phone.c, funcs/func_enum.c, main/manager.c, funcs/func_groupcount.c, tests/test_stringfields.c, tests/test_locale.c, tests/test_devicestate.c, funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c, tests/test_astobj2.c, apps/app_ivrdemo.c, res/res_clioriginate.c, apps/app_jack.c, apps/app_nbscat.c, res/res_calendar_icalendar.c, codecs/codec_a_mu.c, tests/test_ast_format_str_reduce.c, tests/test_dlinklists.c, res/res_convert.c, apps/app_waituntil.c, pbx/pbx_lua.c, utils/astcanary.c, apps/app_queue.c, channels/chan_oss.c, cdr/cdr_tds.c, channels/chan_usbradio.c, apps/app_flash.c, apps/app_senddtmf.c, funcs/func_callerid.c, addons/app_saycountpl.c, cel/cel_pgsql.c, apps/app_dahdibarge.c, channels/chan_local.c, funcs/func_dialgroup.c, tests/test_logger.c, apps/app_record.c, funcs/func_env.c, funcs/func_strings.c, res/res_timing_dahdi.c, apps/app_chanisavail.c, bridges/bridge_multiplexed.c, res/res_rtp_multicast.c, cel/cel_odbc.c, channels/chan_dahdi.c, pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_pcm.c, apps/app_dumpchan.c, main/http.c, res/res_clialiases.c, res/res_calendar_exchange.c, res/res_ais.c, funcs/func_sprintf.c, codecs/codec_g722.c, tests/test_expr.c, cel/cel_tds.c, tests/test_app.c, utils/smsq.c, apps/app_morsecode.c, formats/format_ogg_vorbis.c, tests/test_sched.c, res/res_calendar_ews.c, apps/app_speech_utils.c, tests/test_acl.c, apps/app_sendtext.c, funcs/func_cdr.c, utils/hashtest2.c, utils/ael_main.c, apps/app_mixmonitor.c, formats/format_g726.c, utils/streamplayer.c, res/res_calendar.c, cel/cel_radius.c, channels/chan_vpb.cc, tests/test_heap.c, addons/format_mp3.c, res/res_snmp.c, apps/app_dictate.c, channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c, res/res_jabber.c, funcs/func_uri.c, cel/cel_manager.c, apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, codecs/codec_lpc10.c, apps/app_read.c, cdr/cdr_syslog.c, codecs/codec_alaw.c, res/res_adsi.c, agi/eagi-test.c, utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c, formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c, pbx/pbx_dundi.c, funcs/func_devstate.c, addons/res_config_mysql.c, funcs/func_rand.c, apps/app_readfile.c, addons/chan_ooh323.c, cdr/cdr_sqlite3_custom.c, /, apps/app_sayunixtime.c, apps/app_test.c, res/res_http_post.c, res/res_smdi.c, main/features.c, funcs/func_srv.c, apps/app_amd.c, pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c, formats/format_g719.c, channels/chan_bridge.c, apps/app_privacy.c, apps/app_echo.c, codecs/codec_speex.c, apps/app_saycounted.c, apps/app_dahdiras.c, channels/chan_agent.c, funcs/func_math.c, res/res_ael_share.c, apps/app_transfer.c, res/res_mutestream.c, apps/app_playback.c, res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c, tests/test_skel.c, apps/app_macro.c, apps/app_zapateller.c, codecs/codec_ilbc.c, addons/app_mysql.c, tests/test_substitution.c, utils/muted.c, utils/hashtest.c, funcs/func_sha1.c, formats/format_siren7.c, tests/test_security_events.c, funcs/func_config.c, bridges/bridge_builtin_features.c, funcs/func_volume.c, res/res_agi.c, apps/app_confbridge.c, addons/chan_mobile.c, apps/app_parkandannounce.c, res/res_security_log.c, cdr/cdr_custom.c, apps/app_while.c, res/res_rtp_asterisk.c, funcs/func_dialplan.c, funcs/func_db.c, apps/app_festival.c, res/res_limit.c, res/res_fax.c, apps/app_waitforsilence.c, channels/chan_console.c, apps/app_getcpeid.c, funcs/func_global.c, res/res_srtp.c, funcs/func_extstate.c, tests/test_strings.c, res/res_timing_pthread.c, apps/app_directed_pickup.c, channels/chan_h323.c, cel/cel_sqlite3_custom.c, codecs/codec_ulaw.c, channels/chan_nbs.c, formats/format_g729.c: Merged revisions 328247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ 2011-07-14 19:56 +0000 [r328208] Jonathan Rose * /, res/res_monitor.c: Merged revisions 328207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328207 | jrose | 2011-07-14 14:45:18 -0500 (Thu, 14 Jul 2011) | 13 lines Merged revisions 328205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | 6 lines Monitor application arguments requirements fixed. Monitor was requiring options in spite of no individual option on Monitor being required. Review: https://reviewboard.asterisk.org/r/1320/ ........ ................ 2011-07-14 17:47 +0000 [r328163] Matthew Nicholson * /, main/dsp.c: Merged revisions 328162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul 2011) | 3 lines tune the v21 preamble detector to properly detect the desired sequence of hits and misses ........ 2011-07-13 22:10 +0000 [r328121] David Vossel * /, apps/app_mixmonitor.c: Merged revisions 328120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines Preserve sample rate quality of wideband mixmonitor recordings. MixMonitor has the ability to record in any file format Asterisk supports, but the quality of wideband audio is not preserved. This is because regardless of the sample rate the call is being recorded in, the audio is always downsampled to 8khz and then upsampled to whatever wideband format it is being written as. This patch resolves this by requesting the audio from the audiohook in the signed linear format closest to the sample rate of the format we are writing. This fix is only possible for Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband audio. Review: https://reviewboard.asterisk.org/r/1314/ ........ 2011-07-13 21:06 +0000 [r328079] Leif Madsen * BUGS, UPGRADE-1.10.txt (added), UPGRADE.txt: Add UPGRADE-1.10.txt file from UPGRADE.txt. 2011-07-13 20:40 +0000 [r328075-328076] Russell Bryant * /: set 1.10 merge properties * /: remove 1.8 merge properties 2011-07-13 18:47 +0000 [r328016] Richard Mudgett * /, configs/features.conf.sample: Merged revisions 328014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011) | 1 line Add ATXFER_NULL_TECH note in features.conf.sample. ........ 2011-07-12 23:02 +0000 [r327953] Kevin P. Fleming * main/manager.c, /: Merged revisions 327950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul 2011) | 14 lines Correct double-free situation in manager output processing. The process_output() function calls ast_str_append() and xml_translate() on its 'out' parameter, which is a pointer to an ast_str buffer. If either of these functions need to reallocate the ast_str so it will have more space, they will free the existing buffer and allocate a new one, returning the address of the new one. However, because process_output only receives a pointer to the ast_str, not a pointer to its caller's variable holding the pointer, if the original ast_str is freed, the caller will not know, and will continue to use it (and later attempt to free it). (reported by jkroon on #asterisk-dev) ........ 2011-07-12 20:08 +0000 [r327891] Matthew Nicholson * /, apps/app_directory.c: Merged revisions 327890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue, 12 Jul 2011) | 2 lines search in the current context for 'a' and 'o' instead of 'default' ........ 2011-07-12 19:39 +0000 [r327889] Jason Parker * Makefile, /: Merged revisions 327888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327888 | qwell | 2011-07-12 14:38:44 -0500 (Tue, 12 Jul 2011) | 1 line Fix uninstall target, so that modules dir gets cleared again. ........ 2011-07-12 19:18 +0000 [r327856] Matthew Jordan * /, apps/app_voicemail.c: Merged revisions 327852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) | 12 lines Added additional checks for mailbox / password beginning with '*' character A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character. (closes issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/ ........ 2011-07-12 15:38 +0000 [r327794] Tilghman Lesher * tests/test_substitution.c, /: Merged revisions 327793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011) | 14 lines Use 'printf' (POSIX issue 4) instead of 'echo -n', for portability. The problem with using 'echo -n' is that it is not portable. While BSD systems required that the '-n' option be removed and interpreted, System V required that all strings should be echoed with no interpretation of options. This fundamental difference of behavior means that it is never possible to use the '-n' flag to echo in tests which are meant to be portable. In this case, on Mac OS X 10.6, the /bin/sh shell builtin 'echo' uses the System V semantics of the command, and thus the SHELL test failed on that platform. http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16 ........ 2011-07-12 15:23 +0000 [r327769] Matthew Nicholson * res/res_fax.c, include/asterisk/dsp.h, main/dsp.c: do v21 detection instead of CED detection for the fax gateway 2011-07-12 14:55 +0000 [r327749] David Vossel * main/bridging.c: Send video update frame to new video source in follow_talker correctly. 2011-07-12 14:40 +0000 [r327748] Kinsey Moore * apps/app_confbridge.c: Segfault on shutdown when confbridge is active When undergoing a shutdown and channels are kicked out of a bridge, a segfault occurs because ConfBridge tries to play sounds on the bridge after the underlying channels have been blown away due to the shutdown. (closes ASTERISK-18040) Review: https://reviewboard.asterisk.org/r/1283/ 2011-07-11 20:06 +0000 [r327684] Matthew Nicholson * tests/test_substitution.c: use printf instead of echo -n in substitution test 2011-07-11 19:49 +0000 [r327683] Terry Wilson * /, include/asterisk/jingle.h, channels/chan_gtalk.c: Merged revisions 327682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) | 9 lines Update chan_gtalk to work with changed GMail-based calls The messages sent by the GMail client have changed, but include the old-style messages as well. This patch checks for this case and uses the old-style offer. (closes issue ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/ ........ 2011-07-11 18:44 +0000 [r327640] David Vossel * include/asterisk/bridging.h, bridges/bridge_softmix.c, main/bridging.c: Updates follow_talker video_mode in confbridge application. follow_talker mode originally echoed the same video stream to all participants. As the primary talker switched around, the video stream would result in the talker seeing themselves. Now the primary talker sees the last person who was talking rather than themselves. 2011-07-11 17:23 +0000 [r327469-327598] Matthew Nicholson * res/res_fax.c: renamed fax_gateway_send_ced() to fax_gateway_request_t38() * res/res_fax.c: actually do something with the ced timeout, also added more debug output * res/res_fax.c: write silence on the channel during t.38 negotiation * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul 2011) | 2 lines reset our buffer each iteration when doing variable substitution ........ * res/res_fax.c: Delay sending an CED tone generated T.38 reinvite to give the CED tone generating party time to send its own T.38 reinvite. Also don't forward frames through the gateway if we are negotiating T.38. * res/res_fax.c: fixed wording in a comment 2011-07-11 10:57 +0000 [r327413] Tzafrir Cohen * /, main/Makefile: Merged revisions 327411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327411 | tzafrir | 2011-07-11 13:46:34 +0300 (ב', 11 יול 2011) | 5 lines fix building the Debian armhf (HardFloat) port Fixes http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385 (Missing pthreads) ........ 2011-07-10 01:37 +0000 [r327359] Alexandr Anikin * addons/chan_ooh323.c, configs/chan_ooh323.conf.sample: Full T.38 handshaking and fax detection Add full t.38 handshaking for OOH323 that are required for newest T.38 gateway codes. Add fax detection (cng tone, t38) and dialplan redirection to fax ext on fax event detected. Add OOH323() function to set/get t38support and faxdetect parameters. (closes issue ASTERISK-17754) Reported by: irroot Patches: ooh323_faxdetect.patch uploaded by irroot (license 52) issue19183-final.patch uploaded by may213 (license 454) Tested by: may213, irroot Review: https://reviewboard.asterisk.org/r/1174/ 2011-07-08 22:25 +0000 [r327246] Jason Parker * main/stdtime, utils, codecs, utils/db1-ast/recno, apps, cel, apps/confbridge, cdr, formats, codecs/gsm/src, utils/db1-ast/hash, funcs, bridges, codecs/lpc10, utils/db1-ast/db, codecs/g722, utils/db1-ast/mpool, main, codecs/speex, channels/sip, pbx, res, res/ael, channels, utils/db1-ast/btree: Add .o files to svn:ignore property, since it's only ignored if locally configured to do so. 2011-07-08 21:43 +0000 [r327212] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 327211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines INVITE 403 Forbidden response always retransmits the maximum times. Asterisk sends a 403 Forbidden response if authentication fails for an INVITE as required. However, it ignores the ACK and keeps retransmitting the response. * Made not delete the to-tag in the dialog so the expected ACK can be matched with the dialog and stop the retransmissions. ........ 2011-07-08 20:33 +0000 [r327116-327168] David Vossel * UPGRADE.txt, CHANGES: Adds entry in UPDATES.txt for removal of formats/format_sln16.c. Fixes typo in CHANGES as well. * CHANGES: Updates CHANGES log to reflect new slinear read/write file interpreters. * formats/format_sln.c, formats/format_sln16.c (removed): Support for writing and reading raw slin files 8khz-192khz. * formats/format_attr_silk.c (removed), formats/format_attr_celt.c (removed), res/res_format_attr_silk.c (added), res/res_format_attr_celt.c (added): Moves celt and silk format attribute files into res folder. It was inconsistent to have the silk and celt format attribute modules in the format file interpreter folder. 2011-07-08 19:54 +0000 [r327107] Matthew Nicholson * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul 2011) | 11 lines Reset our ast_str before passing it on to dialplan function backends. It is possible for a dialplan backend to not modify the given buffer or ast_str and still return success. This causes any previous value stored in the buffer to be used as if the new function call provided it. Some functions also append to the given buffer assuming it is empty. The test_substitution unit test has also been modified to detect this problem. (closes issue ASTERISK-17878) ........ 2011-07-08 16:00 +0000 [r327045-327047] Russell Bryant * /, tests/test_netsock2.c: Merged revisions 327046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327046 | russell | 2011-07-08 11:00:05 -0500 (Fri, 08 Jul 2011) | 2 lines Fix an error and add more log message info to help see why this fails on FreeBSD. ........ * channels/chan_dahdi.c, /: Merged revisions 327044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08 Jul 2011) | 2 lines Resolve some set-but-unused-variable warnings. ........ 2011-07-08 01:26 +0000 [r327000] Richard Mudgett * main/pbx.c, /: Merged revisions 326985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) | 12 lines Some code cleanup in pbx.c * Mostly comment and format changes. * ast_context_remove_extension_callerid() and ast_add_extension_nolock() will write lock the found specific context. * ast_context_find() will now tolerate a NULL name. * Eliminated some inlined versions of find_context() and find_context_locked(). ........ 2011-07-07 22:39 +0000 [r326943] Jason Parker * include/asterisk/celt.h: I think reviewboard broke this. The whole file was doubled. 2011-07-07 22:17 +0000 [r326855-326904] David Vossel * formats/format_attr_celt.c (added): Adds the format_attr_celt file which was also missing from the CELT pass through patch. * include/asterisk/celt.h (added): Adds missing celt.h file from celt pass-through support patch. * CHANGES: Fixes spelling errors in CHANGES as well as adding a few entries for CELT and confbridge. * main/channel.c, main/format.c, res/res_rtp_asterisk.c, main/frame.c, main/rtp_engine.c, channels/chan_sip.c, include/asterisk/format.h, configs/codecs.conf.sample: Adds pass-through support for codec CELT. This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ 2011-07-07 19:20 +0000 [r326842] Tilghman Lesher * /, res/res_http_post.c: Merged revisions 326830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326830 | tilghman | 2011-07-07 14:17:19 -0500 (Thu, 07 Jul 2011) | 1 line libgen.h is also needed on Darwin for basename(3) ........ 2011-07-07 17:24 +0000 [r326782] David Vossel * configs/confbridge.conf.sample, apps/confbridge/include/confbridge.h, apps/app_confbridge.c, apps/confbridge/conf_config_parser.c: Updates confbridge.conf video documentation and adds dtmf action for releasing video src. 2011-07-07 16:50 +0000 [r326750] Terry Wilson * utils/astdb2sqlite3.c, main/db.c: Use older functions out of deference to older distros 2011-07-07 16:18 +0000 [r326694] Jonathan Rose * res/res_config_odbc.c, /: Merged revisions 326689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul 2011) | 10 lines res_odbc patch by tilghman to fix integers with null values Addresses some improper sql statements in res_odbc that would cause an update to fail on realtime peers due to trying to set as "(NULL)" rather than an actual NULL. (closes issue #1922STERISK-17791) Reported by: marcelloceschia Patches: 20110505__issue19223.diff.txt uploaded by tilghman (license 14) ........ 2011-07-07 15:28 +0000 [r326682-326684] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 326683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326683 | mnicholson | 2011-07-07 10:28:25 -0500 (Thu, 07 Jul 2011) | 3 lines use sips: or sip: depending on the transport in use when building reply digest URIs ........ * /, channels/chan_sip.c: Merged revisions 326681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326681 | mnicholson | 2011-07-07 10:25:49 -0500 (Thu, 07 Jul 2011) | 3 lines make the uri parameter used in reply digests more standards compliant in certain cases by prepending "sip:" or "sips:" to it ........ 2011-07-07 09:49 +0000 [r326636] Tzafrir Cohen * contrib/scripts/live_ast: live_ast: valgrind: run asterisk under valgrind Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under valgrind. The extra command-line parameters are passed to Asterisk as usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS in live.conf . Review: https://reviewboard.asterisk.org/r/1109/ 2011-07-06 20:58 +0000 [r326589] Terry Wilson * utils/db1-ast/btree/bt_open.c, utils/db1-ast/hash/hash_log2.c, utils/db1-ast/hash/hsearch.c, utils/db1-ast/btree/bt_page.c, utils/db1-ast/hash/page.h, utils/db1-ast/mpool, configure, utils/db1-ast/btree/extern.h, utils/db1-ast/include/db.h, main/db.c, utils/db1-ast/btree/bt_seq.c, utils/db1-ast/recno/recno.h, main/Makefile, utils/db1-ast/btree/bt_utils.c, utils/db1-ast/recno/rec_seq.c, configure.ac, utils/db1-ast/btree/bt_close.c, CHANGES, utils/db1-ast/hash/search.h, utils/db1-ast/hash/README, utils/db1-ast/recno/rec_open.c, utils/db1-ast/hash/hash_bigkey.c, utils/db1-ast/recno/rec_delete.c, Makefile, utils/db1-ast/include, utils/db1-ast/hash/hash_buf.c, utils/db1-ast/db, utils/db1-ast/libdb.map, utils/db1-ast/include/ndbm.h, utils/db1-ast/include/compat.h, utils/db1-ast/mpool/mpool.c, utils/db1-ast/btree/bt_debug.c, main/asterisk.c, utils/db1-ast (added), utils/db1-ast/btree/bt_split.c, utils, utils/db1-ast/recno, utils/db1-ast/btree/bt_delete.c, utils/db1-ast/include/circ-queue.h, tests/test_db.c, utils/db1-ast/Makefile, utils/db1-ast/hash/extern.h, utils/db1-ast/recno/rec_search.c, utils/db1-ast/btree/bt_get.c, utils/db1-ast/hash/hash.c, utils/db1-ast/btree/btree.h, utils/db1-ast/db/db.c, utils/db1-ast/hash/hash.h, utils/db1-ast/include/mpool.h, utils/db1-ast/recno/rec_get.c, utils/db1-ast/hash/hash_func.c, utils/utils.xml, utils/astdb2sqlite3.c (added), utils/db1-ast/btree/bt_overflow.c, UPGRADE.txt, utils/db1-ast/btree/bt_conv.c, utils/db1-ast/btree/bt_search.c, utils/db1-ast/btree/bt_put.c, utils/db1-ast/recno/rec_utils.c, utils/Makefile, utils/db1-ast/hash/hash_page.c, utils/db1-ast/hash, utils/db1-ast/mpool/README, utils/db1-ast/hash/ndbm.c, main/db1-ast (removed), utils/db1-ast/recno/rec_close.c, utils/db1-ast/recno/rec_put.c, utils/db1-ast/recno/extern.h, utils/db1-ast/btree: Replace Berkeley DB with SQLite 3 There were some bugs in the very ancient version of Berkeley DB that Asterisk used. Instead of spending the time tracking down the bugs in the Berkeley code we move to the much better documented SQLite 3. Conversion of the old astdb happens at runtime by running the included astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave identically to the old Berkeley backend, but in the future we could offer a much more robust interface. We do not include the SQLite 3 library in the source tree, but instead rely upon the distribution-provided libraries. SQLite is so ubiquitous that this should not place undue burden on administrators. 2011-07-06 17:39 +0000 [r326485-326544] David Vossel * channels/chan_sip.c: Fixes newlines from being stripped from out of dialog sip MESSAGES. * /, res/res_timing_timerfd.c: Merged revisions 326484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 Jul 2011) | 10 lines Reverts fix for timerfd locking issue. jrose discovered a performance issue with this fix that prevents his analog phones from working when using timerfd as a timing source. Until it is understood what is causing this performance problem, this patch is being reverted. ........ 2011-07-05 22:11 +0000 [r326412] Tilghman Lesher * channels/chan_jingle.c, channels/chan_dahdi.c, funcs/func_speex.c, /, channels/chan_sip.c, codecs/codec_speex.c, funcs/func_aes.c, pbx/pbx_dundi.c, channels/chan_gtalk.c, apps/app_queue.c, channels/chan_iax2.c, res/res_jabber.c, apps/app_stack.c, channels/chan_mgcp.c, apps/app_voicemail.c: Merged revisions 326411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines Add the attribute "type" to each "" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ 2011-07-05 20:25 +0000 [r326368] Kinsey Moore * contrib/scripts/file.convert.sh (added): Prompt conversion script Several variables in the script control which files are converted and the source and destination formats. Patch-by: Trey Blancher (closes AST-560) 2011-07-05 17:35 +0000 [r326321] Richard Mudgett * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 326291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) | 23 lines Used auth= parameter freed during "sip reload" causes crash. If you use the auth= parameter and do a "sip reload" while there is an ongoing call. The peer->auth data points to free'd memory. The patch does several things: 1) Puts the authentication list into an ao2 object for reference counting to fix the reported crash during a SIP reload. 2) Converts the authentication list from open coding to AST list macros. 3) Adds display of the global authentication list in "sip show settings". (closes issue ASTERISK-17939) Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526 ........ 2011-07-05 16:46 +0000 [r326267] Mark Murawki * main/manager.c, CHANGES: New feature: AMI Action FilterAdd This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session (closes issue ASTERISK-16795) Reported by: kobaz Tested by: kobaz,loloski 2011-07-05 13:38 +0000 [r326210] Matthew Jordan * /, main/file.c: Merged revisions 326209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) | 7 lines Updated filestream destructor to block until move is complete when cache is used When a cache directory is used, the process is forked and a mv command is executed to move the temporary file to the permanent location. This caused issues with voicemail, where a race condition occurred when the parent expected the file to be in the permanent location prior to the mv command completing. The parent process is now blocked until the mv command completes. (closes issue ASTERISK-17724) Reported by: Adiren P. Tested by: mjordan ........ 2011-07-01 21:11 +0000 [r326145] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 326144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines Better way to get chan and pvt lock for issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the channels in the defined order and avoid the need for a deadlock avoidance loop. * Lock the channel before getting the pointer to the private structure to be sure that the pointer will not change due to a masquerade or channel hangup. * To preserve sanity, check that chan and p->owner are the same. (Pointer rearangements should not happen without the protection of locks because bad things tend to happen otherwise.) ........ 2011-07-01 16:36 +0000 [r326056-326101] Gregory Nietsky * CHANGES: Change CHANGES move the commits to the right place r296249 r318141 Application changes * CHANGES: Change CHANGES move the commits to the right place in the file missed in review 2011-07-01 12:45 +0000 [r326006] Matthew Nicholson * res/res_fax.c, res/res_fax_spandsp.c: updated irroots info for the authors section 2011-06-30 21:05 +0000 [r325937] David Vossel * channels/chan_bridge.c: Fixes warning message caused by confbridge playback chan not being answered. 2011-06-30 20:47 +0000 [r325936] Richard Mudgett * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 325935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. ........ 2011-06-30 20:33 +0000 [r325931] David Vossel * configs/confbridge.conf.sample, apps/confbridge/include/confbridge.h, include/asterisk/bridging.h, include/asterisk/dsp.h, bridges/bridge_softmix.c, apps/app_confbridge.c, CHANGES, main/bridging.c, main/dsp.c, apps/app_voicemail.c, apps/confbridge/conf_config_parser.c: Video support for ConfBridge. Review: https://reviewboard.asterisk.org/r/1288/ 2011-06-30 20:24 +0000 [r325900] Matthew Jordan * /, apps/app_voicemail.c: Merged revisions 325877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30 Jun 2011) | 9 lines Patched voicemail user option for emailbody / emailsubject Incorporated changes per ASTERISK-16795; updated unit tests to check for vmu->emailbody / vmu->emailsubject (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan ........ 2011-06-30 19:31 +0000 [r325864] Jonathan Rose * /, res/res_musiconhold.c: Merged revisions 325821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun 2011) | 10 lines Fixes an issue with Music on Hold classes losing files in playlist when realtime is used. The bug occurs rather intermittently and I relied on the reporters to test the patch. After a sanity check and some testing, I'm giving it an OK. (closes issue ASTERISK-17875) Reported by: David Cunningham Patches: res_musiconhold.c.mohrt17875_v1 uploaded by Igor Goncharovsky (license #5009) ........ 2011-06-30 18:22 +0000 [r325815-325816] Matthew Nicholson * res/res_fax.c, include/asterisk/res_fax.h, CHANGES, res/res_fax_spandsp.c: Fax gateway functionality (i.e. translating between a T.30 terminal and a T.38 terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the dialplan. Big thanks to irroot for porting this code to use the framehooks api. * main/frame.c: copy all flags on asterisk frames instead of just the timing flag 2011-06-29 21:50 +0000 [r325741] Kinsey Moore * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged revisions 325740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) | 7 lines chan_sip: cleanup from the introduction of ast_str Remove the length field from sip_req and sip_pkt in chan_sip since they are redundant (ast_str holds its own length) and refactor the necessary functions. Review: https://reviewboard.asterisk.org/r/1281/ ........ 2011-06-29 19:02 +0000 [r325674] David Vossel * /, res/res_timing_timerfd.c: Merged revisions 325673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325673 | dvossel | 2011-06-29 13:59:33 -0500 (Wed, 29 Jun 2011) | 6 lines Fixes timerfd locking issue. (closes ASTERISK-17867, ASTERISK-17415) Patches: fix uploaded by kobaz Review: https://reviewboard.asterisk.org/r/1255/ ........ 2011-06-29 18:18 +0000 [r325611-325616] Richard Mudgett * /, apps/app_queue.c: Merged revisions 325614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325614 | rmudgett | 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011) | 5 lines Fixed some error exit cleanup in app_queue.c. * Fixed error exit cleanup in app_queue.c copy_rules() and reload_queue_rules(). ........ * /, apps/app_queue.c: Merged revisions 325610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) | 18 lines Response to QueueRule manager command does not contain ActionID if it was specified. * Add ActionID support as documented for the QueueRule AMI action. * Remove documentation for ActionID with the Queues AMI action. The output does not follow normal AMI response output and there is no place to put an ActionID header. (closes issue AST-602) Reported by: Vlad Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575 ........ 2011-06-29 16:19 +0000 [r325538-325547] Matthew Nicholson * main/channel.c, /: Merged revisions 325545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun 2011) | 2 lines make framehooks prevent native bridging (for real this time) ........ * apps/app_dial.c, main/rtp_engine.c, /: Merged revisions 325537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines don't do native/remote bridging if a framehook is active on the channel ........ 2011-06-29 06:39 +0000 [r325483] Gregory Nietsky * configs/queues.conf.sample, UPGRADE.txt, CHANGES, apps/app_queue.c: Commit "distrotech" app_queue changes to Trunk * Added general option negative_penalty_invalid default off. when set members are seen as invalid/logged out when there penalty is negative. for realtime members when set remove from queue will set penalty to -1. * Added queue option autopausedelay when autopause is enabled it will be delayed for this number of seconds since last successful call if there was no prior call the agent will be autopaused immediately. * Added member option ignorebusy this when set and ringinuse is not will allow per member control of multiple calls as ringinuse does for the Queue. - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty. (closes issue ASTERISK-17421) (closes issue ASTERISK-17391) Reported by: irroot Tested by: irroot, jrose Review: https://reviewboard.asterisk.org/r/1119/ 2011-06-28 21:51 +0000 [r325417] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 325416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines Fix random misspelling noticed on asterisk-users. ........ 2011-06-28 20:32 +0000 [r325345] David Vossel * /, channels/chan_sip.c: Merged revisions 325339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) | 4 lines Fixes locking inversion caused by holding sip pvt lock during async_goto. (closes ASTERISK-17352) ........ 2011-06-28 17:38 +0000 [r325213] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 325212 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines Use the device name and not the channel name to initialize the device state. Correct ASTERISK-11323 implementation as I don't see how it ever worked as claimed when it used the channel name and not the device name. (issue ASTERISK-11323) ........ 2011-06-28 16:04 +0000 [r325153] Jonathan Rose * /, res/res_musiconhold.c: Merged revisions 325152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325152 | jrose | 2011-06-28 10:46:29 -0500 (Tue, 28 Jun 2011) | 5 lines Fixes moh reload breaking custom mode moh classes when the config file is untouched (closes issue ASTERISK-17730) Reported by: sdolloff ........ 2011-06-28 15:34 +0000 [r325151] David Vossel * channels/chan_sip.c: Fixes issue with video and text not being reinvited correctly with directmedia If a SDP does not modify the session, we ignore it. However, we were defaulting no text and video support to true before checking to see if the sdp modified anything or not. This would result in process_sdp ignoring an sdp but removing video and text from the call during direct media reinvites. 2011-06-28 15:12 +0000 [r325092] Leif Madsen * /, build_tools/prep_tarball: Merged revisions 325091 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325091 | lmadsen | 2011-06-28 10:12:00 -0500 (Tue, 28 Jun 2011) | 1 line Remove line from prep_tarball that kills mkrelease. ........ 2011-06-28 00:07 +0000 [r325046] Terry Wilson * channels/chan_sip.c: Don't forget to build the Via when sending MESSAGE 2011-06-27 16:32 +0000 [r324961] Tilghman Lesher * /, main/asterisk.c: Merged revisions 324955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) | 5 lines Save and restore errno from within signal handlers. This is recommended by the POSIX standard, as well as by the sigaction(2) manpage for various platforms that we support (e.g. Mac OS X). ........ 2011-06-27 15:38 +0000 [r324915] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 324914 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines When subscribing MWI to an unsolicited mailbox the first notification is incorrect. A remote peer subscribed to MWI with the unsolicited option and a local phone subscribed to the remote mailbox. The notify message-summary events are sent correctly except for the first one when subscribing, which will always be 0. This means the phone MWI indicator will be wrong until the mailbox read/unread count changes and the event is fired. Looks like this is a regression from ASTERISK-16149. * Fix the logic to check the cache and if allowed then fallback to manually counting mailbox messages. (closes issue ASTERISK-17997) Reported by: rsw686 Patches: jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett Tested by: rsw686 JIRA SWP-3551 ........ 2011-06-24 20:50 +0000 [r324850] Richard Mudgett * /, pbx/pbx_config.c: Merged revisions 324849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011) | 15 lines Syntax errors in dialplan do not display the file name. When issuing the CLI command "dialplan reload" syntax errors and warnings are displayed on the console. The offending line number is displayed on the console, but the file name is not displayed. Errors caught in main/config.c do display the file name. (closes issue ASTERISK-17985) Reported by: ulogic Patches: pbx_config.patch uploaded by ulogic (License #5685) modified format Tested by: rmudgett JIRA SWP-3554 ........ 2011-06-24 16:50 +0000 [r324769] Jonathan Rose * include/asterisk/logger.h, /: Merged revisions 324768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | 11 lines DTMF wasn't being logged on connected consoles when enabled in logger.conf Previously in order for DTMF to be logged in a connected console session, the user would have to do logger set channel DTMF on. This corrects that so that it is on by default. This issue was caused by an off by one error incurred by a logger level count of 6 in logger.h where it should have been 7. (closes issue: ASTERISK-17974) Reported by: Luke H ........ 2011-06-23 18:56 +0000 [r324708-324709] Kinsey Moore * apps/app_confbridge.c: ConfBridge: redundant code cleanup There is no reason to clean up features twice. Review: https://reviewboard.asterisk.org/r/1279/ * /, channels/chan_sip.c: Merged revisions 324678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324678 | kmoore | 2011-06-23 13:29:17 -0500 (Thu, 23 Jun 2011) | 11 lines Merged revisions 324643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | 4 lines Addresses AST-2011-008, memory corruption and remote crash in SIP driver. AST-2011-008 ........ ................ 2011-06-23 18:31 +0000 [r324664-324689] David Vossel * /, channels/sip/reqresp_parser.c: Merged revisions 324685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) | 8 lines Fixes sip crash when calling remove_uri_parameters with NULL AST-2011-009 (closes issue ASTERISK-18017) Reported by: jaredmauch ........ * /, main/features.c, channels/chan_iax2.c, include/asterisk/frame.h: Merged revisions 324652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines 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 ........ ................ ................ 2011-06-23 03:16 +0000 [r324558] Terry Wilson * /, tests/test_netsock2.c: Merged revisions 324557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 Jun 2011) | 5 lines Remove tests for parsing address with invalid port getaddrinfo on OS X returns with EAI_NONAME error when passed a port greater than 65535. Linux throws no error, so remove the tests for now. ........ 2011-06-22 19:17 +0000 [r324495] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 324491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) | 1 line Use correct variable for text SRTP media. ........ 2011-06-22 19:12 +0000 [r324487] Terry Wilson * main/netsock2.c, /, channels/chan_sip.c, include/asterisk/netsock2.h, tests/test_netsock2.c (added): Merged revisions 324484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ ........ 2011-06-22 18:45 +0000 [r324480-324482] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 324481 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 Also fixed a reference leak in an error path in sip_msg_send(). ........ r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines Timout or error on INFO or MESSAGE transaction causes call to be lost. When exchanging INFO messages within a call, 4xx error causes the call to be disconnected although RFC 2976 explicitly states that such transactions do not modify the state of the dialog. When exchanging MESSAGE messages within a call, 4xx error causes the call to be disconnected. To provide least surprise, we should not disconnect the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428 Section 2) (closes issue ASTERISK-17901) Reported by: neutrino88 Review: https://reviewboard.asterisk.org/r/1257/ Review: https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486 ........ * /, channels/chan_sip.c: Merged revisions 324479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) | 1 line Comments and whitespace in chan_sip.c ........ 2011-06-21 21:55 +0000 [r324365-324422] David Vossel * apps/app_confbridge.c: Fixes issue with channel write format being incorrectly restored when MOH is used in confbridge. * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 324364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) | 10 lines Fixes locking inversion issue in ast_async_goto() During this function we can not hold the "chan" lock while doing the masquerade, the explicit goto on the tmp chan, or the channel alloc. Instead we need to get the channel lock, store off information about the channel that we need, and then let the channel lock go for the remainder of the function. Review: https://reviewboard.asterisk.org/r/1275/ ........ 2011-06-21 16:06 +0000 [r324304] 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/ 2011-06-21 15:49 +0000 [r324302] David Vossel * channels/chan_sip.c: Fixes issue with finding correct extension when message context is used. 2011-06-20 18:13 +0000 [r324242] Leif Madsen * /, configs/queuerules.conf.sample: Merged revisions 324241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) | 2 lines Remove extra 'the'. Reported by Vlad Povorozniuc ........ 2011-06-20 17:34 +0000 [r324238] Terry Wilson * /, channels/chan_sip.c: Merged revisions 324237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) | 12 lines Ignore media offers with a port of 0 Section 5.1 of RFC3264 states: A port number of zero in the offer indicates that the stream is offered but MUST NOT be used. (closes issue ASTERISK-17845) Reported by: jacco Patches: issue19281_2.patch uploaded by jacco (license 1277) Tested by: jacco, twilson ........ 2011-06-17 18:52 +0000 [r324177-324179] Leif Madsen * main/manager.c, /: Merged revisions 324178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324178 | lmadsen | 2011-06-17 14:51:16 -0400 (Fri, 17 Jun 2011) | 2 lines Add Username and Secret fields to manager Login action. Pointed out by Vlad Povorozniuc ........ * /, apps/app_meetme.c: Merged revisions 324176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) | 2 lines Fix typo in documentation. Pointed out by Vlad Povorozniuc ........ 2011-06-17 18:23 +0000 [r324175] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 324174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 Jun 2011) | 5 lines Add header string to libpri debug output. Add header string to libpri debug output so the libpri output can be found/extracted easier from huge debug trace files. ........ 2011-06-17 15:32 +0000 [r324131] Leif Madsen * main/pbx.c, /: Merged revisions 324115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324115 | lmadsen | 2011-06-17 11:14:54 -0400 (Fri, 17 Jun 2011) | 3 lines Fix grammar in documentation for Goto() and GotoIf() (closes issue ASTERISK-18023) Reported by: Tim Osman ........ 2011-06-16 22:49 +0000 [r324050] Terry Wilson * main/channel.c, channels/chan_local.c, /, channels/chan_sip.c, include/asterisk/channel.h: Merged revisions 324048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ 2011-06-16 18:13 +0000 [r323991] Richard Mudgett * /, tests/test_event.c: Merged revisions 323990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323990 | rmudgett | 2011-06-16 13:12:32 -0500 (Thu, 16 Jun 2011) | 5 lines The test_event unit test is occasionally failing. Wait for the special posted event to process before adding a new subscription. ........ 2011-06-16 15:59 +0000 [r323673-323933] Terry Wilson * Makefile, /: Merged revisions 323932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323932 | twilson | 2011-06-16 10:58:22 -0500 (Thu, 16 Jun 2011) | 4 lines Don't assume ASTDBDIR exists It most likely doesn't on FreeBSD ........ * /, tests/test_db.c: Merged revisions 323866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323866 | twilson | 2011-06-15 15:03:58 -0500 (Wed, 15 Jun 2011) | 2 lines Remove now-useless cast of ARRAY_LEN ........ * include/asterisk/utils.h, /: Merged revisions 323863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323863 | twilson | 2011-06-15 14:58:18 -0500 (Wed, 15 Jun 2011) | 2 lines Make ARRAY_LEN() return the same type on x86 and x86_64 systems ........ * /, tests/test_db.c: Merged revisions 323859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323859 | twilson | 2011-06-15 14:45:20 -0500 (Wed, 15 Jun 2011) | 2 lines Fix more ARRAY_LEN format string issues ........ * /, main/features.c: Merged revisions 323754 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323754 | twilson | 2011-06-15 13:21:52 -0500 (Wed, 15 Jun 2011) | 23 lines Merged revisions 323733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323733 | twilson | 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) | 16 lines 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 ........ ................ ................ * /, tests/test_db.c: Merged revisions 323672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323672 | twilson | 2011-06-15 10:09:51 -0700 (Wed, 15 Jun 2011) | 5 lines Cast ARRAY_LEN to size_t for ast_logging 32-bit and 64-bit machines return different types for ARRAY_LEN(), so cast it before using in a format string. ........ 2011-06-15 16:49 +0000 [r323671] Richard Mudgett * /, tests/test_event.c, main/event.c: Merged revisions 323669-323670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) | 21 lines [regression] Voicemail MWI is no longer sent. When leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read. If you restart Asterisk, everything comes up at that state correctly, but changes to the messages in voicemail causes the light to not be set appropriately. Very easy to reproduce. * Made ast_event_check_subscriber() return TRUE if there are ANY subscribers to an event type when there are no restricting ie values passed. This allows an event being queued to be queued. (closes issue ASTERISK-18002) Reported by: lmadsen Tested by: lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621) (closes issue ASTERISK-18019) ........ r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 Jun 2011) | 7 lines Add a test to the event unit tests to catch ASTERISK-18002. The new tests check to see if there are ANY subscribers to the event type when ast_event_check_subscriber() is not passed any specific ie values. (issue ASTERISK-18002) ........ 2011-06-15 16:19 +0000 [r323621] Jonathan Rose * res/res_config_pgsql.c, /: Merged revisions 323610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323610 | jrose | 2011-06-15 11:09:24 -0500 (Wed, 15 Jun 2011) | 7 lines Adds PQclear calls on result to various parts of res_conf_pgsql (closes issue ASTERISK-17812) Reported by: byronclark Patches: pgsql_pqclear.patch uploaded by byronclark (license 1200) ........ 2011-06-15 15:33 +0000 [r323609] Sean Bright * main/manager.c, /: Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines 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) ........ ................ ................ 2011-06-15 13:45 +0000 [r323517] Kinsey Moore * apps/app_confbridge.c, CHANGES: CONFBRIDGE_INFO function to get conference data Added the CONFBRIDGE_INFO dialplan function to get information about a conference bridge including locked status and number of parties, admins, and marked users. Review: https://reviewboard.asterisk.org/r/1271/ 2011-06-15 00:51 +0000 [r323397-323457] Richard Mudgett * /, main/event.c: Merged revisions 323456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323456 | rmudgett | 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011) | 1 line Add missing break in ast_event_get_cached(). ........ * main/netsock2.c, main/dnsmgr.c, /: Merged revisions 323392,323394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323392 | rmudgett | 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011) | 6 lines Add more strict hostname checking to ast_dnsmgr_lookup(). Change suggested in review. Review: https://reviewboard.asterisk.org/r/1240/ ........ r323394 | rmudgett | 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) | 2 lines Made ast_sockaddr_split_hostport() port warning msgs more meaningful. ........ 2011-06-14 17:03 +0000 [r323374] Terry Wilson * res/res_rtp_asterisk.c, main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged revisions 323370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ 2011-06-14 16:47 +0000 [r323372] Jonathan Rose * /, channels/chan_sip.c: Merged revisions 323371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this was causing NAT=Yes to always use rport when present which was against 1.6.2 behavior and the check itself was redundant since the only way this segment of code could be reached was if RPORT_PRESENT was already evaluated as true earlier. (closes issue ASTERISK-17789) Reported by: byronclark Patches: use_sip_nat_force_rport.patch uploaded by byronclark (license 1200) ........ 2011-06-14 14:37 +0000 [r323325] David Vossel * channels/chan_sip.c: Store sip peer name as var data on a outofcall msg. 2011-06-13 20:44 +0000 [r323272] Kinsey Moore * apps/confbridge/conf_config_parser.c: Config inheritance doesn't work with ConfBridge() menu definitions Current behavior in ConfBridge menu definitions is that first definition takes precedence, even in templated situations. This change allows inheritance and overriding to work as expected so that the last definition takes precedence. (closes ASTERISK-17986) Review: https://reviewboard.asterisk.org/r/1267/ 2011-06-13 19:54 +0000 [r323214] Leif Madsen * main/channel.c, /: Merged revisions 323213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323213 | lmadsen | 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011) | 6 lines Avoid dividing by zero with L() option to Dial() Reported by: nicolasom Patches: issue-17995.patch - nicolasom (License #5994) ........ 2011-06-13 19:43 +0000 [r323212] David Vossel * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: Addition of "outofcall_message_context" sip.conf option. Review: https://reviewboard.asterisk.org/r/1265/ 2011-06-13 19:03 +0000 [r323155] Leif Madsen * /, res/res_agi.c: Merged revisions 323154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323154 | lmadsen | 2011-06-13 15:00:41 -0400 (Mon, 13 Jun 2011) | 6 lines Tweak documentation for AGI Hangup command. (closes issue ASTERISK-17999) Reported by: Ben Klang Patches: hangup-doc.diff - uploaded by Ben Klang (License #5876) ........ 2011-06-13 14:38 +0000 [r323106-323107] Kinsey Moore * apps/confbridge/include/confbridge.h, apps/app_confbridge.c: MOH for only user not working with ConfBridge This adds the playing_moh flag to the conference_bridge_user struct that signifies when MOH should be playing so code doesn't have to guess whether MOH is playing. This change also adds the necessary checking to ensure that MOH continues playing for a single user in a conference after the join sound is played when configured to do so. (closes ASTERISK-17988) Review: https://reviewboard.asterisk.org/r/1263/ * apps/app_confbridge.c: ConfBridge: Use of bridge or user profiles that don't exist Bridge and user profiles are not checked for existence before use. The lack of a fully formed bridge profile can cause a segfault when sounds are accessed. This change ensures that bridge and user profiles exist prior to usage attempts. Review: https://reviewboard.asterisk.org/r/1264/ 2011-06-10 19:22 +0000 [r323041] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 323040 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) | 5 lines 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 ........ 2011-06-10 15:30 +0000 [r322866-322982] Terry Wilson * /, main/db.c: Merged revisions 322981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) | 11 lines Avoid a DB1 infinite loop bug Explicity check the last entry in the DB and make sure that we don't iterate past it. Since there can be no duplicates, this just makes sure that we stop after matching the last key. This patch also refactors the code to get away from some code duplication. A previous patch added many astdb tests and this patch passed them. Review: https://reviewboard.asterisk.org/r/1259/ ........ * /, tests/test_db.c (added): Merged revisions 322923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322923 | twilson | 2011-06-09 19:33:23 -0700 (Thu, 09 Jun 2011) | 2 lines Add some astdb unit tests ........ * /, include/asterisk/astdb.h: Merged revisions 322865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322865 | twilson | 2011-06-09 15:29:20 -0700 (Thu, 09 Jun 2011) | 4 lines Correct ast_db_deltree documentation ast_db_deltree returns -1 on error, otherwise the number of deletions ........ 2011-06-09 17:43 +0000 [r322808] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 322807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322807 | mnicholson | 2011-06-09 12:37:07 -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 ........ 2011-06-09 16:47 +0000 [r322750] Richard Mudgett * /, apps/app_directed_pickup.c, main/features.c, include/asterisk/features.h: Merged revisions 322749 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines Remove potential deadlock in call pickup race. Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ ........ 2011-06-09 11:05 +0000 [r322544] Damien Wedhorn * channels/chan_skinny.c: Add autoanswer to skinny. Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER. Initial value must be the time to autoanswer in ms, then optionally :BEEP to play a tone when answered and :MUTE to mute the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and play a beep. just 3000 would answer afer 3 secs of ringing with no beep and full two way audio. 2011-06-08 20:48 +0000 [r322426-322485] Richard Mudgett * /, apps/app_queue.c: Merged revisions 322484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) | 15 lines Ring all queue with more than 255 agents will cause crash. 1. Create a ring-all queue with 500 permanent agents. 2. Call it. 3. Asterisk will crash. The watchers array in app_queue.c has a hard limit of 255. Bounds checking is not done on this array. No sane person should put 255 people in a ring-all queue, but we should not crash anyway. * Added bounds checking to the watchers array. JIRA AST-464 JIRA SWP-2903 ........ * main/dnsmgr.c, /: Merged revisions 322425 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) | 16 lines SRV lookup attempted for SIP peers listed as an IP address. Asterisk attempts to SRV lookup a host name even if the host name is an IP address. Regression introduced when IPv6 support was added. * Restored the check in ast_dnsmgr_lookup() to see if the given host name is an IP address. The IP address could be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815) Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621) Review: https://reviewboard.asterisk.org/r/1240/ ........ 2011-06-08 11:38 +0000 [r322381] Damien Wedhorn * channels/chan_skinny.c: Remove skinny do_monitor and use ast_sched_start instead The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything. Review: https://reviewboard.asterisk.org/r/1256/ 2011-06-08 06:45 +0000 [r322323] Gregory Nietsky * /, channels/chan_sip.c: Merged revisions 322322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines Make handle_request_publish do dialog expiration and destruction. This patch fixes handle_request_publish so that it does dialog expiration and destruction. Without this patch the incoming PUBLISH requests will get stuck in the dialog list. Restarting asterisk is the only way to remove them. Personal observation on one system the server hung up while looping through the channels rendering asterisk unusable and all sip phones unregisterd when they try reregister more requests are added. (closes issue #18898) Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review: https://reviewboard.asterisk.org/r/1253 ........ 2011-06-07 23:14 +0000 [r322284] Richard Mudgett * channels/chan_sip.c, include/asterisk/message.h: Correct some whitespace and a reference debug message. 2011-06-07 19:17 +0000 [r322244] Russell Bryant * res/res_jabber.c: Actually check the "sendtodialplan" option setting for xmpp. (closes issue ASTERISK-17978) Reported by: elguero Patches: stop_messages_going_to_dialplan.patch (license #5026) 2011-06-07 18:01 +0000 [r322190] Paul Belanger * configs/sip_notify.conf.sample, /: Merged revisions 322189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun 2011) | 4 lines Use correct syntax for 'sip notify snom-reboot' (closes issue ASTERISK-17915) ........ 2011-06-06 19:39 +0000 [r322111-322128] Gregory Nietsky * apps/app_queue.c: Remove Unused Var Warning rt_handle_member_record * apps/app_queue.c: Refactor rt_handle_member_record Review: https://reviewboard.asterisk.org/r/1172 2011-06-06 19:15 +0000 [r322070] Jonathan Rose * include/asterisk/logger.h, /, main/asterisk.c: Merged revisions 322069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) | 8 lines Fixes level toggling for logger set levels since it was reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H Review: https://reviewboard.asterisk.org/r/1244/ ........ 2011-06-03 22:15 +0000 [r321814-321927] Richard Mudgett * cel/cel_radius.c, /, cdr/cdr_radius.c: Merged revisions 321926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011) | 18 lines Asterisk crash when unloading cdr_radius/cel_radius. The rc_openlog() API call is passed a string that is used by openlog() to format log messages. The openlog() does not copy the string it just keeps a pointer to it. When the module is unloaded, the string is gone from memory. Depending upon module load order and if the other module then has an error, a crash happens. * Pass rc_openlog() a strdup'd string with the understanding that there will be a small memory leak if the cdr_radius/cel_radius modules are unloaded. * Call rc_destroy() to free the rc handle memory when the module is unloaded. JIRA AST-483 JIRA SWP-3062 ........ * /, main/ccss.c: Merged revisions 321924 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) | 5 lines Be more explicit for CCSS generic device state event subscription. Make CCSS generic device state event subscription specify the AST_EVENT_IE_STATE ie exists to be safe. ........ * /, tests/test_event.c, main/event.c: Merged revisions 321871 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) | 27 lines Event subscription fixes. Must commit the subscription fixes together with the integration subscription tests. The subscription fixes cause an erroneously passing test to fail. The new subscription tests detect errors without the subscription fixes. * Added missing event_names[] table entry. * Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to correctly detect if a subscriber exists for the proposed event. * Made match_ie_val() and match_sub_ie_val_to_event() check the buffer length for RAW payload types. * Fixed error handling memory leak in ast_event_sub_activate(), ast_event_unsubscribe(), and ast_event_queue(). * Made ast_event_new() and ast_event_check_subscriber() better protect themselves from an invalid payload type. * Added container lock protection between removing old cache events and adding the new cached event in ast_event_queue_and_cache()/event_update_cache(). * Added new event subscription tests. ........ * include/asterisk/event.h, /, channels/chan_sip.c, main/event.c, channels/chan_iax2.c: Merged revisions 321812-321813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) | 1 line Correct IAX2 and SIP event subscription description string. ........ r321813 | rmudgett | 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription description parameter string. ........ 2011-06-03 18:25 +0000 [r321752] Russell Bryant * tests/test_astobj2.c, main/astobj2.c: Fix some astobj2 iterator breakage, add another unit test. Review: https://reviewboard.asterisk.org/r/1254/ 2011-06-03 13:18 +0000 [r321689] Leif Madsen * /, configs/queues.conf.sample: Merged revisions 321685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) | 5 lines Also document the 'queue-minute' option. (closes issue #19386) Reported by: juanmol ........ 2011-06-02 22:09 +0000 [r321617] Russell Bryant * channels/chan_sip.c: Fix message destination extension. Don't send all messages to 's'. Get the destination from the request URI. (Found using automated test cases). 2011-06-01 23:12 +0000 [r321548] Richard Mudgett * main/cdr.c, /: Merged revisions 321547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321547 | rmudgett | 2011-06-01 18:11:55 -0500 (Wed, 01 Jun 2011) | 1 line CDR comment tweaks. ........ 2011-06-01 21:31 +0000 [r321546] Russell Bryant * main/channel.c, channels/chan_sip.c, configs/jabber.conf.sample, include/asterisk/message.h (added), include/asterisk/jabber.h, include/asterisk/channel.h, configs/sip.conf.sample, include/asterisk/_private.h, CHANGES, res/res_jabber.c, main/message.c (added), channels/sip/include/sip.h, main/asterisk.c: Support routing text messages outside of a call. Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ 2011-06-01 20:11 +0000 [r321538] Brett Bryant * /, apps/app_voicemail.c: Merged revisions 321537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01 Jun 2011) | 8 lines This patch fixes an issue with using the wrong voicemail folders with greetings. (closes issue #17871) Reported by: edhorton Patches: digium_bug_17871_2 uploaded by fhackenberger (license 592) Tested by: edhorton, fhackenberger ........ 2011-06-01 10:45 +0000 [r321529] Alexandr Anikin * addons/chan_ooh323.c, /, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/oochannels.c: Merged revisions 321528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines Fix double alerting, add forced alerting before answer Fix double alerting (it wasn't fixed here by issue #18542) Add forced alerting before connect (if it wasn't before) Try to send all packets from outgoing queue rather than one only Call goes into clearing state when disconnect command is received (closes issue #19361) Reported by: vmikhelson Patches: issue19361-3.patch uploaded by may213 (license 454) Tested by: vmikhelson ........ 2011-05-31 20:55 +0000 [r321518] Richard Mudgett * include/asterisk/acl.h, /, include/asterisk/dnsmgr.h: Merged revisions 321517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321517 | rmudgett | 2011-05-31 15:54:35 -0500 (Tue, 31 May 2011) | 1 line Update some comments. ........ 2011-05-31 19:01 +0000 [r321516] David Vossel * channels/chan_local.c, /: Merged revisions 321515 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 May 2011) | 12 lines Chan_local locking cleanup. This patch removes all of the unnecessary deadlock avoidance loops that occur in chan_local. It also resolves an issue with a deadlock triggered by local channel optimizations. (issue #18028) Review: https://reviewboard.asterisk.org/r/1231/ ........ 2011-05-31 16:06 +0000 [r321512] Leif Madsen * /, channels/chan_sip.c: Merged revisions 321511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) | 8 lines Enhance NOTICE message to know who couldn't access the dialplan. (closes issue #19390) Reported by: lmadsen Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10) Tested by: russell ........ 2011-05-28 00:29 +0000 [r321338-321445] Richard Mudgett * /, res/res_agi.c: Merged revisions 321436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321436 | rmudgett | 2011-05-27 19:27:52 -0500 (Fri, 27 May 2011) | 4 lines Some hagi launch cleanup. Inspired by issue 19256. This patch would also fix the crash. ........ * main/srv.c, /: Merged revisions 321392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) | 12 lines Crash when using hagi and no servers are available. When none of the servers returned by the SRV querey respond, asterisk crashes. The problem is that if the loop over all the SRV entries finishes then the srv_context has already been cleaned up. * Make ast_srv_cleanup() check to see if the context is already cleaned up. (closes issue #19256) Reported by: byronclark ........ * /, apps/app_privacy.c, UPGRADE.txt, CHANGES: Merged revisions 321337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert -r321331 and -r321332. ........ r321337 | rmudgett | 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. (closes issue #19273) Reported by: mdavenport ........ 2011-05-27 21:40 +0000 [r321334] Leif Madsen * /, main/features.c: Merged revisions 321333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321333 | lmadsen | 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011) | 7 lines Allow parking lot hints and musicclass to be set. (closes issue #19378) Reported by: sboily_proformatique Patches: pf_parkinghint_music_fix uploaded by sboily proformatique (license 206) Tested by: russell ........ 2011-05-27 21:37 +0000 [r321331-321332] Richard Mudgett * UPGRADE.txt: Add note about PrivacyManager to UPGRADE.txt * /, apps/app_privacy.c, CHANGES: Merged revisions 321330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. The trunk(v1.10) version will remove the unused options position. (closes issue #19273) Reported by: mdavenport ........ 2011-05-27 16:35 +0000 [r321289] Jonathan Rose * /, channels/sip/reqresp_parser.c: Merged revisions 321273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | 3 lines markm committed a patch I was working on yesterday, this fixes it to mesh up with suggestions by mnicholson. ........ 2011-05-27 08:37 +0000 [r321212] Alec L Davis * /, main/features.c: Merged revisions 321211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May 2011) | 16 lines 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: review1232-1.8.diff.txt alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1232/ ........ 2011-05-26 21:50 +0000 [r321101-321156] Mark Murawki * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged revisions 321155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | 10 lines Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic. Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails. (closes issue #19346) Reported by: kobaz Tested by: kobaz,JonathanRose Review: [full review board URL with trailing slash] ........ * main/netsock2.c, /, channels/chan_sip.c: Merged revisions 321100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | 11 lines ast_sockaddr_resolve() in netsock2.c may deref a null pointer Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables (closes issue #19346) Reported by: kobaz Patches: netsock2.patch uploaded by kobaz (license 834) Tested by: kobaz, Marquis ........ 2011-05-26 18:10 +0000 [r321045] Richard Mudgett * /, include/asterisk/netsock2.h: Merged revisions 321044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321044 | rmudgett | 2011-05-26 13:10:17 -0500 (Thu, 26 May 2011) | 1 line Update ast_sockaddr comment with an important note. ........ 2011-05-26 17:35 +0000 [r321043] Terry Wilson * main/rtp_engine.c, /: Merged revisions 321042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321042 | twilson | 2011-05-26 10:29:54 -0700 (Thu, 26 May 2011) | 6 lines Initialize stack-allocated ast_sockaddrs before use It is important to always initialize ast_sockaddrs before use--even if they are passed to ast_sockaddr_copy as the underlying storage could be bigger than what ends up being copied--leaving part of the data unitialized. ........ 2011-05-26 16:54 +0000 [r321003] Russell Bryant * /, channels/chan_alsa.c: Merged revisions 320947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320947 | russell | 2011-05-26 10:57:13 -0500 (Thu, 26 May 2011) | 2 lines Remove some variables that were set but unused. ........ 2011-05-26 15:55 +0000 [r320946] Terry Wilson * main/channel.c, main/utils.c, include/asterisk/stringfields.h: Use va_copy for stringfields The ast_string_field_build_va functions were written to take to separate va_lists to work around FreeBSD 4 not having va_copy defined. In the end, we don't support anything using gcc < 3 anyway because we use va_copy all over the place anyway. This patch just simplifies things by removing the second va_list function arguments in favor of va_copy. Review: https://reviewboard.asterisk.org/r/1233/ --This line, and those below, will be ignored-- M include/asterisk/stringfields.h M main/utils.c M main/channel.c 2011-05-25 22:28 +0000 [r320820-320884] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 320883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) | 17 lines Native SIP CCSS sends bad CC cancel SUBSCRIBE message. The SUBSCRIBE message used to cancel a CC request has incorrect To/From SIP headers. They are reversed and the dialog tags are the same when they should not be. If pedantic mode was disabled, then the cancel would have succeeded despite the incorrect message. * The SIP_OUTGOING flag was not set correctly for the dialog and I had to move some CC subscribe handling code as a result. * Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a CC request SUBSCRIBE message comes in and the CC instance is not found, the 404 response was duplicated. JIRA AST-568 JIRA SWP-3493 ........ * apps/app_dial.c, main/channel.c, main/manager.c, /, apps/app_meetme.c, apps/app_fax.c, main/features.c, CHANGES, apps/app_queue.c, UPGRADE-1.8.txt: Merged revisions 320823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ * main/channel.c, /, main/format_cap.c, main/features.c, include/asterisk/channel.h: Merged revisions 320796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25 May 2011) | 17 lines Give zombies a safe channel driver to use. Recent crashes from zombie channels suggests that they need a safe home to goto. When a masquerade happens, the physical part of the zombie channel is hungup. The hangup normally sets the channel private pointer to NULL. If someone then blindly does a callback to the channel driver, a crash is likely because the private pointer is NULL. The masquerade now sets the channel technology of zombie channels to the kill channel driver. Related to the following issues: (issue #19116) (issue #19310) Review: https://reviewboard.asterisk.org/r/1224/ ........ 2011-05-25 15:43 +0000 [r320772] Gregory Nietsky * funcs/func_channel.c, CHANGES: CHANNEL(pickupgroup) Allow Setting / Reading the pickupgroup of a channel with func_channel.c (closes issue #19045) Reported by: irroot Review: https://reviewboard.asterisk.org/r/1148/ 2011-05-25 00:52 +0000 [r320717] Terry Wilson * /, addons/chan_mobile.c: Merged revisions 320716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320716 | twilson | 2011-05-24 17:49:10 -0700 (Tue, 24 May 2011) | 4 lines Cast data as char * before using S_OR This is required for compiling successfully under dev mode ........ 2011-05-23 18:00 +0000 [r320651] Richard Mudgett * main/manager.c, /, CHANGES: Merged revisions 320650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 May 2011) | 16 lines Add ConnectedLineNum/Name headers to output of AMI action Status. * Add ConnectedLineNum and ConnectedLineName headers to the output of the AMI action Status. This makes it easier to find out who the channel is connected to without having to lookup BridgedChannel or when they are connected to an application (e.g.: VoiceMail) which has no bridged channel. * Bridged channels with no CallerID had "" instead of "" output, that might be a bug as "" was what older versions used. (closes issue #18158) Reported by: gareth Patches: svn-292308.diff uploaded by gareth (license 208) ........ 2011-05-23 16:28 +0000 [r320606] David Vossel * main/tcptls.c, /: Merged revisions 320568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320568 | dvossel | 2011-05-23 11:18:33 -0500 (Mon, 23 May 2011) | 14 lines Merged revisions 320562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011) | 9 lines 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) ........ ................ 2011-05-23 16:20 +0000 [r320579] Tilghman Lesher * /, configure, configure.ac: Merged revisions 320573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320573 | tilghman | 2011-05-23 11:19:32 -0500 (Mon, 23 May 2011) | 7 lines GNU libiconv uses symbol "libiconv_open" instead of "iconv_open". (closes issue #19344) Reported by: rohanl Patches: iconv-check.patch uploaded by rohanl (license 1284) ........ 2011-05-23 15:48 +0000 [r320561] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 320560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320560 | kpfleming | 2011-05-23 10:47:14 -0500 (Mon, 23 May 2011) | 4 lines Don't generate spurious "No: command not found" messages when running the configure script on a system that has neither gmime-config nor pkg-config. ........ 2011-05-23 14:40 +0000 [r320505] Jonathan Rose * /, channels/chan_sip.c: Merged revisions 320504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320504 | jrose | 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) | 10 lines Fixes segfault occuring in chan_sip.c at __set_address_from_contact Checks to see if domain contains anything before sending it off to ast_sockaddr_resolve which is where the segfault was occuring due to null str. (closes issue #18857) Reported by: sybasesql Review: https://reviewboard.asterisk.org/r/1225/ ........ 2011-05-22 23:36 +0000 [r320446] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 320445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320445 | tilghman | 2011-05-22 18:34:57 -0500 (Sun, 22 May 2011) | 15 lines Merged revisions 320444 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011) | 8 lines 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 21:40 +0000 [r320340] David Vossel * main/tcptls.c, /: Merged revisions 320338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320338 | dvossel | 2011-05-20 16:39:36 -0500 (Fri, 20 May 2011) | 14 lines Merged revisions 320271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011) | 8 lines 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 20:53 +0000 [r320238] Richard Mudgett * /, apps/app_meetme.c: Merged revisions 320237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320237 | rmudgett | 2011-05-20 15:49:03 -0500 (Fri, 20 May 2011) | 27 lines Merged revisions 320236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500 (Fri, 20 May 2011) | 20 lines 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 18:49 +0000 [r320181] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 320180 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May 2011) | 16 lines 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 ........ 2011-05-20 18:29 +0000 [r320178] Jonathan Rose * /, apps/app_voicemail.c: Merged revisions 320162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May 2011) | 15 lines Fixes an imapfolder related crash imapfolders being set in the general section of voicemail would cause the inbox folder name to change. Since sound file names are made based on the names of the folders, this would cause the audio related to that folder name to change and if Asterisk attempted to play it, the channel would instantly hang up when the audio file couldn't be found. This patch searches for the name of the folder first to leave existing behavior in tact and if that fails, it uses the normal inbox name to get the sound file instead. (closes issue #16104) Reported by: blkline Review: https://reviewboard.asterisk.org/r/1215/ ........ 2011-05-20 17:04 +0000 [r320058-320060] Richard Mudgett * /, main/features.c: Merged revisions 320059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320059 | rmudgett | 2011-05-20 12:03:49 -0500 (Fri, 20 May 2011) | 1 line Misc comment cleanup in features.c. ........ * main/channel.c, /, main/features.c: Merged revisions 320057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320057 | rmudgett | 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011) | 19 lines Crash while transferring a call during DTMF feature timeout. When a call is being attended transferred during the time between AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the transferred channel becomes a zombie (so tech data is not available), making ast_dtmf_stream() segfault when it tries to send the DTMF digit (at least with SIP channels). Patch based on feature-end-zombie.patch uploaded by Irontec (license 1256) * Check for zombies when ast_channel_bridge() returns. * Guarantee that the fo parameter value is initialized in ast_channel_bridge() before any returns. (closes issue #19116) Reported by: Irontec Tested by: rmudgett ........ 2011-05-20 16:27 +0000 [r320040] Jonathan Rose * funcs/func_strings.c, CHANGES: Adds STRREPLACE function Adds a new STRREPLACe function to func_strings.c that allows users to search and replace against a variable in the dialplan. (closes issue #18023) Reported by: wdoekes Review: https://reviewboard.asterisk.org/r/1219/ 2011-05-20 16:20 +0000 [r319998-320013] Richard Mudgett * /, apps/app_directed_pickup.c, main/features.c: Merged revisions 320007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) | 2 lines Change some variable names to make pickup code easier to understand. ........ * /, apps/app_directed_pickup.c, main/features.c: Merged revisions 319997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) | 25 lines Crash when using directed pickup applications. The directed pickup applications can cause a crash if the pickup was successful because the dialplan keeps executing. This patch does the following: * Completes the channel masquerade on a successful pickup before the application returns. The channel is now guaranteed a zombie and must not continue executing the dialplan. * Changes the return value of the directed pickup applications to return zero if the pickup failed and nonzero(-1) if the pickup succeeded. * Made some code optimizations that no longer require re-checking the pickup channel to see if it is still available to pickup. (closes issue #19310) Reported by: remiq Patches: issue19310_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, remiq, rmudgett Review: https://reviewboard.asterisk.org/r/1221/ ........ 2011-05-20 13:42 +0000 [r319867-319939] Jonathan Rose * /, channels/chan_sip.c, configs/sip.conf.sample, channels/sip/include/sip.h: Merged revisions 319938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ * /, main/features.c: Merged revisions 319866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319866 | jrose | 2011-05-19 13:32:38 -0500 (Thu, 19 May 2011) | 11 lines Fix Randomize option on Park() The randomize option was generally not working like it should have at all on Park(). This patch restores intended functionality. (closes issue #18862) Reported by: davidw Tested by: jrose Review: https://reviewboard.asterisk.org/r/1222/ ........ 2011-05-19 18:12 +0000 [r319813] Mark Murawki * cel/cel_odbc.c, /: Merged revisions 319812 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319812 | markm | 2011-05-19 13:59:01 -0400 (Thu, 19 May 2011) | 9 lines In cel_odbc, an uninitialized RWLIST is attempted to be locked. Added INIT and DESTROY for the RWLIST odbc_tables (closes issue #19331) Reported by: kobaz Patches: odbc_cel.patch uploaded by kobaz (license 834) ........ 2011-05-19 16:52 +0000 [r319759] Richard Mudgett * /, main/ccss.c: Merged revisions 319758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) | 21 lines CCSS generic agent with POTS and ISDN phones fail caller busy call-back test. If the following is true after a CCSS activation: * The generic agent is for an analog phone or ISDN phone. (Caller party) * The called party becomes available. * The caller party is not available. When the caller party becomes available, the caller is not alerted to the called party being available. The generic agent still thinks the caller is busy. * Fixed the generic agent device state event subscription to look for all device states that are considered available. * Encapsulated the device state test for CCSS generic device available in cc_generic_is_device_available(). Made the generic agent and monitor use the new function instead of the manually coded inline equivalent. JIRA AST-559 JIRA SWP-3462 ........ 2011-05-18 23:18 +0000 [r319530-319661] Terry Wilson * /, channels/chan_sip.c: Merged revisions 319654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines Merged revisions 319653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines 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. ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 319552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines Unbreak the storing of registrations for restart The fix for issue 18882 broke retrieving non-realtime peers from the ast_db on restart/reload. This patch tries to unbreak things while leaving the intent of the original fix intact. (closes issue #19318) Reported by: remiq Patches: diff.txt uploaded by twilson (license 396) Tested by: lmadsen, remiq ........ * apps/app_dial.c, /: Merged revisions 319529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319529 | twilson | 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) | 24 lines Merged revisions 319528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines 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) ........ ................ ................ 2011-05-17 22:04 +0000 [r319471] Richard Mudgett * /, channels/misdn/isdn_lib.c: Merged revisions 319469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines Merged revision 319468 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines The mISDN HDLC mode is prevented on dialed channels. The use of mISDN HDLC mode is prevented if the mISDN dial technology option 'h1' is used when config option astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC mode. Instead of setting the channel to HDLC mode it is set to transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the logging message is correct, but the if condition is not. Make check the nodsp and hdlc flags. JIRA ABE-2787 JIRA SWP-3437 .......... ................ 2011-05-17 21:59 +0000 [r319470] Damien Wedhorn * channels/chan_skinny.c: Remove extraneous line variables. The vars were either explicitly or implicitly not used. 2011-05-17 20:13 +0000 [r319427] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c: Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message. The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how much of the current time is put in Q931_IE_TIME_DATE. * Send date/time ie never. * Send date/time ie date only. * Send date/time ie date and hour. * Send date/time ie date, hour, and minute. * Send date/time ie date, hour, minute, and second. * Send date/time ie default: Libpri will send date and hhmm only when in NT PTMP mode to support ISDN phones. (closes issue #19221) Reported by: kenner JIRA SWP-3396 2011-05-17 12:54 +0000 [r319366-319368] Leif Madsen * /, apps/app_voicemail.c: Merged revisions 319367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319367 | lmadsen | 2011-05-17 07:53:50 -0500 (Tue, 17 May 2011) | 10 lines Don't create [general] voicemail context when using users.conf Prior to this patch, app_voicemail would create a [general] context when parsing users.conf. (closes issue #18891) Reported by: pdugas Patches: app_voicemail-ignore-general.patch uploaded by pdugas (license 1222) app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71) Tested by: pdugas ........ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 319365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319365 | lmadsen | 2011-05-17 07:39:37 -0500 (Tue, 17 May 2011) | 6 lines Make Debian init script lsb compliant (closes issue #18896) Reported by: manwe Patches: debian_init_lsb.patch uploaded by manwe (license 1223) ........ 2011-05-16 21:39 +0000 [r319316] Damien Wedhorn * channels/chan_skinny.c: Fix up skinny hints. Probably haven't been working for a couple of years. May still need some more love, but they are now working, both as a hint device and monitoring a hint. Changes centre around the long ago change to remove the requirement for a device name in a skinny line, and changes to the transmit_* functions. 2011-05-16 21:08 +0000 [r319262] Jonathan Rose * main/dsp.c: Merged revisions 319261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | 2 lines Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths. ........ 2011-05-16 20:41 +0000 [r319260] Richard Mudgett * /, main/ccss.c: Merged revisions 319259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) | 13 lines Deadlock between generic CCSS agent and native ISDN CCSS. Deadlock can occur when the generic CCSS agent is deleting duplicate CC offers and the native ISDN CC driver is processing an incoming CC message. The cc_core_instances container lock cannot be held when an agent or monitor callback is invoked without the possibility of a deadlock. * Make kill_duplicate_offers() remove the reference in cc_core_instances outside of the container lock. JIRA AST-566 JIRA SWP-3469 ........ 2011-05-16 18:21 +0000 [r319212] Terry Wilson * /, channels/chan_sip.c: Merged revisions 319204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319204 | twilson | 2011-05-16 13:17:43 -0500 (Mon, 16 May 2011) | 11 lines Merged revisions 319202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) | 4 lines Unlink a peer from peers_by_ip when expiring a registration Review: https://reviewboard.asterisk.org/r/1218/ ........ ................ 2011-05-16 15:58 +0000 [r319146] David Vossel * /, channels/chan_sip.c: Merged revisions 319145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319145 | dvossel | 2011-05-16 10:57:26 -0500 (Mon, 16 May 2011) | 9 lines Merged revisions 319144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) | 2 lines Fixes issue with peer ref-counting during handle_request_subscribe. (closes issue #19293) Reported by: irroot ........ ................ 2011-05-16 15:54 +0000 [r319143] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 319142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319142 | mnicholson | 2011-05-16 10:53:26 -0500 (Mon, 16 May 2011) | 8 lines 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 ........ 2011-05-16 14:56 +0000 [r319087] Gregory Nietsky * channels/chan_sip.c, res/res_fax.c, CHANGES, channels/sip/include/sip.h: When a error in T.38 negotiation happens or its rejected on a channel the state of the channel reverts to unknown this should be rejected. this is important for negotiating T.38 gateway see #13405 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected. Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states. (closes issue #18889) Reported by: irroot Tested by: irroot, darkbasic, mnicholson Review: https://reviewboard.asterisk.org/r/1115 2011-05-16 14:38 +0000 [r319086] Paul Belanger * /, configure, include/asterisk/autoconfig.h.in, configure.ac, res/res_http_post.c: Merged revisions 319085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319085 | pabelanger | 2011-05-16 10:35:21 -0400 (Mon, 16 May 2011) | 10 lines Support gmime-2.4 (closes issue #18863) Reported by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir (license 46) Tested by: tzafrir Review: https://reviewboard.asterisk.org/r/1213/ ........ 2011-05-16 14:29 +0000 [r319084] David Vossel * /, formats/format_wav.c: Merged revisions 319083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319083 | dvossel | 2011-05-16 09:26:33 -0500 (Mon, 16 May 2011) | 5 lines Fixes Big Endian build issue. (closes issue #19298) Reported by: tzafrir ........ 2011-05-15 23:17 +0000 [r319024] Damien Wedhorn * channels/chan_skinny.c: Add activatesub and dialandactivate sub. When called, activatesub first cleans up the active sub and then handles the sub passed. dialandactivatesub first sets sub->exten and then calls activatesub. Revise handle_offhook to utilise the callid sent to chan_skinny. Some other minor fixes especially around d->hookstate (which still needs some more work). 2011-05-13 18:10 +0000 [r318918-318922] Brett Bryant * main/channel.c, /: Merged revisions 318921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318921 | bbryant | 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) | 8 lines Fixes a segmentation fault in dynamic hints when a channel technology isn't loaded for a hint. (closes issue #18495) Reported by: bertrand Tested by: bertrand ........ * /, res/res_srtp.c: Merged revisions 318919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) | 10 lines This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too much time has passed between sending audio. (closes issue #18206) Reported by: bernhardsi Patches: res_srtp_unhold.patch uploaded by bernhards (license 1138) Tested by: bernhards, notthematrix ........ * /, channels/chan_sip.c: Merged revisions 318917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318917 | bbryant | 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) | 11 lines This patch allows TCP peers into the ast_db where they were previously restricted. (closes issue #18882) Reported by: cmaj Patches: patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt uploaded by cmaj (license 830) Tested by: cmaj ........ 2011-05-13 16:30 +0000 [r318869] Richard Mudgett * /, main/features.c: Merged revisions 318868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) | 19 lines CDR's are being written immediately on caller hangup. CDR's are being written immediately on caller hangup. The dialplan is not able to modify it in the h exten. The h exten in the initial context is not run before closing CDR's when the bridge is unlinked if a macro is active and does not have an h exten. * Make ast_bridge_call() check for an h exten in the current context and if a macro is active then the initial context. The first h exten found is then run before closing the CDR. (closes issue #18212) Reported by: leearcher Patches: issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/ ........ 2011-05-13 08:33 +0000 [r318833] Damien Wedhorn * channels/chan_skinny.c: Move exten used for dialing from device to subchannel. There were some issues where if a simple switch was cancelled and a new switch started before the first had timed out where the d->exten would be used for both subchannels. This was bad leading to possible invalid extensions if some digits had been entered in the abandoned simple switch and the second one was completed before the first timed out, or the second would be cancelled because d->exten would be set to nothing on the time out of the first. 2011-05-13 01:55 +0000 [r318785] Matthew Nicholson * /, channels/sip/reqresp_parser.c: Merged revisions 318720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318720 | mnicholson | 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) | 4 lines Handle ipv6 addresses in the sent-by Via: field. This change fixes a regression in via header parsing and ipv6 handling. (closes issue #18951) ........ 2011-05-13 01:50 +0000 [r318784] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 318783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) | 14 lines PRI early media won't ring. And another way to pass early media. Don't indicate that there is inband information present, just assume that the B channel is connected. * Restore clearing the dialing flag Rx squelch unconditionally when a PROCEEDING message comes in. (closes issue #19268) Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett (license 664) Tested by: tbsky ........ 2011-05-12 22:56 +0000 [r318672] Alec L Davis * /, channels/chan_sip.c, apps/app_directed_pickup.c, main/features.c, include/asterisk/features.h: Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ 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/ ........ 2011-05-12 20:44 +0000 [r318600-318635] Damien Wedhorn * channels/chan_skinny.c: Consolidate setsubstate_* into setsubstate and use a switch. Consolidate the functions and add some debugging info. Allows to be able to set a substate without explicitly knowing what the state is. * channels/chan_skinny.c: Add setsubstate_onhook. Add the setsubstate_onhook to complete the initial substate handling procedures. Added dumpsub(sub, forcehangup) which is the common way of calling setsubstate_onhook. Dumpsub attempts to activate another sub after setting the current one onhook. 2011-05-11 18:52 +0000 [r318551-318552] Terry Wilson * /, channels/chan_sip.c: Merged revisions 318550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318550 | twilson | 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) | 2 lines Comment out the REF_DEBUG that slipped in during debugging ........ * /, channels/chan_sip.c: Merged revisions 318549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318549 | twilson | 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) | 27 lines Merged revisions 318548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) | 19 lines 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/ Review: https://reviewboard.asterisk.org/r/1210/ ........ ................ 2011-05-10 23:42 +0000 [r318500] Richard Mudgett * /, channels/sig_pri.c, channels/sig_ss7.c: Merged revisions 318499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) | 15 lines Unable to pickup DAHDI/PRI call because call state is reported as DIALING. The channel state is not updated to RINGING when an ALERTING message is received. Regression caused when sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added missing channel state update to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes issue #19257) Reported by: alecdavis Patches: issue19257_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett ........ 2011-05-10 15:16 +0000 [r318437] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 318436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318436 | russell | 2011-05-10 10:13:16 -0500 (Tue, 10 May 2011) | 2 lines chan_iax2: change LOG_NOTICE to LOG_DEBUG in iax2_read(). ........ 2011-05-10 00:22 +0000 [r318400] Terry Wilson * /, channels/chan_sip.c: Merged revisions 318337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318337 | twilson | 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) | 18 lines Merged revisions 318331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) | 12 lines 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 23:16 +0000 [r318283-318352] Richard Mudgett * /, res/Makefile, res/res_features.exports.in (removed): Merged revisions 318351 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) | 6 lines Remove references to res_features and its export file. The contents of res/res_features.c was moved to into main/features.c awhile ago. There is no longer any need for the res/Makefile to reference res_features or the res_features linker exports file to exist. ........ * /, main/features.c: Merged revisions 318282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) | 24 lines Hangup extension executed twice. When a user hangs up a call, in certain circumstances, the hangup extension can end up being executed twice: 1) If a call is bridged and the 'h' extension executes the Hangup application, then the 'h' extension will be executed twice. 2) If a call is bridged within a macro (Dial or Queue), it has its own 'h' extension, the main context also has an 'h' extension, and the macro 'h' extension executes the Hangup application, then both 'h' extensions will be executed. * Revert originally commited fix for #16106 and just set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call(). The bridge code just executed an 'h' extension so the main PBX loop does not need to execute one as well. (issue #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd ........ 2011-05-09 17:13 +0000 [r318234] David Vossel * /, channels/chan_sip.c: Merged revisions 318233 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318233 | dvossel | 2011-05-09 12:09:55 -0500 (Mon, 09 May 2011) | 14 lines Merged revisions 318230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) | 7 lines 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) ........ ................ 2011-05-09 17:00 +0000 [r318232] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 318231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318231 | rmudgett | 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) | 41 lines Don't get early media for ISDN on outgoing calls. It looks to be a long-standing misinterpretation of the progress indicator ie values: 1 - Call is not end-to-end ISDN; further call progress information may be available in-band. 8 - In-band information or an appropriate pattern is now available. Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled as early media probably because the meaning of the second half of it's description was overlooked. * Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path. (closes issue #18868) Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: satish_lx .......... No inband progress on PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider sends an ALERTING with "Inband information or appropriate pattern now available", but Asterisk only generates and passes the RING to the SIP extension, not the inband message. Unfortunately, the inband message is not a ringback tone but a prompt that says the number is not in service. The SIP extension then hears two rings and the call is hungup which confuses the caller. * Post an AST_CONTROL_PROGRESS as well as opening the media path if inband audio is indicated with an ALERTING message. (closes issue #19246) Reported by: cristiandimache Patches: issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: cristiandimache ................ 2011-05-09 14:41 +0000 [r318194] Leif Madsen * main/app.c: Increase prepend filename length. (closes issue #19238) Reported by: byronclark Patches: increase_prepend_filename_length.patch uploaded by byronclark (license 1200) 2011-05-09 14:37 +0000 [r318162-318193] Jonathan Rose * main/features.c: Minor change to 318141 to improve parsing behavior. * /, configs/features.conf.sample: Merged revisions 318148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | 4 lines Documenting an observed behavior of features in features.conf. Since parkinglots use an integer for the parkinglot extensions, leading zeros specified in the configuration file are ignored. ........ 2011-05-09 14:11 +0000 [r318143] Matthew Nicholson * main/channel.c, /: Merged revisions 318142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318142 | mnicholson | 2011-05-09 09:09:38 -0500 (Mon, 09 May 2011) | 9 lines Make indicate/control frames WRITE events on framehooks. Also, if a framehook returns a non-control frame, don't forward it to the channel. (closes issue #19251) Reported by: irroot Patches: (modified) framehook_indicate.patch2 uploaded by irroot (license 52) Tested by: irroot ........ 2011-05-09 13:56 +0000 [r318141] Jonathan Rose * main/features.c, CHANGES: Allows ParkedCall application to specify a parkinglot. When invoking the app parkedcall, the argument can now include '@parkinglot' after the extension. (closes issue #18777) Reported by: cartama Patches: 0018777.diff uploaded by cartama (license 1157) Review: https://reviewboard.asterisk.org/r/1209/ 2011-05-09 07:40 +0000 [r318106] Damien Wedhorn * channels/chan_skinny.c: Add setsubstate_callwait. If a call is made to a line that already has a call and the device is offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN. 2011-05-07 23:36 +0000 [r318056-318058] Russell Bryant * res/res_config_curl.c, /: Merged revisions 318057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318057 | russell | 2011-05-07 18:35:37 -0500 (Sat, 07 May 2011) | 8 lines res_config_curl: fix a crash with static realtime. (closes issue #18413) Reported by: jmls Patches: 20101202__issue18413.diff.txt uploaded by tilghman (license 14) Tested by: jmls ........ * /, channels/chan_iax2.c: Merged revisions 318055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318055 | russell | 2011-05-07 18:24:18 -0500 (Sat, 07 May 2011) | 7 lines chan_iax2: Don't overwrite port found with an SRV lookup. (closes issue #17291) Reported by: jcovert Patches: chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert (license 551) ........ 2011-05-06 23:07 +0000 [r317996-318019] Damien Wedhorn * channels/chan_skinny.c: Only allow voicemail if substate is OFFHOOK or no channel active (UNSET). (closes issue #17901) Reported by: salecha * channels/chan_skinny.c: Rename sub->parent to sub->line. Improve readability of code, eg, (sub->parent == d->activeline) becomes (sub->line == d->activeline). * channels/chan_skinny.c: Move the hookstate from line to device. Long time coming, finally moving the hookstate from line to device. This may fix some issues where a device has multiple lines. Previously we had to run through all lines on a device to see if it was actually onhook or not. 2011-05-06 21:49 +0000 [r317968-317970] Russell Bryant * /, apps/app_meetme.c: Merged revisions 317969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines Use the right variable to print the time in a debug message. The original patch also increased some buffer sizes, but that was already done in this version. (closes issue #17034) Reported by: sysreq Patches: asterisk-issue-17034.patch uploaded by sysreq (license 1009) ........ * /, apps/app_meetme.c: Merged revisions 317967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) | 2 lines Fix some more "set but unused" compiler warnings. ........ 2011-05-06 21:10 +0000 [r317920] David Vossel * res/res_rtp_asterisk.c, /: Merged revisions 317918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317918 | dvossel | 2011-05-06 16:06:55 -0500 (Fri, 06 May 2011) | 7 lines Fixes missing colon from To/From headers in RTCP manager events. (closes issue #18221) Reported by: clegall_proformatique Patches: 18221_1.patch uploaded by ebroad (license 878) ........ 2011-05-06 21:07 +0000 [r317843-317919] Russell Bryant * main/pbx.c, /: Merged revisions 317917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317917 | russell | 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011) | 7 lines Fix calculation of free RAM to make minmemfree option work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff uploaded by loic (license 1020) ........ * contrib/scripts/import-cdr-csv-mysql.pl (added): Add a cdr_csv to MySQL import script to contrib/scripts. (closes issue #17036) Reported by: precisenetworks Patches: import-cdr-csv-mysql.pl uploaded by precisenetworks (license 1010) * apps/app_userevent.c, CHANGES: Add the Uniqueid header to Userevent. (closes issue #16962) Reported by: jlpedrosa Patches: patch.diff uploaded by jlpedrosa (license 1002) * /, channels/chan_sip.c: Merged revisions 317867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) | 10 lines chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer. Don't duplicate variables on the sip_pvt. Just reset the variable list each time. (closes issue #19202) Reported by: wdoekes Patches: issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717) ........ * /, channels/chan_sip.c: Merged revisions 317865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) | 11 lines chan_sip: fix a deadlock in check_rtp_timeout. Don't block doing silly deadlock avoidance. Just return and try again later. The funciton gets called often enough that it's fine. Also, this change was already made in trunk. (closes issue #18791) Reported by: irroot Patches: chan_sip.rtptimeout.patch uploaded by irroot (license 52) ........ * addons/app_mysql.c, /: Merged revisions 317837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) | 11 lines Fix a crash in the MySQL() application. This code was not handling channel datastores safely. The channel must be locked. (closes issue #17964) Reported by: wuwu Patches: issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license 71) Tested by: wuwu ........ 2011-05-06 19:23 +0000 [r317818-317833] Matthew Nicholson * CHANGES: Updated CHANGES to note the autoservice changes for pbx_lua * configs/extensions.lua.sample: Updated the sample pbx_lua config file to reflect autoservice changes. 2011-05-06 19:15 +0000 [r317807] Russell Bryant * /, contrib/realtime/mysql/sipfriends.sql: Merged revisions 317805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317805 | russell | 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011) | 7 lines Add a new sipfriends.sql for MySQL that has more fields in it. (closes issue #16399) Reported by: pabelanger Patches: sipfriends.sql.v3 uploaded by pabelanger (license 224) ........ 2011-05-06 19:14 +0000 [r317721-317806] Matthew Nicholson * pbx/pbx_lua.c, UPGRADE.txt: Default to starting an autoservice in pbx_lua. The autoservice is automatically stopped when applications are executed, so this shouldn't cause any problems. * pbx/pbx_lua.c, UPGRADE.txt: Make pbx_lua handle managing the autoservice better. Make autoservice_start() and autoservice_stop() return nothing. Also check if the autoservice flag is set before starting or stopping the autoservice and stop and start the autoservice when returning control to and getting control from the pbx engine. * UPGRADE.txt: Added note about changes in pbx_lua's behavior when applications do dialplan jumps * CHANGES: Use two spaces after periods for the recent pbx_lua change descriptions * CHANGES: Updated CHANGES for hints support in pbx_lua * pbx/pbx_lua.c, CHANGES: Detect Goto in pbx_lua. This code will actually detect any dialplan jump from any application that calls ast_explicit_goto(). This change is only being done in trunk as it may change the way some dialplans execute. 2011-05-06 16:23 +0000 [r317671] Richard Mudgett * /, channels/chan_sip.c: Merged revisions 317670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) | 22 lines Fix SIP connected line updates. This patch fixes a couple SIP connected line update problems: 1) The connected line needs to be updated when the initial INVITE is sent if there is a peer callerid configured. Previously, the connected line information did not get reported until the call was connected so SIP could not report connected line information in ringing or progress messages. 2) The connected line should not be updated on initial connect if there is no connected line information. Previously, all it did was wipe out any default preset CONNECTEDLINE information set by the dialplan with empty strings. (closes issue #18367) Reported by: GeorgeKonopacki Patches: issue18367_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1199/ ........ 2011-05-06 08:21 +0000 [r317596] Terry Wilson * /, apps/app_queue.c: Merged revisions 317584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317584 | twilson | 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) | 20 lines Merged revisions 317575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines 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 23:47 +0000 [r317426-317531] Russell Bryant * Makefile, /: Merged revisions 317530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317530 | russell | 2011-05-05 18:46:54 -0500 (Thu, 05 May 2011) | 10 lines If the configure script runs, force a rebuild of menuselect-tree. Some contents in the menuselect tree are dependent on configure script parameters, namely --enable-dev-mode. (closes issue #17219) Reported by: Nick_Lewis Patches: issue_17219.rev1.txt uploaded by russell (license 2) ........ * /, contrib/realtime/mysql/queue_log.sql, contrib/realtime/mysql/sipfriends.sql: Merged revisions 317486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317486 | russell | 2011-05-05 18:15:53 -0500 (Thu, 05 May 2011) | 9 lines Fix some more realtime MySQL schema issues. (closes issue #18537) Reported by: denzs Patches: sipfriends.sql.svndiff uploaded by denzs (license 1182) queue_log.sql.svndiff uploaded by denzs (license 1182) meetme.sql.svndiff uploaded by denzs (license 1182) ........ * /, contrib/realtime/mysql/meetme.sql, contrib/realtime/mysql/sipfriends.sql: Merged revisions 317484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317484 | russell | 2011-05-05 18:12:35 -0500 (Thu, 05 May 2011) | 8 lines Fix some errors in sample MySQL realtime schema files. (closes issue #18915) Reported by: Dovid Patches: sipfriends.patch uploaded by Dovid (license 652) meetme.patch uploaded by Dovid (license 652) ........ * CHANGES, res/res_calendar.c: Add "calendar show types" CLI command. (closes issue #18246) Reported by: junky Patches: calendar_types.diff uploaded by junky (license 177) * cel/cel_pgsql.c, UPGRADE.txt, configs/cel_pgsql.conf.sample, CHANGES: Add CEL extra field to cel_pgsql. (closes issue #18462) Reported by: joscas Patches: bug_18462.diff uploaded by snuffy (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180) * /, cdr/cdr_syslog.c: Merged revisions 317480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317480 | russell | 2011-05-05 18:00:55 -0500 (Thu, 05 May 2011) | 8 lines Don't lose cdr_syslog config on a reload. (closes issue #18679) Reported by: enegaard Patches: issue18679_seanbright.patch uploaded by seanbright (license 71) Tested by: enegaard ........ * channels/chan_unistim.c, channels/chan_usbradio.c, channels/chan_dahdi.c, /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_console.c, channels/chan_oss.c, channels/chan_mgcp.c, channels/misdn_config.c: Merged revisions 317478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines Fix some consistency issues with jitterbuffer config. Store the defaults noted in the sample config files in the jitterbuffer config data structure. This makes the CLI commands that output these settings show the right thing. Also only show the settings that are relevant in the settings CLI commands, based on which jitterbuffer is selected and whether it's enabled. (closes issue #19083) Reported by: rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202) ........ * /, pbx/pbx_lua.c: Merged revisions 317476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317476 | russell | 2011-05-05 17:47:57 -0500 (Thu, 05 May 2011) | 8 lines 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 ........ * cel/cel_pgsql.c, channels/chan_jingle.c, channels/sip/sdp_crypto.c, res/res_config_odbc.c, /, channels/chan_sip.c, res/res_crypto.c, pbx/pbx_lua.c, channels/iax2-provision.c, pbx/pbx_dundi.c, channels/chan_console.c, cdr/cdr_radius.c, channels/chan_iax2.c, res/res_jabber.c, res/res_config_sqlite.c: Merged revisions 317474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317474 | russell | 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) | 2 lines Fix more "set but unused" warnings. ........ * /, main/dsp.c: Merged revisions 317429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317429 | russell | 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011) | 5 lines Only display inband DTMF warning if inband DTMF detection is enabled. (closes issue #18901) Reported by: irroot ........ * /, apps/app_rpt.c: Merged revisions 317427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317427 | russell | 2011-05-05 16:58:45 -0500 (Thu, 05 May 2011) | 7 lines Fix potential memory leak, and use of uninitialized memory. (closes issue #16476) Reported by: junky Patches: M16476.diff uploaded by junky (license 177) ........ * main/manager.c, /: Merged revisions 317425 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317425 | russell | 2011-05-05 16:53:13 -0500 (Thu, 05 May 2011) | 7 lines Add missing ActioID handling to Events action. (closes issue #18949) Reported by: edersohe Patches: 0018949.patch uploaded by edersohe (license 1228) ........ 2011-05-05 21:20 +0000 [r317395] Sean Bright * main/asterisk.c: Add some new editline bindings by default, and allow for user specified configuration. I excluded the part of this patch that used the HOME environment variable since the built-in editline library goes to great lengths to disallow that. Instead only settings the EDITRC environment variable will use a user specified file. Also, the default environment variable use to determine the edit more is AST_EDITMODE instead of AST_EDITOR (although the latter is still supported). (closes issue #15929) Reported by: kkm Patches: astcli-editrc-v2.diff uploaded by kkm (license 888) 015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888) Tested by: seanbright 2011-05-05 20:46 +0000 [r317382] Damien Wedhorn * channels/chan_skinny.c: Move hold stuff to the setsubstate arrangement. skinny_hold moved to setsubstate_hold and skinny_unhold integrated into setsubstate_connected. Removed sub->onhold and replaced with SUBSTATE_HOLD. Also fixed inbound call answering by queueing an AST_CONTROL_ANSWER on answering a SUBSTATE_RINGIN sub (was a typo). 2011-05-05 20:27 +0000 [r317377] Sean Bright * /, addons/res_config_mysql.c: Merged revisions 317370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May 2011) | 10 lines Don't duplicate our data on the stack and just use the MYSQL_ROW directly. With large result sets we were blowing out the stack. (closes issue #19090) Reported by: mickecarlsson Patches: issue19090_trunk_svn.patch uploaded by seanbright (license 71) Tested by: mickecarlsson ........ 2011-05-05 19:56 +0000 [r317337] Russell Bryant * /, apps/app_queue.c: Merged revisions 317336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317336 | russell | 2011-05-05 14:55:58 -0500 (Thu, 05 May 2011) | 7 lines Increase buffer size to be PATH_MAX for a path. (closes issue #19239) Reported by: byronclark Patches: queue_announce_length.patch uploaded by byronclark (license 1200) ........ 2011-05-05 19:33 +0000 [r317334] Jonathan Rose * /, channels/chan_sip.c: Merged revisions 317283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) | 10 lines Resolves a deadlock that occurs during sip_new This is based on an uncommitted patch by jpeeler for the issue. Instead of relocking and then unlocking the channel though, we keep the lock on the channel until we are finished doing what we need to the channel. (closes issue #18441) Reported by: Alric ........ 2011-05-05 18:46 +0000 [r317282] Russell Bryant * /, channels/chan_sip.c: Merged revisions 317281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317281 | russell | 2011-05-05 13:39:44 -0500 (Thu, 05 May 2011) | 29 lines Merged revisions 317255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317255 | russell | 2011-05-05 13:29:53 -0500 (Thu, 05 May 2011) | 22 lines 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 18:09 +0000 [r317198] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 317196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317196 | mnicholson | 2011-05-05 13:02:52 -0500 (Thu, 05 May 2011) | 8 lines 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 18:08 +0000 [r317197] David Vossel * bridges/bridge_softmix.c, funcs/func_jitterbuffer.c: Fixes reliability issues with func_jitterbuffer's usage in the new ConfBridge application. 2011-05-05 15:06 +0000 [r317059-317105] Leif Madsen * /, contrib/scripts/safe_asterisk: Merged revisions 317104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317104 | lmadsen | 2011-05-05 11:04:24 -0400 (Thu, 05 May 2011) | 15 lines 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 ........ ................ * Makefile, configs/cel.conf.sample, /: Merged revisions 317058 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317058 | lmadsen | 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011) | 7 lines Remove unused directory and clear up some documentation. (closes issue #19193) Reported by: bchia Patches: cel-csv.diff uploaded by lathama (license 1028) Tested by: lathama, Marquis42 ........ 2011-05-05 09:03 +0000 [r316994-317026] Damien Wedhorn * channels/chan_skinny.c: Add setsubstate_congestion and setsubstate_progress. Move handling of both state handling from skinny_indicate to it's own sub. Also, modified behaviour to not hangup the sub and let the dialplan have a chance in doing what it wants for congestion. Added various states to substate2str and added these states where applicable for other set_substate_ procs. * channels/chan_skinny.c: Add setsubstate_busy. Move handling of setting busy state from skinny_indicate to it's own sub. Also, modified behaviour to not hangup the sub and let the dialplan have a chance in doing what it wants (eg busy(10); hangup() in the dialplan now gives a busy indication for 10 secs and then hangs up. 2011-05-05 07:09 +0000 [r316962] Stefan Schmidt * main/astobj2.c: Adding the Move to Front Hash functionality Moving a found object to the front of its bucket to reduce the necessary traversal steps to find an object. This change improves the search time on large system with many data or in link lists. (closes issue #19233) Reported by: schmidts Review: https://reviewboard.asterisk.org/r/1201/ 2011-05-05 02:34 +0000 [r316920] Sean Bright * main/manager.c, /, main/http.c, main/utils.c: Merged revisions 316917-316919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316917 | seanbright | 2011-05-04 22:23:28 -0400 (Wed, 04 May 2011) | 5 lines Make sure that tcptls_session is properly initialized. (issue #18598) Reported by: ksn ........ r316918 | seanbright | 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) | 5 lines Look at the correct buffer for our digest info instead of an empty one. (issue #18598) Reported by: ksn ........ r316919 | seanbright | 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) | 10 lines Use the correct HTTP method when generating our digest, otherwise we always fail. When calculating the 'A2' portion of our digest for verification, we need the HTTP method that is currently in use. Unfortunately our mapping function was incorrect, resulting in invalid hashes being generated and, in turn, failures in authentication. (closes issue #18598) Reported by: ksn ........ 2011-05-04 21:44 +0000 [r316885] Damien Wedhorn * channels/chan_skinny.c: Add setsubstate_ringout (equivalent to AST_STATE ringing). Renamed previous setsubstate_ringout to setsubstate_dialing for a state when attempting to dial a number, substate ringout now for when core has indicated that the channel is actually ringing on the other end. Also added substate2str for debugging purposes. 2011-05-04 18:57 +0000 [r316832] Richard Mudgett * /, apps/app_meetme.c: Merged revisions 316831 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) | 9 lines Wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits. (closes issue #18418) Reported by: MrHanMan Tested by: rmudgett ........ 2011-05-04 16:42 +0000 [r316798] David Vossel * channels/chan_sip.c, CHANGES: Reverts rev 316218 as it breaks parsing the [general] section of sip.conf. The functionality this patch attempts to achieve should already be possible using [general](+) in the config file. issue #17957 2011-05-04 16:17 +0000 [r316664-316711] Sean Bright * /, apps/app_voicemail.c: Merged revisions 316709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316709 | seanbright | 2011-05-04 12:15:32 -0400 (Wed, 04 May 2011) | 22 lines Merged revisions 316708 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316708 | seanbright | 2011-05-04 12:10:59 -0400 (Wed, 04 May 2011) | 15 lines 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 ........ ................ ................ * main/manager.c, /: Merged revisions 316663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316663 | seanbright | 2011-05-04 10:35:05 -0400 (Wed, 04 May 2011) | 8 lines Only return a single error via AMI when requesting a forbidden action. (closes issue #19216) Reported by: oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright (license 71) Tested by: seanbright ........ 2011-05-04 14:26 +0000 [r316618-316657] David Vossel * /, apps/app_chanspy.c: Merged revisions 316650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316650 | dvossel | 2011-05-04 09:25:03 -0500 (Wed, 04 May 2011) | 15 lines Merged revisions 316644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011) | 9 lines 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: Merged revisions 316617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316617 | dvossel | 2011-05-04 08:44:41 -0500 (Wed, 04 May 2011) | 19 lines Merged revisions 316616 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) | 12 lines 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 08:25 +0000 [r316552-316584] Damien Wedhorn * channels/chan_skinny.c: Add setsubstate_ringin. Added setsubstate_ringin. skinny_call now calls sss_ringin rather than inline. Fixed previous issue so that setsubstate_connected now use SUBSTATE_RINGIN to determine is an AST_CONTROL_ANSWER should be queued. * channels/chan_skinny.c: Make skinny_answer use setsubsate_connected. Cosolidated the code so that skinny_answer now uses the setsubstate procedures rather than doing the handling inline. 2011-05-04 07:13 +0000 [r316520] Tzafrir Cohen * autoconf/ast_check_pwlib.m4, /, configure: Merged revisions 316193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316193 | tzafrir | 2011-05-03 13:57:16 +0300 (ג', 03 מאי 2011) | 8 lines Re-fix bashism in ./configure: s/let/$(( ))/ A forward-port in r278985 accidentally re-introduced issue 17485. Fixing it. Thanks to Jilles Tjoelker for the good report. (closes issue #17485) ........ 2011-05-04 07:10 +0000 [r316519] Damien Wedhorn * channels/chan_skinny.c: Cleanup skinny callinfo. Cosolidated the working out of the callinfo to be sent into transmit_callinfo. Replaced ambiguous sub->outgoing with calldirection which can be SKINNY_INCOMING or SKINNY_OUTGOING (same value as the skinny protocol). 2011-05-04 02:39 +0000 [r316477] Sean Bright * /, apps/app_meetme.c: Merged revisions 316476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316476 | seanbright | 2011-05-03 22:34:01 -0400 (Tue, 03 May 2011) | 17 lines Merged revisions 316475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May 2011) | 10 lines 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 ........ ................ 2011-05-04 00:13 +0000 [r316428-316430] Tilghman Lesher * /, addons/cdr_mysql.c, addons/res_config_mysql.c: Merged revisions 316429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316429 | tilghman | 2011-05-03 19:12:25 -0500 (Tue, 03 May 2011) | 7 lines Escape column names in case they contain illegal characters ('-') or reserved words. (closes issue #19063) Reported by: festr Patches: patch uploaded by festr (license 443) ........ * channels/chan_sip.c, CHANGES: If multiple [general] contexts occur from sip.conf (usually due to external includes), merge them. The original implementation of this did the merging of all contexts with the same name in the realtime layer, but that implementation severely breaks drivers which use the same context name (e.g. iax.conf, type={peer,user}). Therefore, the implementation needs to do the merging for particular entries only, based upon what contexts would allow that in the channel driver itself. This implementation is for chan_sip only, but others could be added in the future. (closes issue #17957) Reported by: marcelloceschia Patches: chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079) Tested by: tilghman 2011-05-03 22:16 +0000 [r316337] Russell Bryant * /, channels/chan_skinny.c, pbx/pbx_dundi.c, channels/chan_mgcp.c: Merged revisions 316336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316336 | russell | 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011) | 8 lines Use htons() instead of ntohs() in some places. (closes issue #19200) Reported by: wdoekes Patches: issue19200-trunk.patch uploaded by wdoekes (license 717) issue19200-1.8.x.patch uploaded by wdoekes (license 717) ........ 2011-05-03 22:07 +0000 [r316335] David Vossel * main/channel.c, /: Merged revisions 316334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316334 | dvossel | 2011-05-03 17:05:59 -0500 (Tue, 03 May 2011) | 8 lines Fixes framehook segfault on indicate (closes issue #19215) Reported by: irroot Patches: framehook_indicate.patch uploaded by irroot (license 52) ........ 2011-05-03 21:48 +0000 [r316333] Russell Bryant * /, apps/app_minivm.c: Merged revisions 316331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316331 | russell | 2011-05-03 16:41:11 -0500 (Tue, 03 May 2011) | 2 lines Resolve another warning. ........ 2011-05-03 21:45 +0000 [r316332] David Vossel * channels/chan_local.c, /: Merged revisions 316330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316330 | dvossel | 2011-05-03 16:37:59 -0500 (Tue, 03 May 2011) | 24 lines Merged revisions 316329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316329 | dvossel | 2011-05-03 16:29:55 -0500 (Tue, 03 May 2011) | 17 lines 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/ ........ ................ ................ 2011-05-03 20:45 +0000 [r316293] Russell Bryant * channels/chan_unistim.c, main/udptl.c, main/fskmodem_float.c, main/rtp_engine.c, /, res/res_musiconhold.c, apps/app_ices.c, apps/app_followme.c, main/config.c, main/channel.c, main/cdr.c, channels/chan_phone.c, funcs/func_enum.c, main/manager.c, channels/chan_skinny.c, apps/app_minivm.c, main/features.c, main/plc.c, res/res_agi.c, apps/app_amd.c, main/pbx.c, res/res_fax.c, formats/format_wav.c, apps/app_festival.c, channels/chan_agent.c, apps/app_originate.c, apps/app_queue.c, codecs/lpc10/dyptrk.c, include/asterisk/linkedlists.h, main/file.c, main/audiohook.c, pbx/pbx_config.c, main/asterisk.c, main/dsp.c, res/res_calendar.c, apps/app_voicemail.c: Merged revisions 316265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ 2011-05-03 19:22 +0000 [r316240] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /, channels/sig_pri.c: Merged revisions 316224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316224 | rmudgett | 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011) | 16 lines The dahdi_hangup() call does not clean up the channel fully. After dahdi_hangup() has supposedly hungup an ISDN channel there is still traffic on the S0-bus because the channel was not cleaned up fully. Shuffled the hangup code to include some missing cleanup. Also fixed some code formatting in the area. I think the primary missing clean up code was the call to tone_zone_play_tone() to turn off any active tones on the channel. (closes issue #19188) Reported by: jg1234 Patches: issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested by: jg1234 ........ 2011-05-03 19:00 +0000 [r316216-316218] David Vossel * /, channels/chan_sip.c: Merged revisions 316217 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316217 | dvossel | 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011) | 9 lines Never put the Require: timer header in an Invite. This has already been discussed and should have been resolved earlier. View revsion 285565's log for more information about why it is important to not put timer in the Require header. (closes issue #18704) Reported by: mfrager ........ * /, res/res_odbc.c: Merged revisions 316215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316215 | dvossel | 2011-05-03 13:49:48 -0500 (Tue, 03 May 2011) | 9 lines Fixes a random crash (NULL reference) in res_odbc.c. (closes issue #19180) Reported by: pruiz Patches: tmp.diff uploaded by pruiz (license 1152) Tested by: pruiz, seanbright ........ 2011-05-03 18:23 +0000 [r316213] Sean Bright * main/manager.c, /: Merged revisions 316206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316206 | seanbright | 2011-05-03 14:17:36 -0400 (Tue, 03 May 2011) | 8 lines If we aren't interested in events, don't generate the FullyBooted event on AMI login. (closes issue #19089) Reported by: bklang Patches: issue19089-1.8-r316204.patch uploaded by seanbright (license 71) Tested by: seanbright ........ 2011-05-02 19:15 +0000 [r316095] Tilghman Lesher * funcs/func_curl.c, /: Merged revisions 316094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316094 | tilghman | 2011-05-02 14:09:55 -0500 (Mon, 02 May 2011) | 15 lines Merged revisions 316093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011) | 8 lines 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 ........ ................ 2011-05-02 15:58 +0000 [r316054] Paul Belanger * apps/app_meetme.c: Formatting change, remove red blobs 2011-04-27 19:15 +0000 [r315895] Matthew Nicholson * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged revisions 315894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315894 | mnicholson | 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) | 28 lines Merged revisions 315893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines 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) ........ ................ ................ 2011-04-27 17:51 +0000 [r315855-315856] David Vossel * apps/app_confbridge.c: Makes the new ConfBridge join and leave sounds be used by default rather than beep and beeperr. * main/channel.c: Clears exception flag during ast_read when func_jitterbuffer is enabled 2011-04-27 15:56 +0000 [r315811] Russell Bryant * /, main/asterisk.c: Merged revisions 315810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315810 | russell | 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011) | 2 lines Set the copyright year to 2011 in the startup message. ........ 2011-04-27 12:37 +0000 [r315766] Leif Madsen * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 315765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315765 | lmadsen | 2011-04-27 07:36:17 -0500 (Wed, 27 Apr 2011) | 4 lines Enable Russian core sound selection in menuselect. (closes issue #18724) Reported by: pbxware ........ 2011-04-26 23:10 +0000 [r315670-315675] Terry Wilson * /, channels/chan_sip.c: Merged revisions 315673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315673 | twilson | 2011-04-26 15:56:19 -0700 (Tue, 26 Apr 2011) | 25 lines Merged revisions 315672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315672 | twilson | 2011-04-26 15:52:25 -0700 (Tue, 26 Apr 2011) | 18 lines 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 ........ ................ ................ * channels/chan_sip.c: Make sure to create the caps structure for autocreated peers Because crashing is bad. * apps/app_dial.c, main/features.c, apps/app_queue.c: Merged revisions 315644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines Merged revisions 315643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines 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 22:18 +0000 [r315649] Richard Mudgett * main/pbx.c, /: Merged revisions 315645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) | 21 lines The 'e' special extension fails to trigger in at least two cases. The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if any of them do not exist. Many of the places the 'e' extension was supposed to be invoked fail because the priority was set wrong. There were two places where the 'e' extension was not even checked for fall back. * Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T' extension check and added the 'e' extension as a fall back to the two missing locations. * Prioritized and optimized some hangup tests associated with the 'e' extension. (closes issue #19136) Reported by: kshumard Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1196/ ........ 2011-04-26 19:38 +0000 [r315504] Tilghman Lesher * include/asterisk/select.h, /: Merged revisions 315503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315503 | tilghman | 2011-04-26 14:32:50 -0500 (Tue, 26 Apr 2011) | 28 lines Merged revisions 315502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315502 | tilghman | 2011-04-26 14:22:52 -0500 (Tue, 26 Apr 2011) | 21 lines 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 18:02 +0000 [r315453] Richard Mudgett * apps/app_dial.c, /: Merged revisions 315452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315452 | rmudgett | 2011-04-26 13:00:34 -0500 (Tue, 26 Apr 2011) | 1 line Add missing set of name valid flag when dialing. ........ 2011-04-26 17:41 +0000 [r315447] Russell Bryant * channels/chan_local.c, /: Merged revisions 315446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315446 | russell | 2011-04-26 12:40:23 -0500 (Tue, 26 Apr 2011) | 14 lines chan_local: resolve a deadlock. This patch resolves a fairly complex deadlock that can occur with the combination of chan_local and a dialplan switch, such as dynamic realtime extensions, which pulls autoservice into the picture when doing a dialplan lookup. (closes issue #18818) Reported by: nic Patches: issue18818.patch uploaded by jthurman (license 614) 18818.v1.txt uploaded by russell (license 2) Tested by: nic, jthurman, kterzi, steve-howes, sysreq, IshMalik ........ 2011-04-26 02:21 +0000 [r315395] Paul Belanger * /, pbx/pbx_config.c: Merged revisions 315394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315394 | pabelanger | 2011-04-25 22:18:50 -0400 (Mon, 25 Apr 2011) | 14 lines Merged revisions 315393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr 2011) | 7 lines 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 21:55 +0000 [r315350] Richard Mudgett * /, channels/chan_mgcp.c: Merged revisions 315349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315349 | rmudgett | 2011-04-25 16:49:00 -0500 (Mon, 25 Apr 2011) | 9 lines When using MGCP realtime gateway definitions, random crashes occur. Fixed incorrect linked list node removal for realtime gateways. (closes issue #18291) Reported by: nahuelgreco Patches: dangling-pointers-when-pruning.patch uploaded by nahuelgreco (license 162) ........ 2011-04-25 19:40 +0000 [r315214-315260] Russell Bryant * /, formats/format_wav.c: Merged revisions 315259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315259 | russell | 2011-04-25 14:37:32 -0500 (Mon, 25 Apr 2011) | 24 lines Merged revisions 315258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315258 | russell | 2011-04-25 14:31:44 -0500 (Mon, 25 Apr 2011) | 17 lines 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: Merged revisions 315213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315213 | russell | 2011-04-25 14:04:28 -0500 (Mon, 25 Apr 2011) | 14 lines Merged revisions 315212 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011) | 7 lines 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 07:17 +0000 [r315054] Alec L Davis * channels/chan_local.c, /: Merged revisions 315053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315053 | alecdavis | 2011-04-25 19:14:32 +1200 (Mon, 25 Apr 2011) | 23 lines Merged revisions 315052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315052 | alecdavis | 2011-04-25 19:11:12 +1200 (Mon, 25 Apr 2011) | 16 lines 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 23:01 +0000 [r315002] Alec L Davis * channels/chan_dahdi.c, /: Merged revisions 315001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315001 | alecdavis | 2011-04-23 10:59:18 +1200 (Sat, 23 Apr 2011) | 12 lines chan_dahdi: Can't return to normal ring after distinctive ring on FXS clear a previous distinctivering pattern before each new call (closes issue #18985) Reported by: bromont Patches: bug18985.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, bromont ........ 2011-04-22 21:33 +0000 [r314960] Matthew Nicholson * /, channels/chan_agent.c: Merged revisions 314959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314959 | mnicholson | 2011-04-22 16:20:08 -0500 (Fri, 22 Apr 2011) | 24 lines Merged revisions 314958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314958 | mnicholson | 2011-04-22 15:49:45 -0500 (Fri, 22 Apr 2011) | 17 lines 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 14:49 +0000 [r314824] Tzafrir Cohen * channels/chan_unistim.c, /, res/res_fax_spandsp.c: Merged revisions 314779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314779 | tzafrir | 2011-04-22 16:59:43 +0300 (ו', 22 אפר 2011) | 2 lines Fix a few typos (shown by Lintian) ........ 2011-04-22 14:08 +0000 [r314781] Russell Bryant * /, res/res_agi.c: Merged revisions 314780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314780 | russell | 2011-04-22 09:02:23 -0500 (Fri, 22 Apr 2011) | 18 lines Merged revisions 314778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314778 | russell | 2011-04-22 08:58:03 -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 ........ ................ 2011-04-21 22:53 +0000 [r314733-314735] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: Implement AMI action PRIShowSpans. PRIShowSpans works like the AMI action DAHDIShowChannels but for PRI spans. It is similar to the CLI command "pri show spans". (closes issue #15980) Reported by: dwery * channels/sig_pri.c: Simplify sig_pri.c:build_status(). * channels/chan_dahdi.c, /: Merged revisions 314732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314732 | rmudgett | 2011-04-21 17:38:44 -0500 (Thu, 21 Apr 2011) | 1 line Correct DAHDIShowChannels XML documentation. ........ 2011-04-21 18:32 +0000 [r314666] Matthew Nicholson * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c, main/http.c, configs/sip.conf.sample, configs/skinny.conf.sample, channels/sip/include/sip.h, configs/http.conf.sample: Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines 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 18:11 +0000 [r314598] David Vossel * configs/confbridge.conf.sample (added), apps/confbridge (added), bridges/bridge_softmix.c, UPGRADE.txt, include/asterisk/channel.h, res/res_musiconhold.c, CHANGES, apps/confbridge/conf_config_parser.c (added), main/channel.c, include/asterisk/bridging_technology.h, bridges/bridge_builtin_features.c, apps/confbridge/include/confbridge.h (added), apps/Makefile, include/asterisk/bridging_features.h, include/asterisk/bridging.h, include/asterisk/dsp.h, apps/app_confbridge.c, apps/confbridge/include (added), main/bridging.c, main/dsp.c: New HD ConfBridge conferencing application. Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ 2011-04-21 00:29 +0000 [r314551] Terry Wilson * /, channels/chan_sip.c: Merged revisions 314550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314550 | twilson | 2011-04-20 17:23:04 -0700 (Wed, 20 Apr 2011) | 13 lines Merged revisions 314549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) | 6 lines 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. ........ ................ 2011-04-20 20:52 +0000 [r314509] David Vossel * main/channel.c, main/abstract_jb.c, funcs/func_jitterbuffer.c (added), include/asterisk/channel.h, CHANGES, include/asterisk/abstract_jb.h: Introduction of the JITTERBUFFER dialplan function. Review: https://reviewboard.asterisk.org/r/1157/ 2011-04-20 19:56 +0000 [r314471] Shaun Ruffell * codecs/codec_dahdi.c: codec_dahdi: DAHDI still advertises formats using the old bitfields. Previously, the DAHDI format bit fields matched up with the Asterisk bitfields. Since the Asterisk codec bit fields were replaced in r306010, codec_dahdi needs to contain the formats itself. In the future, the DAHDI formats should either change to something other than bitfields, or the bitfields need to move from include/dahdi/kernel.h to include/dahdi/user.h. Signed-off-by: Shaun Ruffell 2011-04-20 16:55 +0000 [r314418] Richard Mudgett * /, include/asterisk/frame.h: Merged revisions 314417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314417 | rmudgett | 2011-04-20 11:54:02 -0500 (Wed, 20 Apr 2011) | 1 line AST_CONTROL_XXX comment changes. ........ 2011-04-20 16:37 +0000 [r314415] David Vossel * codecs/codec_resample.c: Fixes error with frame datalen being calculated from samples when this is not allwaya accurate. 2011-04-20 05:28 +0000 [r314359] Terry Wilson * main/lock.c, /: Merged revisions 314358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314358 | twilson | 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011) | 4 lines Initialize track pointer ast_reentrancy_init checks to see if it is NULL before initializing with calloc ........ 2011-04-19 15:42 +0000 [r314204-314252] Leif Madsen * main/tcptls.c, /: Merged revisions 314251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314251 | lmadsen | 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011) | 8 lines Use SSLv23_client_method instead of old SSLv2 only. (closes issue #19095) (closes issue #19138) Reported by: tzafrir Patches: no_ssl2.diff uploaded by tzafrir (license 46) Tested by: russell, chazzam ........ * /, funcs/func_channel.c: Merged revisions 314206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314206 | lmadsen | 2011-04-19 09:28:15 -0500 (Tue, 19 Apr 2011) | 14 lines Merged revisions 314205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011) | 6 lines 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, /: Merged revisions 314203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314203 | lmadsen | 2011-04-19 09:24:25 -0500 (Tue, 19 Apr 2011) | 15 lines Merged revisions 314202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011) | 7 lines 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 ........ ................ 2011-04-19 08:22 +0000 [r314158] Olle Johansson * apps/app_meetme.c: Add explanation of strange flag setup in app_meetme (stolen from Mark's message to asterisk-dev) 2011-04-18 19:48 +0000 [r314079-314116] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, include/asterisk/autoconfig.h.in, configure.ac, channels/sig_pri.c: Problems with ISDN MWI to phones. The "controlling user number" is always the number of the voice mail box which is identical with the subscriber number itself. This number which is listed in the ISDN phone MWI menu cannot be called back to contact the voice mail box. The controlling user number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 * /, res/res_agi.c: Merged revisions 314069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011) | 22 lines The AsyncAGI command loop is lax in the value it returns for the return status. * Return correct status: SUCCESS/FAILED/HANGUP. Previously, abnormal exits from the command loop such as hangup would return SUCCESS. * The "asyncagi break" command now returns SUCCESS and is now the only way to break the command loop with that status. Previously, it returned FAILED. * The AMI event AsyncAGI End is no longer sent if the AsyncAGI Start event is not sent. Previously, this happened because of an error setting up the AGI pipes. * All executed AGI commands now get an AsyncAGI Exec result event. Previously, if the command returned failure (because of hangup), the command loop just exited with FAILURE and did not send the AsyncAGI Exec result event. * Makes sure that the channel frame queue is empty on hangup. Review: https://reviewboard.asterisk.org/r/1183/ ........ * apps/app_dial.c, /: Merged revisions 314068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011) | 7 lines Unclear code in app_dial.c. Make code formatting clear. (closes issue #19134) Reported by: oej ........ 2011-04-18 16:22 +0000 [r314018-314078] David Vossel * /, channels/chan_sip.c: Merged revisions 314067 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) | 22 lines Remove the need for deadlock avoidance in chan_sip do_monitor. Deadlock avoidance between the sip pvt and the pvt->owner is very difficult. Now that channel's are ao2 objects, this complication is no longer necessary. It turns out the pvt's msg queue only exists because of deadlock avoidance (when deadlock avoidance fails msgs were added to a queue to be processed later), so this goes away as well. The technique used in the new sip_lock_pvt_full() function should be used as a template for replacing all locations where deadlock avoidance occurs between a channel tech_pvt and the pvt's owner. My hope is that this will begin a reversal of the invalid channel driver locking architecture we have been using for so long. This patch also resolves an issue where the pvt->owner gets unlocked during processing the msg queue. (closes issue #18690) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/1182/ ........ * main/rtp_engine.c, /, channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged revisions 314017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines sip codec negotiation of dynamic rtp payloads error fix This patch fixes how chan_sip handles dynamic rtp payload types it does not understand. At the moment if a dynamic payload's mime type does not match one we understand, the payload does not get removed from our payload table. As a result of this, the payload is set to whatever dynamic codec we use internally for that payload number on outgoing INVITES. This is incorrect. This patch fixes this by properly checking the rtpmap set function's return code to make sure it was found. The function can return both -1 and -2 depending on the source of the mismatch. We were just checking -1 explicitly. Review: https://reviewboard.asterisk.org/r/1169/ ........ 2011-04-17 09:28 +0000 [r313980] Damien Wedhorn * channels/chan_skinny.c: Consolidate all new call calls to run through new setsubstate_ringout. (closes issue #17907) Reported by: wedhorn Patches: cleanup.stateringout.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn 2011-04-17 01:28 +0000 [r313907-313944] Alexandr Anikin * addons/chan_ooh323.c: fix compile error from r313907 * addons/chan_ooh323.c: fix trivial error with set_max_datagram on pvt->udptl 2011-04-15 15:20 +0000 [r313867] Jonathan Rose * /, main/cli.c: Merged revisions 313860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313860 | jrose | 2011-04-15 10:08:05 -0500 (Fri, 15 Apr 2011) | 17 lines Merged revisions 313859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) | 10 lines 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/ ........ ................ 2011-04-14 21:53 +0000 [r313822] Terry Wilson * res/res_rtp_asterisk.c: Sets video mark bit on format field correctly This fixes a regression in the media architecture change where video frames did not have their video mark set correctly. dvossel wrote this. twilson kindly committed this, mmichelson found the bug. 2011-04-14 21:02 +0000 [r313606-313781] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 313780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 Apr 2011) | 20 lines Leftover debug messages unconditionally sent to the console. Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config option enabled outputs the following debug messages unconditionally: Dialing T1847555121 on 1 Dialing www2w on 1 * Made debug messages in my_dial_digits() normal debug messages that do not get output unless enabled. * Reworded some debug messages in my_dial_digits() to be clearer. * Replace strncpy() with ast_copy_string() in my_dial_digits() which does the same job better. (closes issue #18847) Reported by: vmikhelson Tested by: rmudgett ........ * CREDITS, main/ccss.c, configs/ccss.conf.sample: Add Device State Information CCSS for Generic Devices. Add Asterisk Device State information and callbacks to the Call Completion Supplemental Services for generic agents. There are currently not many devices that have native support for CCSS. Even as the devices become available there may be other reasons why one may choose to not take advantage of the native abilities and stick with the generic implementation. The generic implementation is quite capable and could be greatly enhanced by adding device state capabilities. A phone could then subscribe to the device state with a BLF key in conjunction with Asterisk hints. The advantages of the device state information would allow a single button to: request CCSS, cancel a CCSS request, and display the current state of a CCSS request. For example, you may have a single button that when not lit, there is no active CCSS request. When you press that button, the dialplan can query the DEVICE_STATE() associated with that caller to determine whether they should be calling CallCompletionRequest() or CallCompletionCancel(). If there is currently a pending request, then the dialplan would cancel it. This also has the advantage of showing the true state of a request, which is an asynchronous call, even when CallCompletionRequest() thinks it was successful. The actual request could ultimately fail. Once lit, further feedback can be provided to the caller about the current state of their request since it will be updated by the CCSS State Machine as appropriate. The DEVICE_STATE mapping is configurable since the BLF being used on a given phone type may vary. The idea is to allow some level of customization as to the phone's behavior. As an example, you may want the BLF key to go solid once you have requested a callback. You may then want the LED to blink (typically ringing) when either the callback is in process, which is a visual indication that the incoming call is the desired callback. You may want it to blink when the callee is ready but you are busy, giving you a visual indication that the target is available as you may want to get off the line so that the callback can be successful. Device state information is sent back via the ast_devstate_prov_add() callback for any generic CCSS device as it traverses through the state machine. You simply provide a map between CC_STATE values and the corresponding AST_DEVICE state values. You could then generate hints against these states similar to what is possible today with Custom Devstates or MeetMe states. For example, you may have an extension 3000 that is currently associated with device SIP/3000. You could then create a feature code for that extension that may look something like: exten => *823000,hint,ccss:sip/3000 You would then subscribe a BLF button to *823000 which would point to the dialplan that handled CCSS requests/cancels using the available DEVICE_STATE() information about ccss:sip/3000 to make the decision about what to do. (closes issue #18788) Reported by: p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p lindheimer (license 558) Modified with final reviewboard comments. Tested by: p_lindheimer, loloski Review: https://reviewboard.asterisk.org/r/1105/ * /, res/res_agi.c: Merged revisions 313700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313700 | rmudgett | 2011-04-13 17:52:47 -0500 (Wed, 13 Apr 2011) | 5 lines Revert flushing stale AsyncAGI commands from -r313615. It looks like it was intentional to leave any commands or in-flight commands in the queue in case Async AGI is run again on the call. ........ * /, res/res_agi.c: Merged revisions 313658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313658 | rmudgett | 2011-04-13 12:47:43 -0500 (Wed, 13 Apr 2011) | 2 lines Miscellaneous AGI diagnostic message cleanup and code optimization. ........ * /, res/res_agi.c: Merged revisions 313615 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313615 | rmudgett | 2011-04-13 12:18:49 -0500 (Wed, 13 Apr 2011) | 5 lines * Add missing channel lock to handle_cli_agi_add_cmd(). * Flush any Async AGI commands left over from earlier Async AGI control of the call. ........ * main/channel.c, /, res/res_agi.c: Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines 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/ ........ ................ ................ 2011-04-13 15:49 +0000 [r313528] Leif Madsen * configs/iax.conf.sample, configs/users.conf.sample, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, channels/chan_iax2.c, channels/sip/include/sip.h: Add 'description' field for CLI and Manager output (closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ 2011-04-13 15:23 +0000 [r313527] Richard Mudgett * /, apps/app_dumpchan.c: Merged revisions 313517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011) | 12 lines Bring the dumpchan application inline with "core show channel". * Added fields that are in "core show channel" to dumpchan output. * Fixed reuse of formatbuf before the previous string stored there was used by snprintf. All output strings now have their own buffer. * Adjusted the buffer sizes to not be so abusive of the stack now that there are more buffers. Change requested by oej. ........ 2011-04-12 21:59 +0000 [r313482] Alexandr Anikin * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooLogChan.h, addons/chan_ooh323.c, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/ooGkClient.h, addons/ooh323c/src/ooports.c, addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooh323ep.h, addons/ooh323c/src/ootypes.h, addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooSocket.h, addons/ooh323c/src/ooq931.c: IPv6 support for chan_ooh323 IPv6 support for ooh323, bindaddr, peers and users ip can be IPv4 or IPv6 addr correction for multi-homed mode (0.0.0.0 or :: bindaddr) can work in dual 6/4 mode with :: bindaddr gatekeeper mode isn't supported in v6 mode while (issue #18278) Reported by: may213 Patches: ipv6-ooh323.patch uploaded by may213 (license 454) Review: https://reviewboard.asterisk.org/r/1004/ 2011-04-12 18:53 +0000 [r313437-313438] Jonathan Rose * /: blocking fix from 313436 that was already made in this commit * channels/chan_dahdi.c, /: Merged revisions 313435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 also went ahead and fixed the problem it introduces before committing. ........ r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr 2011) | 1 line 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/ ........ ........ ........ 2011-04-11 23:20 +0000 [r313367-313383] Richard Mudgett * apps/app_dial.c, /: Merged revisions 313368-313369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 Apr 2011) | 2 lines Backport a restructuring change from trunk to make the next change stand out. ........ r313369 | rmudgett | 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines Frames from the inbound channel should go to all outbound channels in app_dial.c. In app_dial.c:wait_for_answer() frames from the inbound channel should be sent to all outbound channels instead of only if there is just one outbound channel. Control frames like AST_CONTROL_CONNECTED_LINE need to be passed to all of the the outbound channels. This can happen if a blond transfer is done by a remote switch on the inbound channel. JIRA AST-443 JIRA SWP-2730 ........ * /, main/cli.c: Merged revisions 313366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313366 | rmudgett | 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) | 2 lines Added "Connected Line ID" and "Connected Line ID Name" to "core show channel" output. ........ 2011-04-11 19:39 +0000 [r313280] Leif Madsen * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 313279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313279 | lmadsen | 2011-04-11 14:36:40 -0500 (Mon, 11 Apr 2011) | 21 lines Merged revisions 313278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500 (Mon, 11 Apr 2011) | 14 lines 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 15:47 +0000 [r313191] Richard Mudgett * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions 313190 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313190 | rmudgett | 2011-04-11 10:40:30 -0500 (Mon, 11 Apr 2011) | 39 lines Merged revisions 313189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313189 | rmudgett | 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) | 32 lines 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 ........ ................ ................ 2011-04-09 21:00 +0000 [r313143] Alexandr Anikin * addons/chan_ooh323.c, /: Merged revisions 313142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313142 | may | 2011-04-10 00:56:17 +0400 (Sun, 10 Apr 2011) | 3 lines fix trivial bug in ooh323_indicate on AST_CONTROL_SRC... check p->rtp is not null ........ 2011-04-08 16:17 +0000 [r313100] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, channels/sig_ss7.h, channels/sig_pri.c, channels/sig_ss7.c: Add private lock deadlock avoidance callback to PRI and SS7. Factor out the equivalent function for analog. 2011-04-07 13:42 +0000 [r313049] Jonathan Rose * /, main/features.c: Merged revisions 313048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313048 | jrose | 2011-04-07 08:35:33 -0500 (Thu, 07 Apr 2011) | 16 lines Merged revisions 313047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) | 9 lines 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 10:30 +0000 [r313003-313005] Alec L Davis * /, channels/sig_pri.c: Merged revisions 313001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr 2011) | 13 lines Fix ISDN calling subaddr User Specified Odd/Even Flag Calculation of the Odd/Even flag was wrong. Implement correct algo, and set odd/even=0 if data would be truncated. Only allow automatic calculation of the O/E flag, don't let dialplan influence. (closes issue #19062) Reported by: festr Patches: bug19062.diff2.txt uploaded by alecdavis (license 585) Tested by: festr, alecdavis, rmudgett ........ * apps/app_voicemail.c: app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICE 2011-04-05 18:47 +0000 [r312868-312950] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions 312949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) | 6 lines Crash if ISDN span layer 1 is down on initial load. Regression from -r312575 B channel shifting during negotiation. * Also combine updating the alarm flag with clearing the resetting flag. ........ * /, channels/chan_sip.c: Merged revisions 312889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312889 | rmudgett | 2011-04-05 11:19:35 -0500 (Tue, 05 Apr 2011) | 5 lines Add 416 response to OPTIONS packet. RFC3261 Section 11.2 says the response code to an OPTIONS packet needs to be the same as if it were an INVITE. ........ * /, channels/chan_sip.c: Merged revisions 312866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312866 | rmudgett | 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) | 15 lines Responding to OPTIONS packet with 404 because Asterisk not looking for "s" extension. The get_destination() function was not using the "s" extension when the request URI did not specify an extension. This is a regression caused when the URI parsing code was extracted into parse_uri(). Made get_destination() substitute the "s" extension when the parsed URI results in an empty string. (closes issue #18348) Reported by: shmaize Patches: issue18348_v1.8.patch uploaded by rmudgett (license 664) Tested by: shmaize ........ 2011-04-05 14:16 +0000 [r312767] Matthew Nicholson * main/manager.c, /, configs/manager.conf.sample: Merged revisions 312766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312766 | mnicholson | 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) | 22 lines Merged revisions 312764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312764 | mnicholson | 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) | 15 lines 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 13:55 +0000 [r312756] Jonathan Rose * apps/app_meetme.c: Minor change to 'L' option for meetme to include some verb statements for the option. 2011-04-04 19:31 +0000 [r312716] Richard Mudgett * channels/sig_pri.c: Remove the channel parameter from sig_pri_handle_subcmds(). It was only used in a debug message and may not be correct anyway. 2011-04-04 17:37 +0000 [r312678-312680] Jonathan Rose * pbx/pbx_config.c: In handle_cli_dialplan_add_extension, const char pointer *into_context is used instead of a->argv[5] to improve readability. * CHANGES, pbx/pbx_config.c: Makes 'dialplan add extension' create the specified context if it does not already exist. If the user invokes 'dialplan add extension' into a non-existing context, the context will be created and a message informing the user of the context being created will be issued in cli. (closes issue #17431) Reported by: leearcher Patches: context_auto_create.diff uploaded by kobaz (license 834) Tested by: leearcher, kobaz, jrose 2011-04-04 16:17 +0000 [r312579] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions 312575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines Merged revisions 312574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines 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 23:17 +0000 [r312462-312510] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 312509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01 Apr 2011) | 22 lines When a call going out an NT-PTMP port gets rejected, Asterisk crashes. If a call is sent to an ISDN phone that rejects the call with RELEASE_COMPLETE(cause: call reject(21), or busy(17)) Asterisk crashes. I could not get my setup to crash. However, I could see the possibility from a race condition between queuing an AST_CONTROL_BUSY to the core and then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is processed before the AST_CONTROL_HANGUP is queued, the ast_channel could be destroyed out from under chan_misdn. Avoid this particular crash scenario by not queueing the AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy JIRA SWP-2679 ........ * /, main/ccss.c: Merged revisions 312461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail. The CallCompletionRequest()/CallCompletionCancel() dialplan applications exit nonzero on normal failure conditions. The nonzero exit causes the dialplan to hangup immediately. The dialplan author has no opportunity to report success/failure to the user. * Made always return zero so the dialplan can continue. * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. Also documented the values set. * Reduced the warning about no core instance in CallCompletionCancel() to a debug message. It is a normal event and should not be output at the WARNING level. (closes issue #18763) Reported by: p_lindheimer Patches: ccss.patch uploaded by p lindheimer (license 558) Modified Tested by: p_lindheimer, rmudgett JIRA SWP-3042 ........ 2011-04-01 17:28 +0000 [r312384-312423] Jonathan Rose * channels/chan_dahdi.c: Fixing bad line break from 312384 * channels/chan_dahdi.c, include/asterisk/dsp.h, CHANGES, main/dsp.c: New Feature for chan_dahdi. 4 length pattern matching. In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett 2011-04-01 10:59 +0000 [r312289] Tilghman Lesher * include/asterisk/select.h, /, addons/cdr_mysql.c, main/asterisk.c: Merged revisions 312286,312288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines Reload must react correctly against a possibly changed table, so dropping the conditional reload flag. ................ r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines Merged revisions 312287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines 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 09:08 +0000 [r312118-312212] Alec L Davis * /, apps/app_voicemail.c: Merged revisions 312211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312211 | alecdavis | 2011-04-01 22:03:11 +1300 (Fri, 01 Apr 2011) | 36 lines Merged revisions 312210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines 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 312117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312117 | alecdavis | 2011-04-01 20:32:12 +1300 (Fri, 01 Apr 2011) | 29 lines Merged revisions 312103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines 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/ ........ ................ ................ 2011-03-31 20:12 +0000 [r311981-312023] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 312022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) | 14 lines chan_misdn segfaults when DEBUG_THREADS is enabled. The segfault happens because jb->mutexjb is uninitialized from the ast_malloc(). The internals of ast_mutex_init() were assuming a nonzero value meant mutex tracking initialization had already happened. Recent changes to mutex tracking code to reduce excessive memory consumption exposed this uninitialized value. Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc(). Also eliminated redundant zero initialization code in the routine. (closes issue #18975) Reported by: irroot ........ * include/asterisk/channel.h: Fix function reference in comment. 2011-03-31 06:44 +0000 [r311931] Tilghman Lesher * /, configs/cdr_mysql.conf.sample: Merged revisions 311930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311930 | tilghman | 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011) | 6 lines Incorrect default example; the field is actually internally named "clid", not "callerid". (closes issue #19040) Reported by: wcselby Tested by: tilghman ........ 2011-03-30 01:57 +0000 [r311875] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 311874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311874 | rmudgett | 2011-03-29 20:56:05 -0500 (Tue, 29 Mar 2011) | 1 line Update some setup_dahdi_int() comments. ........ 2011-03-29 08:33 +0000 [r311806] Tilghman Lesher * cel/cel_odbc.c, /: Merged revisions 311799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311799 | tilghman | 2011-03-29 02:08:39 -0500 (Tue, 29 Mar 2011) | 7 lines Remove extraneous check from integer-type fields. (closes issue #19027) Reported by: mlehner Review: https://reviewboard.asterisk.org/r/1149/ ........ 2011-03-28 22:00 +0000 [r311752] Russell Bryant * /, apps/app_voicemail.c: Merged revisions 311751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311751 | russell | 2011-03-28 17:00:01 -0500 (Mon, 28 Mar 2011) | 2 lines Cross-reference VoiceMail() and VoiceMailMain() in the xml docs. ........ 2011-03-27 21:49 +0000 [r311688] Alexandr Anikin * addons/chan_ooh323.c, /: Merged revisions 311687 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311687 | may | 2011-03-28 01:47:13 +0400 (Mon, 28 Mar 2011) | 2 lines correct return values in ooh323_indicate for AST_CONTROL_T38_PARAMETERS ........ 2011-03-23 21:55 +0000 [r311613-311616] Brett Bryant * /, apps/app_meetme.c: Merged revisions 311615 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) | 8 lines This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. (closes issue #18070) Reported by: mav3rick Review: https://reviewboard.asterisk.org/r/1132/ ........ * /, channels/sip/reqresp_parser.c: Merged revisions 311612 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) | 9 lines Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null value. (closes issue #18821) Reported by: cmaj Patches: patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx uploaded by cmaj (license 830) ........ 2011-03-23 02:51 +0000 [r311559] Terry Wilson * /, channels/sip/reqresp_parser.c: Merged revisions 311558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) | 5 lines Don't use static declared buf in parse_name_andor_addr This function isn't used anywhere yet, but we definitely don't want to keep the same value for buf between calls to the function. ........ 2011-03-22 15:26 +0000 [r311498] David Vossel * /, apps/app_meetme.c: Merged revisions 311497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311497 | dvossel | 2011-03-22 10:25:24 -0500 (Tue, 22 Mar 2011) | 9 lines Merged revisions 311496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011) | 2 lines Fixes memory leak in MeetMe AMI action ........ ................ 2011-03-18 19:05 +0000 [r311427] Jonathan Rose * CHANGES, apps/app_followme.c: Adds an option to FollowMe that isn't useful for the bug it was made to solve. Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost. 2011-03-18 16:27 +0000 [r311385] David Vossel * codecs/codec_resample.c: Remove libresample dependency from codec_resample.c 2011-03-18 16:24 +0000 [r311373] Jonathan Rose * /, channels/chan_sip.c, res/res_fax.c, res/res_jabber.c: Merged revisions 311352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | 10 lines Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL. This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings. (closes issue #18759) Reported by: bklang Patches: null-strings.patch uploaded by bklang (license 919) ........ 2011-03-18 16:03 +0000 [r311343] Matthew Nicholson * /, res/res_fax.c: Merged revisions 311342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311342 | mnicholson | 2011-03-18 11:02:50 -0500 (Fri, 18 Mar 2011) | 2 lines Properly populate the LOCALSTATIONID channel variable. ........ 2011-03-18 03:00 +0000 [r311296-311298] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 311297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines Race condition when ISDN CallRerouting/CallDeflection invoked. The queued AST_CONTROL_BUSY could sometimes be processed before the call_forward dial string is recognized. * Moved setting the call_forwarding dial string after sending a response to the initiator and just queue an empty frame to wake up the media thread instead of an AST_CONTROL_BUSY. * Added check for empty rerouting/deflection number and respond with an error. ........ * apps/app_dial.c, /: Merged revisions 311295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines Merged revision 310986 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines Dial() o option broke when connected line feature added. The patch restores the o option behavior and adds the ability to specify the CallerID. The Dial o and f options are complementary to each other. The o option stores the CallerID on the outgoing channel as the channel's CallerID. The f option forces the CallerID sent by the outgoing channel. o(x) - The argument 'x' is optional. If not present, then specify that the CallerID that was present on the *calling* channel be stored as the CallerID on the *called* channel. This was the behavior of Asterisk 1.0 and earlier. If present, then specify the CallerID stored on the *called* channel. Note that o(${CALLERID(all)}) is similar to option o without parameters. f(x) - The argument 'x' is optional and its presence changes the behavior of this option. If not present, then force the outgoing CallerID on a call-forward or deflection to the dialplan extension for this Dial() using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If present, then force the outgoing CallerID to 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA SWP-3096 .......... ................ 2011-03-17 19:05 +0000 [r311198] Jonathan Rose * /, apps/app_chanspy.c: Merged revisions 311197 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | 11 lines This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy. (closes issue #18742) Reported by: jkister Tested by: jkister, jcovert, jrose Review: http://reviewboard.digium.internal/r/106/ ........ 2011-03-17 15:02 +0000 [r311142] Matthew Nicholson * main/manager.c, /: Merged revisions 311141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311141 | mnicholson | 2011-03-17 10:00:33 -0500 (Thu, 17 Mar 2011) | 11 lines Merged revisions 311140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar 2011) | 4 lines Don't write items to the manager socket twice. AST-2011-003 (closes issue 0018987) Reported by: ks-steven ........ ................ 2011-03-17 10:51 +0000 [r311051] Alec L Davis * /, configs/indications.conf.sample: Merged revisions 311050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311050 | alecdavis | 2011-03-17 23:49:41 +1300 (Thu, 17 Mar 2011) | 24 lines Merged revisions 311049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300 (Thu, 17 Mar 2011) | 17 lines 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 19:51 +0000 [r310941-311001] Terry Wilson * main/tcptls.c, /: Merged revisions 310999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310999 | twilson | 2011-03-16 14:47:59 -0500 (Wed, 16 Mar 2011) | 18 lines Merged revisions 310998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) | 11 lines 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, /: Merged revisions 310993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310993 | twilson | 2011-03-16 14:26:57 -0500 (Wed, 16 Mar 2011) | 11 lines Merged revisions 310992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) | 4 lines Don't keep trying to write to a closed connection See security advisory AST-2011-003. ........ ................ * /, main/features.c: Merged revisions 310902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines Merged revisions 310889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines 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 01:49 +0000 [r310835] Tilghman Lesher * addons/chan_ooh323.c, /: Merged revisions 310834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310834 | tilghman | 2011-03-14 20:48:25 -0500 (Mon, 14 Mar 2011) | 2 lines Fix branch compile. ........ 2011-03-15 01:36 +0000 [r310833] Alec L Davis * /, main/utils.c: Merged revisions 310781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310781 | alecdavis | 2011-03-15 14:00:55 +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 21:51 +0000 [r310735] Alexandr Anikin * addons/chan_ooh323.c, addons/ooh323c/src/ooCapability.c, /, addons/ooh323c/src/ooCalls.h: Merged revisions 310734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 (closes issue #18693) ........ r310734 | may | 2011-03-15 00:45:53 +0300 (Tue, 15 Mar 2011) | 12 lines Introduce t.38 parameters control functionality not full but enough for Send/RcvFax support Introduce t.38 controls between asterisk core and channel/proto layers. Not all parameters are transferred from proto layers but *Fax apps tested and work ok. (issue #18693) Reported by: benngard2 Patches: issue-18693.patch uploaded by may213 (license 454) ........ 2011-03-14 16:55 +0000 [r310637] Richard Mudgett * /, main/callerid.c: Merged revisions 310636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310636 | rmudgett | 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) | 39 lines Merged revisions 310635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines 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 15:40 +0000 [r310547-310588] Jonathan Rose * /, funcs/func_volume.c: Merged revisions 310587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310587 | jrose | 2011-03-14 10:27:57 -0500 (Mon, 14 Mar 2011) | 15 lines Merged revisions 310585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | 8 lines 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/ ........ ................ * main/audiohook.c: Fixes null reference bug introduced by audio hook changes that affects various OS distributions. Thanks David. 2011-03-12 20:42 +0000 [r310416-310500] Tilghman Lesher * /, pbx/pbx_ael.c: Merged revisions 310462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310462 | tilghman | 2011-03-12 14:27:54 -0600 (Sat, 12 Mar 2011) | 45 lines Merged revisions 310448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310448 | tilghman | 2011-03-12 14:24:54 -0600 (Sat, 12 Mar 2011) | 38 lines 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: Merged revisions 310415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310415 | tilghman | 2011-03-12 14:05:46 -0600 (Sat, 12 Mar 2011) | 14 lines Merged revisions 310414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011) | 7 lines Transactional handles should be used for the insertbuf, if available. Also, fix a possible resource leak. (closes issue #18943) Reported by: irroot ........ ................ 2011-03-11 18:54 +0000 [r310373] Jonathan Rose * include/asterisk/audiohook.h, main/audiohook.c, CHANGES, apps/app_mixmonitor.c: Mix Monitor: Now with r and t options. 2011-03-11 15:09 +0000 [r310332] Kevin P. Fleming * Makefile, configure, codecs/gsm/Makefile, configure.ac, makeopts.in, codecs/lpc10/Makefile: Use "-march=native" when possible. Recent versions of GCC have a tuning option value of 'native', which causes the compiler to optimize the build for the CPU the compile is performed on. Since most people are building Asterisk on the machine they plan to run it on, the configure script and build system will now use this value unless a different value is specified by the user in CFLAGS when the configure script is executed. In addition, this value will be used for building the GSM and LPC10 codecs as well, in preference to the logic that has been in their Makefiles forever to optimize for certain types of CPUs. 2011-03-11 06:56 +0000 [r310288] Alec L Davis * main/rtp_engine.c, /: Merged revisions 310287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call If the channel condition is one of the following after breaking out of the loop, don't try to update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81 Review: https://reviewboard.asterisk.org/r/1128/ ........ 2011-03-10 16:09 +0000 [r310241] Terry Wilson * main/manager.c, /, res/res_phoneprov.c: Merged revisions 310240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) | 13 lines Add \r\n to remaining http headers passed to ast_http_send r309204 changed the behavior of ast_http_send. It now requires headers to be passed with trailing \r\n. This change updates the remaining instances in the code that did not pass the \r\n. (closes issue #18186) Reported by: nivaldomjunior Patches: res_phoneprov.c.diff uploaded by lathama (license 1028) manager.diff.txt uploaded by twilson (license 396) Tested by: lathama ........ 2011-03-10 15:28 +0000 [r310238] Mark Michelson * /, channels/chan_sip.c: Merged revisions 310231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar 2011) | 9 lines Be more tolerant of what URI we accept for call completion PUBLISH requests. (closes issue #18946) Reported by: GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki ........ 2011-03-10 05:54 +0000 [r310143] Tilghman Lesher * res/res_config_odbc.c, /, funcs/func_odbc.c, apps/app_voicemail.c: Merged revisions 310142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310142 | tilghman | 2011-03-09 23:53:29 -0600 (Wed, 09 Mar 2011) | 19 lines Merged revisions 310141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines 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 ........ ................ ................ 2011-03-08 20:34 +0000 [r310089] Jonathan Rose * /, channels/sip/dialplan_functions.c: Merged revisions 310088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) | 9 lines Returns with an error notice if CHANNEL function of SIP channel is read without arguments. (Closes issue #18653) Reported by: wuwu Patches: diff.patch uploaded by jrose (license 1225) Tested by: jrose ........ 2011-03-08 18:19 +0000 [r310045] Terry Wilson * /, res/res_calendar.c: Merged revisions 310039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310039 | twilson | 2011-03-08 10:10:50 -0800 (Tue, 08 Mar 2011) | 11 lines Spelling fix in "calendar show calendar" s/Cartegories/Catagories/ (closes issue #18931) Reported by: pdugas Patches: res_calendar.c.patch uploaded by pdugas (license 1222) Review: [full review board URL with trailing slash] ........ 2011-03-08 16:46 +0000 [r309996] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 309994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) | 1 line Make pri parameter description consistent. ........ 2011-03-07 22:16 +0000 [r309859] Jonathan Rose * /, apps/app_mixmonitor.c: Merged revisions 309858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309858 | jrose | 2011-03-07 16:07:25 -0600 (Mon, 07 Mar 2011) | 22 lines Merged revisions 309857 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309857 | jrose | 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) | 15 lines 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 ........ ................ ................ 2011-03-07 01:01 +0000 [r309809] Tilghman Lesher * channels/chan_dahdi.c, /, configure, include/asterisk/autoconfig.h.in, main/ast_expr2f.c, configure.ac, main/ast_expr2.fl: Merged revisions 309808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines Merged revisions 309251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines 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. ........ ................ 2011-03-07 00:14 +0000 [r309766] Mark Michelson * /, configs/sip.conf.sample: Merged revisions 309765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, 06 Mar 2011) | 3 lines Indicate that Asterisk uses the Allow header to determine if MESSAGE requests should be sent. ........ 2011-03-05 17:53 +0000 [r309721] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 309720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) | 6 lines Fix caller id passed to openr2_chan_make_call (closes issue #18894) Reported by: malufrj Tested by: moy ........ 2011-03-05 10:30 +0000 [r309679] Tilghman Lesher * /, main/asterisk.c: Merged revisions 309678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309678 | tilghman | 2011-03-05 04:29:30 -0600 (Sat, 05 Mar 2011) | 14 lines Merged revisions 309677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) | 7 lines 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 23:22 +0000 [r309640] Terry Wilson * configs/calendar.conf.sample, include/asterisk/calendar.h, CHANGES, res/res_calendar.c: Add setvar option to calendaring Adding the setvar option with variable substitution on the value allows things like setting the outbound caller id name to the summary of a calendar event, etc. Values could be chained together as they are appended in order to do some scripting if necessary. Review: https://reviewboard.asterisk.org/r/1134/ 2011-03-04 19:38 +0000 [r309493-309587] Matthew Nicholson * /, pbx/pbx_lua.c: Merged revisions 309585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309585 | mnicholson | 2011-03-04 13:38:25 -0600 (Fri, 04 Mar 2011) | 9 lines Merged revisions 309584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, 04 Mar 2011) | 2 lines Restore mysterious lua_pushvalue() call removed in r309494. The mystery has been solved. ........ ................ * /, pbx/pbx_lua.c: Merged revisions 309542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309542 | mnicholson | 2011-03-04 13:00:33 -0600 (Fri, 04 Mar 2011) | 11 lines Merged revisions 309541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar 2011) | 4 lines 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: Merged revisions 309495 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309495 | mnicholson | 2011-03-04 12:10:23 -0600 (Fri, 04 Mar 2011) | 9 lines Merged revisions 309494 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, 04 Mar 2011) | 2 lines remove mysterious lua_pushvalue() that is never used ........ ................ * pbx/pbx_lua.c, configs/extensions.lua.sample: Add support for defining hints from pbx_lua (closes issue #16024) Reported by: mnicholson 2011-03-04 17:40 +0000 [r309491] Russell Bryant * channels/chan_nbs.c: Fix a buglet that prevented chan_nbs from loading (and subsequently stopped Asterisk). In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants. 2011-03-04 16:00 +0000 [r309449] Matthew Nicholson * /, pbx/pbx_lua.c: Merged revisions 309448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309448 | mnicholson | 2011-03-04 09:59:25 -0600 (Fri, 04 Mar 2011) | 8 lines Export global symbols from pbx_lua to allow modules to be loaded. Fixes a regression introduced in r278132. (closes issue #18671) Reported by: Igels Patches: pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96) Tested by: Igels ........ 2011-03-04 15:28 +0000 [r309446] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, funcs/func_channel.c, channels/sig_pri.c, UPGRADE-1.8.txt: Merged revisions 309445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i/[:]- There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ 2011-03-04 01:52 +0000 [r309404] David Ruggles * /, apps/app_externalivr.c: Merged revisions 309403 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines Merged revisions 309356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines 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) ........ ................ ................ 2011-03-02 21:08 +0000 [r309209-309300] Jason Parker * main/channel.c: Add HangupRequest manager event, to specify when/where a channel gets hung up. (closes issue #18226) Reported by: clegall_proformatique Patches: asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139) * /, channels/chan_sip.c: Merged revisions 309256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines Merged revisions 309255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines 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) ........ ................ * /, main/http.c: Merged revisions 309204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | 7 lines Fix consistency of CRLFs on HTTP headers that get sent out. (closes issue #18186) Reported by: nivaldomjunior Patches: 18186-httpheadernewline.diff uploaded by qwell (license 4) ........ 2011-03-01 21:57 +0000 [r309127-309171] Richard Mudgett * /, funcs/func_channel.c: Merged revisions 309170 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 Mar 2011) | 7 lines Document CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Added XML documentation for CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML documentation for CHANNEL(reversecharge). ........ * channels/sig_analog.c, /: Merged revisions 309126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) | 16 lines Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal. Looks like an unintended change when sig_analog.c was extracted from chan_dahdi.c. Removed useless conditional around needed code and fixed resulting compiler warning. (closes issue #18667) Reported by: enegaard Patches: issue18667.patch uploaded by enegaard (license 1197) Tested by: enegaard JIRA SWP-2965 ........ 2011-03-01 16:22 +0000 [r309090] David Vossel * /, channels/chan_sip.c: Merged revisions 309084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309084 | dvossel | 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) | 15 lines Merged revisions 309083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines 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 11:16 +0000 [r308992-309036] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, main/ast_expr2f.c, configure.ac, main/ast_expr2.fl: Merged revisions 309035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines Merged revisions 309033-309034 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines 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. ........ r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify meaning, removing double negative (stupid!) ........ ................ * /, funcs/func_odbc.c: Merged revisions 308991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308991 | tilghman | 2011-02-28 03:33:22 -0600 (Mon, 28 Feb 2011) | 14 lines Merged revisions 308990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) | 7 lines 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) ........ ................ 2011-02-25 18:58 +0000 [r308946] Alec L Davis * /, channels/chan_sip.c: Merged revisions 308945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines Fix Deadlock with attended transfer of SIP call Call path sip_set_rtp_peer (locks chan then pvt) transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (locks p->owner) But by the time p->owner lock was attempted, seems as though chan and p->owner were different. So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods. (closes issue #18837) Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81, cmaj Review: [https://reviewboard.asterisk.org/r/1126/] ........ 2011-02-24 21:43 +0000 [r308904] Richard Mudgett * main/channel.c, /: Merged revisions 308903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines Invalid read in ast_channel_set_caller_event(). Valgrind reported that ast_channel_set_caller_event() was reading data from a freed buffer when using the pre_set structure. Rearange things to pre-calculate the name and number pointer before updating the caller party structure to see if the name or number was changed. ........ 2011-02-24 17:59 +0000 [r308816] Terry Wilson * main/manager.c, /: Merged revisions 308815 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines Merged revisions 308814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines 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 15:10 +0000 [r308724] Matthew Nicholson * main/udptl.c, /: Merged revisions 308723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308723 | mnicholson | 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) | 16 lines Merged revisions 308722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) | 9 lines 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 03:49 +0000 [r308680] Terry Wilson * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 308679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines Merged revisions 308678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines 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/ ........ ................ 2011-02-23 23:55 +0000 [r308623-308624] Richard Mudgett * main/translate.c: Fix compiler warning. * /, channels/sig_pri.c: Merged revisions 308622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails. (closes issue #18874) Reported by: cmaj Patches: patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830) JIRA SWP-3172 ........ 2011-02-22 23:04 +0000 [r308582] David Vossel * main/format.c, funcs/func_speex.c, main/frame.c, main/rtp_engine.c, include/asterisk/silk.h (added), codecs/speex/fixed_generic.h (added), bridges/bridge_softmix.c, channels/chan_gtalk.c, bridges/bridge_multiplexed.c, channels/chan_iax2.c, main/format_pref.c, codecs/speex/resample.c (added), main/channel.c, funcs/func_pitchshift.c, include/asterisk/audiohook.h, channels/chan_skinny.c, main/format_cap.c, funcs/func_volume.c, codecs/speex (added), codecs/codec_resample.c, include/asterisk/format.h, codecs/speex/arch.h (added), include/asterisk/frame.h, include/asterisk/rtp_engine.h, codecs/speex/stack_alloc.h (added), main/bridging.c, apps/app_jack.c, configs/codecs.conf.sample, res/res_rtp_asterisk.c, formats/format_attr_silk.c (added), channels/chan_sip.c, main/translate.c, main/slinfactory.c, codecs/codec_speex.c, include/asterisk/_private.h, CHANGES, codecs/speex/speex_resampler.h (added), res/res_mutestream.c, include/asterisk/format_cap.h, codecs/Makefile, channels/chan_jingle.c, main/data.c, channels/iax2.h, main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c, main/asterisk.c, include/asterisk/slinfactory.h, include/asterisk/translate.h, codecs/speex/resample_sse.h (added), include/asterisk/time.h: Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ 2011-02-22 15:33 +0000 [r308527] Andrew Latham * main/http.c: 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 15:04 +0000 [r308417] Matthew Nicholson * main/udptl.c, /: Merged revisions 308416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308416 | mnicholson | 2011-02-21 09:02:20 -0600 (Mon, 21 Feb 2011) | 19 lines Merged revisions 308414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines 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 ........ ................ ................ 2011-02-21 14:14 +0000 [r308372] Andrew Latham * main/http.c: Add HTTP URI Debug logging and update notice enable reporting of the request URI / URL in debugging change funny debug note to a serious note. 2011-02-21 13:58 +0000 [r308371] Tzafrir Cohen * main/pbx.c: fix a memory leak in device state The callback handle_statechange (pbx.c) fails to release its data pointer, leaking memory in the process. Reported by: tzafrir Patches: 18735_pbx_free_callback.diff uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/1110/ 2011-02-19 14:07 +0000 [r308331] 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. 2011-02-19 11:03 +0000 [r308289] Tilghman Lesher * utils, /: Merged revisions 308288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308288 | tilghman | 2011-02-19 05:02:49 -0600 (Sat, 19 Feb 2011) | 2 lines A few more (copies of) files to ignore in this directory. ........ 2011-02-18 00:11 +0000 [r308243] Alexandr Anikin * addons/chan_ooh323.c, /, addons/ooh323cDriver.c, addons/ooh323cDriver.h: Merged revisions 308242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 lines added g729onlyA option for announce only AnnexA g.729 codec in h.323 capabilities. Option can be global or per user/peer. ........ 2011-02-17 20:21 +0000 [r308205] Richard Mudgett * channels/chan_dahdi.c: Add more verbage to CLI command 'pri show channels' usage. 2011-02-16 22:02 +0000 [r308157] Paul Belanger * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308150 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb 2011) | 2 lines Fix FreeBSD builds. ........ 2011-02-16 08:06 +0000 [r308099] Alexandr Anikin * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308098 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 lines ifdef __linux__ keepalive variables also ........ 2011-02-15 23:34 +0000 [r308013] Jason Parker * /, apps/app_queue.c: Merged revisions 308010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines Merged revisions 308007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines 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 23:07 +0000 [r307969] Alexandr Anikin * addons/ooh323c/src/ooSocket.c: include tcp keepalive socket calls only on linux, freebsd and others don't have these options on sockets. 2011-02-15 21:42 +0000 [r307963-307964] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: Add CLI "pri show channels" command. List the current mapping of DAHDI B channels to Asterisk channel names and which calls are on hold or call-waiting. Calls on hold or call-waiting are not associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 * apps/app_dial.c, /: Merged revisions 307962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) | 1 line Don't crash when forcing caller id. ........ 2011-02-15 18:09 +0000 [r307927] David Vossel * channels/chan_phone.c: Fixes compile error in chan_phone for big endian 2011-02-15 16:18 +0000 [r307883] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/chan_sip.c, main/ccss.c, channels/sig_pri.c, include/asterisk/ccss.h: Merged revisions 307879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines No response sent for SIP CC subscribe/resubscribe request. Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ 2011-02-15 07:03 +0000 [r307751-307838] Tilghman Lesher * /, funcs/func_odbc.c: Merged revisions 307837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307837 | tilghman | 2011-02-15 01:02:45 -0600 (Tue, 15 Feb 2011) | 15 lines Merged revisions 307836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) | 8 lines 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: Merged revisions 307793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307793 | tilghman | 2011-02-14 14:16:55 -0600 (Mon, 14 Feb 2011) | 15 lines Merged revisions 307792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011) | 8 lines 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 ........ ................ * addons/chan_ooh323.c, addons/ooh323c/src/ooCmdChannel.c: Making trunk compile again. * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 307750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines Calling a gosub routine defined in AEL from Dial/Queue ceased to work. A bug in AEL did not distinguish between the "s" extension generated by AEL and an "s" extension that was required to exist by the chan_dahdi (or another channel) that was not supplied with a starting extension. Therefore, AEL made incorrect assumptions about what commands were permissable in the context. This was fixed by making AEL generate a different extension name. However, Dial and Queue make additional assumptions about the name of the default gosub extension. Therefore, they needed to be brought into line with a "macro" rendered by AEL (as a gosub), without breaking traditional dialplans written without the aid of AEL. Related to (issue #18480) Reported by: nivek (closes issue #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt uploaded by tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) Tested by: kkm ........ 2011-02-13 10:50 +0000 [r307677-307713] Alexandr Anikin * addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooCmdChannel.c: lc not found - it's warning, not error, change malloc to ast_calloc again * addons/chan_ooh323.c, addons/ooh323cDriver.c: change malloc to ast_calloc calls to prevent crash of asterisk 2011-02-10 22:43 +0000 [r307537] Jason Parker * contrib/init.d/rc.debian.asterisk, /, main/asterisk.c: Merged revisions 307536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines Merged revisions 307535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines 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 ........ ................ ................ 2011-02-10 17:45 +0000 [r307468] Mark Michelson * /, configs/ccss.conf.sample: Merged revisions 307467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu, 10 Feb 2011) | 5 lines Fix a gaffe in the CCSS sample configuration. Discovered by Philippe Lindheimer and pointed out on #asterisk-dev ........ 2011-02-10 17:12 +0000 [r307433] David Vossel * channels/chan_sip.c, main/format_cap.c, include/asterisk/format_cap.h: Fixes bug in chan_sip where nativeformats are not set correctly. The nativeformats field was being overwritten when it should have been appended too. This caused some format capabilities to be lost briefly and some log warnings to be output. 2011-02-10 13:29 +0000 [r307396] Alexandr Anikin * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c: Corrections for properly work with H.323v2 (older) endpoints and other small fixes. Interpret remote side H.225 version. Corrections for H.323v2 endpoints: don't start TCS and MSD before connect, don't start TCS and MSD by accepting H.245 connection, start TCS and MSD by StartH245 facility message. Other fixes: fix non zeroended remoteDisplayName issue, small fixes in call clearing by closing H.245 connection, tcp keepalive introduced on TCP connections (now is hardcoded, will be configurable in the future), don't force H.245tunneling if FastStart is active, don't send Alerting singal more than once per call. (closes issue #18542) Reported by: vmikhelson Patches: issue18542-final-3.patch uploaded by may213 (license 454) Tested by: vmikhelson 2011-02-09 22:48 +0000 [r307359] Jeff Peeler * apps/app_meetme.c, CHANGES: Add new manager action MeetmeListRooms. From the submitter: I've added a new manager action to list only the active conferences on an Asterisk system. It shows the same data displayed when you run a 'meetme list' on the Asterisk CLI. (closes issue #17905) Reported by: rcasas Patches: app_meetme.c.patch uploaded by rcasas (license 641) Review: https://reviewboard.asterisk.org/r/874/ 2011-02-09 21:46 +0000 [r307315] 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 21:08 +0000 [r307229-307274] Jeff Peeler * /, main/astobj2.c: Merged revisions 307273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307273 | jpeeler | 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011) | 8 lines Add missing debug info for ao2_link for use with REF_DEBUG in ao2 callback. (closes issue #18758) Reported by: rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) ........ * main/features.c, CHANGES: Allow parkedmusicclass to be settable for non-default parking lots. (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff * /, main/features.c: Merged revisions 307228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307228 | jpeeler | 2011-02-09 13:52:51 -0600 (Wed, 09 Feb 2011) | 17 lines Merged revisions 307227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) | 11 lines 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 ........ ................ 2011-02-09 19:17 +0000 [r307192] Tzafrir Cohen * main/loader.c: clarify warning when no loadable module support Clarify warning message when LOADABLE_MODULES is disabled but we still try to load a module. 2011-02-09 05:53 +0000 [r307143] Tilghman Lesher * main/lock.c, /: Merged revisions 307142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307142 | tilghman | 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011) | 3 lines Initialize tracking variable in structure properly. Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by me.) ........ 2011-02-08 21:24 +0000 [r307097] Jason Parker * /, main/logger.c: Merged revisions 307092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | 9 lines Fix issue with verbose messages not showing on remote console. This code was reworked recently, and since the logchannel list hadn't been created yet at this point, and it was a verbose message, it was being dropped on the floor. Now it'll continue on to where it should be handled. (closes issue #18580) Reported by: pabelanger ........ 2011-02-08 21:18 +0000 [r307071] Mark Michelson * /, main/ccss.c: Merged revisions 307065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb 2011) | 6 lines Add a couple of useful channel variables for the CC recall macro. CC_EXTEN and CC_CONTEXT will allow you to determine the channel and context that will be called when the recall occurs. ........ 2011-02-08 20:42 +0000 [r307061] Terry Wilson * /, channels/chan_sip.c: Merged revisions 306979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306979 | twilson | 2011-02-08 12:18:08 -0800 (Tue, 08 Feb 2011) | 16 lines Merged revisions 306973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306973 | twilson | 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) | 9 lines 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 20:31 +0000 [r307041] Andrew Latham * /, doc/asterisk.8, configs/asterisk.conf.sample, configs/voicemail.conf.sample, doc/asterisk.sgml: Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir 2011-02-08 19:42 +0000 [r306867-306968] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 306967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306967 | jpeeler | 2011-02-08 13:41:42 -0600 (Tue, 08 Feb 2011) | 16 lines Merged revisions 306966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) | 9 lines 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 306962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306962 | jpeeler | 2011-02-08 13:25:38 -0600 (Tue, 08 Feb 2011) | 22 lines Merged revisions 306961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines 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 306866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306866 | jpeeler | 2011-02-08 10:21:45 -0600 (Tue, 08 Feb 2011) | 16 lines Merged revisions 306865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) | 9 lines 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 ........ ................ ................ 2011-02-08 02:05 +0000 [r306827] Andrew Latham * doc/asterisk.sgml: Documentation Updates. Start updates to the man pages. (issue #16505) Reported by: tzafrir Tested by: lathama 2011-02-08 00:43 +0000 [r306755-306793] Richard Mudgett * configs/chan_dahdi.conf.sample: Define the MCID acronym in chan_dahdi.conf.sample. * channels/sig_pri.h: Use correct conditional for MCID send. * channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, funcs/func_frame_trace.c, main/features.c, CHANGES, channels/sig_pri.c, include/asterisk/frame.h: Pass a MCID request to the bridged channel. Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 2011-02-07 22:46 +0000 [r306670-306675] Terry Wilson * /, main/features.c: Merged revisions 306674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306674 | twilson | 2011-02-07 14:43:22 -0800 (Mon, 07 Feb 2011) | 24 lines Merged revisions 306673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306673 | twilson | 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) | 17 lines 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 306619 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306619 | twilson | 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) | 24 lines Merged revisions 306618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines 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/ ........ ................ ................ 2011-02-07 17:55 +0000 [r306576] Mark Michelson * /, main/ccss.c: Merged revisions 306575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb 2011) | 9 lines Rearrange a bit of code in the generic CC recall operation. By waiting to call the callback macro after the CC_INTERFACES, extension, priority, and context have been set, this information can be accessed more easily within the callback macro. Reported by Philippe Lindheimer. ........ 2011-02-07 16:33 +0000 [r306541] David Vossel * channels/chan_sip.c: Fixes use of ast_format_cap_append where ast_format_cap_copy is necessary. 2011-02-05 22:16 +0000 [r306499] Alexandr Anikin * addons/chan_ooh323.c: fix trivial issue after dvossel patch, initial zero fill user and peer structure before cap structure allocated. 2011-02-05 02:55 +0000 [r306464] Richard Mudgett * channels/chan_dahdi.c: Ignore voice frames in chan_dahdi native bridging. Hardware is handling them. 2011-02-04 22:37 +0000 [r306432] Jeff Peeler * main/manager.c: Send manager event for blackfilter only if it DOES NOT match. The logic got reversed, oops. Works properly now when multiple blackfilters are present. (closes issue #18283) Reported by: telecos82 Patches: ast_managereventfilter.patch uploaded by telecos82 (license 687) 2011-02-04 20:30 +0000 [r306396] Richard Mudgett * apps/app_dial.c, channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure, include/asterisk/autoconfig.h.in, configure.ac, CHANGES, channels/sig_pri.c: Add ISDN display ie text handling options to chan_dahdi.conf. The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 2011-02-04 19:24 +0000 [r306359] Jason Parker * /, apps/app_queue.c: Merged revisions 306356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306356 | qwell | 2011-02-04 13:24:29 -0600 (Fri, 04 Feb 2011) | 16 lines Merged revisions 306346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines 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) ........ ................ 2011-02-04 19:09 +0000 [r306325-306326] Richard Mudgett * tests/test_format_api.c: Fix compiler warning. * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 306324 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) | 9 lines Don't send redirecting updates to the caller if the dialplan forked the call. Each fork in the dial could be redirected and confuse the caller. For ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN redirects calls in sequence not in parallel. * Also fixed a formatting inconsistency in app_dial.c and make a warning message more useful about what frame type could not be written. ........ 2011-02-04 18:16 +0000 [r306258-306292] Paul Belanger * utils/extconf.c: Revert changes to extconf.c It seems extconf.c already defines some local ast_debug() functions. Theses should be removed and replaced with logger.h. A patch will be added to reviewboard shortly. * cel/cel_radius.c, addons/chan_ooh323.c, apps/app_meetme.c, main/say.c, channels/chan_gtalk.c, main/taskprocessor.c, res/res_http_post.c, res/res_musiconhold.c, channels/chan_iax2.c, res/res_jabber.c, pbx/pbx_loopback.c, main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c, main/manager.c, res/res_smdi.c, channels/chan_skinny.c, main/features.c, res/res_agi.c, main/http.c, main/logger.c, res/ais/evt.c, main/app.c, res/res_config_ldap.c, apps/app_rpt.c, res/res_rtp_asterisk.c, main/pbx.c, channels/chan_sip.c, apps/app_fax.c, include/asterisk/channel.h, channels/sig_pri.c, channels/chan_misdn.c, include/asterisk/sched.h, utils/extconf.c, codecs/codec_ilbc.c, main/audiohook.c, res/res_odbc.c, main/xmldoc.c, apps/app_voicemail.c: Replace ast_log(LOG_DEBUG, ...) with ast_debug() (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ 2011-02-04 16:42 +0000 [r306257] David Vossel * codecs/codec_ilbc.c, codecs/ex_ilbc.h: Fix compile error in codec ilbc translator. 2011-02-03 23:50 +0000 [r306216] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 306215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) | 20 lines Fix SIP deadlock involving state changes. Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper) has caused locking problems. Both of these functions lock the channel when the channel argument is passed in! In this case, the suspected problem (the backtrace makes it impossible to tell) was the private being locked in sip_set_rtp_peer and then: transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to verify that the fix was only required in 1.8 and later.) (closes issue #18491) Reported by: cmaj Patches: chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830) Tested by: cmaj ........ 2011-02-03 21:13 +0000 [r306128] Terry Wilson * channels/chan_local.c, /: Merged revisions 306127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306127 | twilson | 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) | 23 lines Merged revisions 306126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306126 | twilson | 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) | 16 lines 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 20:51 +0000 [r306125] Jeff Peeler * /, main/features.c: Merged revisions 306124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306124 | jpeeler | 2011-02-03 14:50:48 -0600 (Thu, 03 Feb 2011) | 17 lines Merged revisions 306123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) | 10 lines 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 18:37 +0000 [r306086] Jason Parker * main/frame.c: Modify alignment of 'core show codecs', since the ID is no longer a huge int. 2011-02-03 18:12 +0000 [r306010-306053] David Vossel * main/frame.c: Fixes output of "core show codecs" to display image types correctly. * apps/app_dahdibarge.c, channels/chan_local.c, main/frame.c, apps/app_record.c, apps/app_alarmreceiver.c, bridges/bridge_softmix.c, formats/format_sln16.c, apps/app_ices.c, bridges/bridge_multiplexed.c, channels/chan_iax2.c, main/astobj2.c, res/res_rtp_multicast.c, channels/chan_dahdi.c, include/asterisk/bridging_technology.h, funcs/func_pitchshift.c, pbx/pbx_spool.c, include/asterisk/audiohook.h, channels/chan_skinny.c, channels/sip/include/globals.h, apps/app_dumpchan.c, formats/format_pcm.c, formats/format_h263.c, main/bridging.c, codecs/ex_ulaw.h, channels/sip/include/sip.h, main/pbx.c, codecs/codec_g722.c, formats/format_wav.c, codecs/codec_g726.c, formats/format_ogg_vorbis.c, bridges/bridge_simple.c, include/asterisk/channel.h, apps/app_talkdetect.c, channels/iax2-parser.c, include/asterisk/format_cap.h (added), apps/app_speech_utils.c, channels/iax2-parser.h, main/data.c, funcs/func_channel.c, main/audiohook.c, codecs/codec_dahdi.c, include/asterisk/frame_defs.h, formats/format_g726.c, apps/app_mixmonitor.c, main/asterisk.c, res/res_calendar.c, apps/app_voicemail.c, channels/chan_vpb.cc, addons/format_mp3.c, formats/format_sln.c, apps/app_dictate.c, codecs/ex_g722.h, codecs/codec_gsm.c, codecs/ex_g726.h, channels/chan_gtalk.c, include/asterisk/abstract_jb.h, main/channel.c, apps/app_mp3.c, codecs/codec_resample.c, formats/format_h264.c, formats/format_siren14.c, apps/app_rpt.c, channels/chan_mgcp.c, codecs/codec_lpc10.c, channels/chan_sip.c, codecs/ex_lpc10.h, include/asterisk/format_pref.h (added), codecs/codec_alaw.c, res/res_adsi.c, tests/test_format_api.c (added), apps/app_originate.c, channels/chan_jingle.c, formats/format_vox.c, main/abstract_jb.c, include/asterisk/bridging.h, main/callerid.c, main/file.c, apps/app_sms.c, formats/format_g723.c, main/dsp.c, main/format.c (added), main/udptl.c, main/rtp_engine.c, addons/chan_ooh323.c, codecs/codec_adpcm.c, apps/app_test.c, addons/chan_ooh323.h, include/asterisk/speech.h, codecs/ex_adpcm.h, codecs/ex_alaw.h, formats/format_wav_gsm.c, include/asterisk/data.h, codecs/ex_gsm.h, main/indications.c, main/format_pref.c (added), main/cli.c, main/features.c, include/asterisk/mod_format.h, apps/app_amd.c, addons/ooh323cDriver.c, channels/chan_alsa.c, formats/format_jpeg.c, addons/ooh323cDriver.h, formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c, formats/format_g719.c, channels/h323/ast_h323.cxx, channels/chan_bridge.c, apps/app_echo.c, apps/app_fax.c, codecs/codec_speex.c, include/asterisk/slin.h, channels/chan_agent.c, channels/iax2-provision.c, codecs/ex_speex.h, channels/chan_misdn.c, include/asterisk/image.h, channels/iax2.h, codecs/codec_ilbc.c, apps/app_chanspy.c, res/res_fax_spandsp.c, include/asterisk/slinfactory.h, include/asterisk/translate.h, channels/chan_unistim.c, channels/chan_multicast_rtp.c, main/ccss.c, apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c, formats/format_siren7.c, formats/format_ilbc.c, include/asterisk/file.h, include/asterisk/callerid.h, channels/chan_phone.c, main/dial.c, main/manager.c, main/format_cap.c (added), funcs/func_frame_trace.c, res/res_agi.c, main/app.c, apps/app_confbridge.c, include/asterisk/format.h (added), main/image.c, include/asterisk/rtp_engine.h, include/asterisk/frame.h, addons/chan_mobile.c, apps/app_parkandannounce.c, apps/app_jack.c, res/res_clioriginate.c, res/res_rtp_asterisk.c, apps/app_nbscat.c, codecs/codec_a_mu.c, res/res_fax.c, apps/app_festival.c, apps/app_waitforsilence.c, include/asterisk/astobj2.h, main/slinfactory.c, main/translate.c, channels/chan_console.c, channels/h323/chan_h323.h, channels/chan_oss.c, channels/chan_usbradio.c, channels/chan_h323.c, codecs/codec_ulaw.c, include/asterisk/pbx.h, channels/chan_nbs.c, formats/format_g729.c: Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ 2011-02-03 16:13 +0000 [r305988] 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 00:29 +0000 [r305939] Richard Mudgett * main/channel.c, main/manager.c, /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions 305923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines 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 20:06 +0000 [r305845] Tilghman Lesher * /, funcs/func_env.c: Merged revisions 305844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305844 | tilghman | 2011-02-02 14:05:43 -0600 (Wed, 02 Feb 2011) | 5 lines Eliminate a file descriptor leak when using the FILE() dialplan function. (closes issue #18731) Reported by: marioabajo ........ 2011-02-02 19:30 +0000 [r305759-305843] Andrew Latham * configs/iax.conf.sample, funcs/func_enum.c, configs/dundi.conf.sample, funcs/func_callcompletion.c, /, configs/mgcp.conf.sample, configs/iaxprov.conf.sample, configs/unistim.conf.sample, apps/app_externalivr.c, configs/sip.conf.sample, configs/skinny.conf.sample, configs/h323.conf.sample, configs/sla.conf.sample, apps/app_voicemail.c: Replacing doc/* and asterisk.pdf with wiki links Adding links to http(s)://wiki.asterisk.org * configs/chan_dahdi.conf.sample, /, configs/extconfig.conf.sample, configs/res_snmp.conf.sample, main/ast_expr2f.c, res/res_timing_dahdi.c, configs/ccss.conf.sample, configs/sip.conf.sample, configs/skinny.conf.sample, main/config.c, configs/h323.conf.sample, configs/sla.conf.sample, main/ast_expr2.fl, res/res_srtp.c: Replacing doc/* with wiki links Adding links to http(s)://wiki.asterisk.org * /, channels/chan_sip.c: Replace link to old doc with new wiki page. Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions 2011-02-01 22:48 +0000 [r305693] Jason Parker * /, channels/chan_iax2.c: Merged revisions 305692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305692 | qwell | 2011-02-01 16:48:16 -0600 (Tue, 01 Feb 2011) | 7 lines Reverse sense of an error test when reading from astdb. (closes issue #18545) Reported by: jcovert Patches: chan_iax2.c.patch uploaded by jcovert (license 551) ........ 2011-02-01 21:16 +0000 [r305650] Andrew Latham * configs/sip.conf.sample: SIP Configuration Documentation sip show settings reports qualifyfreq in milliseconds. sip.conf configures qualifyfreg in seconds. 2011-02-01 19:27 +0000 [r305604] Brett Bryant * cel/cel_pgsql.c, /: Merged revisions 305603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305603 | bbryant | 2011-02-01 14:23:20 -0500 (Tue, 01 Feb 2011) | 4 lines Add a possible solution to a customer problem with reloading cel_pgsql.so quickly. ........ 2011-02-01 18:03 +0000 [r305561] Andrew Latham * /: doc/tex dir removed, but corresponding entries still exists Update README, CHANGES, and Makefile. Direct users to http://wiki.asterisk.org for documentation or to the AST.txt and AST.pdf included in the tarball. (closes issue #18443) Reported by: bas Patches: changes.diff uploaded by lathama (license 1028) readme.diff uploaded by lathama (license 1028) Tested by: lathama bas 2011-02-01 17:05 +0000 [r305474] Jason Parker * /, res/res_musiconhold.c: Merged revisions 305473 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305473 | qwell | 2011-02-01 11:04:23 -0600 (Tue, 01 Feb 2011) | 23 lines Merged revisions 305472 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305472 | qwell | 2011-02-01 11:02:09 -0600 (Tue, 01 Feb 2011) | 16 lines 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 ........ ................ ................ 2011-02-01 16:05 +0000 [r305433] Brett Bryant * apps/app_confbridge.c: Add's two features to confbridge: confbridge kick, and confbridge list. (closes issue #14389) (closes issue #18007) Reported by: jcollie Patches: 0001-Fix-up-bridging-module-so-that-menuselect-works.patch uploaded by jcollie (license 412) 0002-Add-confbridge-list-and-confbridge-kick-CLI-comm.patch uploaded by jcollie (license 412) Tested by: file Review: https://reviewboard.asterisk.org/r/1084/ 2011-02-01 00:07 +0000 [r305344] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 305343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305343 | rmudgett | 2011-01-31 18:01:09 -0600 (Mon, 31 Jan 2011) | 21 lines Merged revisions 305342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) | 14 lines 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 23:08 +0000 [r305132-305255] Jason Parker * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305254 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines Merged revisions 305253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines 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) ........ ................ ................ * main/tcptls.c, /, configs/sip.conf.sample: Merged revisions 305247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | 7 lines Add alternative name for config option. The SIP sample configuration had "tlscadir" as the option name, but chan_sip used the more correct "tlscapath". Now both are accepted. Discovered (sort of) by a user on IRC in #asterisk ........ * /, res/res_musiconhold.c: Merged revisions 305198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305198 | qwell | 2011-01-31 15:30:44 -0600 (Mon, 31 Jan 2011) | 2 lines Fix compile error. pseudofd no longer exists. ........ * /, res/res_musiconhold.c: Merged revisions 305131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305131 | qwell | 2011-01-31 15:00:25 -0600 (Mon, 31 Jan 2011) | 16 lines Merged revisions 305130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305130 | qwell | 2011-01-31 14:59:37 -0600 (Mon, 31 Jan 2011) | 9 lines 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 13:57 +0000 [r305084] 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 07:52 +0000 [r304951-305041] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 305040 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305040 | tilghman | 2011-01-31 01:51:40 -0600 (Mon, 31 Jan 2011) | 2 lines Use the non-specific API aliases, to avoid a problem with building the utils directory. ........ * /, apps/app_voicemail.c: Merged revisions 304985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304985 | tilghman | 2011-01-31 01:27:13 -0600 (Mon, 31 Jan 2011) | 16 lines Merged revisions 304978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600 (Mon, 31 Jan 2011) | 9 lines 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. ........ ................ ................ * main/lock.c, /, main/heap.c, main/utils.c, include/asterisk/lock.h, .cleancount: Merged revisions 304950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304950 | tilghman | 2011-01-31 00:41:36 -0600 (Mon, 31 Jan 2011) | 18 lines Change mutex tracking so that it only consumes memory in the core mutex object when it's actually being used. This reduces the overall size of a mutex which was 3016 bytes before this back down to 216 bytes (this is on 64-bit Linux with a glibc-implemented mutex). The exactness of the numbers here may vary slightly based upon how mutexes are implemented on a platform, but the long and short of it is that prior to this commit, chan_iax2 held down 98MB of memory on a 64-bit system for nothing more than a table of 32767 locks. After this commit, the same table occupies a mere 7MB of memory. (closes issue #18194) Reported by: job Patches: 20110124__issue18194.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/1066 ........ 2011-01-30 00:22 +0000 [r304913] Andrew Latham * funcs/func_callcompletion.c, /, apps/app_externalivr.c, apps/app_queue.c, apps/app_voicemail.c, funcs/func_realtime.c, res/res_calendar.c: Add Function and Application Relationships to documentation Add and extend the see-also sections to the documentation for applications and functions in an effort to expand the online documentation of the wiki. Also check for and update any links to moved documentation in the doc folder. 2011-01-29 23:10 +0000 [r304639-304867] Sean Bright * /, res/res_config_ldap.c: Merged revisions 304866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304866 | seanbright | 2011-01-29 18:07:18 -0500 (Sat, 29 Jan 2011) | 14 lines Merged revisions 304865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304865 | seanbright | 2011-01-29 18:05:25 -0500 (Sat, 29 Jan 2011) | 7 lines 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: Merged revisions 304777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304777 | seanbright | 2011-01-29 13:09:37 -0500 (Sat, 29 Jan 2011) | 22 lines Merged revisions 304776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan 2011) | 15 lines 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: Merged revisions 304774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304774 | seanbright | 2011-01-29 12:54:43 -0500 (Sat, 29 Jan 2011) | 16 lines Merged revisions 304773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan 2011) | 9 lines 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: Merged revisions 304730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304730 | seanbright | 2011-01-29 12:15:27 -0500 (Sat, 29 Jan 2011) | 22 lines Merged revisions 304729 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan 2011) | 15 lines 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: Merged revisions 304727 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304727 | seanbright | 2011-01-29 11:28:27 -0500 (Sat, 29 Jan 2011) | 16 lines Merged revisions 304726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan 2011) | 9 lines 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: Merged revisions 304683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304683 | seanbright | 2011-01-28 17:54:23 -0500 (Fri, 28 Jan 2011) | 16 lines Merged revisions 304659,304682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan 2011) | 5 lines 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. ........ r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri, 28 Jan 2011) | 2 lines Revert part of the previous commit that snuck in. ........ ................ * /, main/acl.c: Merged revisions 304638 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304638 | seanbright | 2011-01-28 15:19:08 -0500 (Fri, 28 Jan 2011) | 11 lines Restore some conditionals that we lost in r277814. There are some cases where ast_append_ha() is called with a NULL instead of a valid int pointer. So if we get a NULL, don't try to dereference it. (closes issue #18162) Reported by: imcdona Patches: issue0018162.patch uploaded by pabelanger (license 224) Tested by: enegaard ........ 2011-01-27 20:09 +0000 [r304600] Brett Bryant * res/res_config_pgsql.c: Patch that fixes the "realtime show pgsql cache" command crash when giving a table name, because of the use of an uninitialized variable. Fixes an error introduced in r300882. (closes issue #18605) Reported by: romain_proformatique Patches: res_config_pgsql_fix.patch uploaded by romain proformatique (license 975) Tested by: romain_proformatique 2011-01-27 20:07 +0000 [r304599] Kevin P. Fleming * res/res_fax.c: Fix bug with 'F' option for ReceiveFAX and SendFAX. Skipping the call to set_t38_fax_caps() caused the FAX session details to not be marked as supporting audio FAX either... the function's name is a bit misleading. This patch restores the single bit of non-T.38 behavior from that function when audio mode is forced. 2011-01-27 19:12 +0000 [r304555] Richard Mudgett * /, main/ccss.c: Merged revisions 304554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r304554 | rmudgett | 2011-01-27 13:08:14 -0600 (Thu, 27 Jan 2011) | 4 lines Warning message if CALLCOMPLETION(cc_callback_macro or cc_agent_dialstring) are empty. Test if the value pointer is not NULL instead of not ast_strlen_zero(). ........ 2011-01-27 17:03 +0000 [r304463-304467] Jason Parker * /, configure, configure.ac: Merged revisions 304466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304466 | qwell | 2011-01-27 11:03:01 -0600 (Thu, 27 Jan 2011) | 23 lines Merged revisions 304465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304465 | qwell | 2011-01-27 11:01:24 -0600 (Thu, 27 Jan 2011) | 16 lines 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 304462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304462 | qwell | 2011-01-27 10:48:44 -0600 (Thu, 27 Jan 2011) | 16 lines Merged revisions 304461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304461 | qwell | 2011-01-27 10:48:00 -0600 (Thu, 27 Jan 2011) | 9 lines 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. ........ ................ ................ 2011-01-27 15:57 +0000 [r304422] Kevin P. Fleming * res/res_fax.c: Rename the SendFAX/ReceiveFAX 'force audio' option. The recently added option to disable usage of T.38 for a single session should have been named 'F' for 'force audio', since that is really what the user is asking to happen (and it's a positive option instead of a negative option that way). 2011-01-27 00:06 +0000 [r304385] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, channels/sig_pri.c: Merged from revision 304341 https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, 26 Jan 2011) | 7 lines Add connected line chan_dahdi.conf pricpndialplan option. * Added from_channel value to prilocaldialplan option. JIRA ABE-2731 JIRA SWP-2842 .......... 2011-01-26 23:41 +0000 [r304384] Jeff Peeler * apps/app_followme.c: Add option to followme to delay answer until ready to bridge call. Followme answers an incoming call if it hasn't already been answered and starts MOH. Some poorly designed autodialers see the answer and start playing their message to the hold music. The 'N' option has been added to indicate ringing and not answer until the call is accepted. (closes issue #18479) Reported by: ianc Patches: trunk_followme.diff uploaded by ianc (license 998) 2011-01-26 22:39 +0000 [r304342] Kevin P. Fleming * res/res_fax.c: Add ability to disable T.38 usage for specific SendFAX/ReceiveFAX sessions. Sometimes during troubleshooting it can be useful to disable T.38 usage in order to narrow down a problem. This patch adds an 'n' option to SendFAX and ReceiveFAX so that can be done without having to disable T.38 usage entirely for the peer that Asterisk is communicating with. (inspired by trying to assist Bryant Zimmerman on asterisk-users) 2011-01-26 22:27 +0000 [r304340] Jeff Peeler * /, main/features.c: Merged revisions 304339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304339 | jpeeler | 2011-01-26 16:27:30 -0600 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 Jan 2011) | 2 lines Change delimiter used internally for GOTO_ON_BLINDXFR to commas to match 76703. ........ ................ 2011-01-26 21:03 +0000 [r304252] Mark Michelson * main/udptl.c, /: Merged revisions 304250 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600 (Wed, 26 Jan 2011) | 9 lines 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 20:44 +0000 [r304246] Matthew Nicholson * main/netsock2.c, /, channels/chan_sip.c, channels/sip/reqresp_parser.c, include/asterisk/netsock2.h, channels/sip/include/sip.h, channels/sip/include/reqresp_parser.h: Merged revisions 304245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304245 | mnicholson | 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) | 20 lines Merged revisions 304244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) | 13 lines 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 20:25 +0000 [r304195] Sean Bright * /, configs/queues.conf.sample: Merged revisions 304186 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304186 | seanbright | 2011-01-26 15:23:48 -0500 (Wed, 26 Jan 2011) | 16 lines Merged revisions 304181 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304181 | seanbright | 2011-01-26 15:22:47 -0500 (Wed, 26 Jan 2011) | 9 lines 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 19:58 +0000 [r304152] Matthew Nicholson * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions 303907 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r303907 | mnicholson | 2011-01-25 14:56:12 -0600 (Tue, 25 Jan 2011) | 2 lines Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. ........ 2011-01-26 19:40 +0000 [r304151] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 304150 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304150 | rmudgett | 2011-01-26 13:39:35 -0600 (Wed, 26 Jan 2011) | 16 lines Merged revisions 304149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600 (Wed, 26 Jan 2011) | 9 lines 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 01:27 +0000 [r304098] Sean Bright * /, main/file.c: Merged revisions 304097 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304097 | seanbright | 2011-01-25 20:26:26 -0500 (Tue, 25 Jan 2011) | 19 lines Merged revisions 304096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan 2011) | 12 lines 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 23:31 +0000 [r304008] Richard Mudgett * /, main/features.c: Merged revisions 304007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304007 | rmudgett | 2011-01-25 17:28:25 -0600 (Tue, 25 Jan 2011) | 22 lines Merged revisions 304006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304006 | rmudgett | 2011-01-25 17:25:32 -0600 (Tue, 25 Jan 2011) | 15 lines 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 22:15 +0000 [r303963] Terry Wilson * /, channels/chan_sip.c: Merged revisions 303962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303962 | twilson | 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) | 30 lines Merged revisions 303960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines 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 18:56 +0000 [r303861] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 303860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303860 | tilghman | 2011-01-25 12:55:27 -0600 (Tue, 25 Jan 2011) | 12 lines Merged revisions 303858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011) | 5 lines 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 17:58 +0000 [r303772] Richard Mudgett * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_pri.c, channels/sig_ss7.c: Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines 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 17:05 +0000 [r303679] Jeff Peeler * /, apps/app_voicemail.c: Merged revisions 303678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303678 | jpeeler | 2011-01-25 11:02:38 -0600 (Tue, 25 Jan 2011) | 33 lines Merged revisions 303677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines 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) ........ ................ ................ 2011-01-25 15:52 +0000 [r303638] Matthew Nicholson * main/utils.c: Use unsigned char in comparison for UTF8 check to quiet a compiler warning. 2011-01-24 20:57 +0000 [r303547-303551] Russell Bryant * main/channel.c, main/pbx.c, /, apps/app_meetme.c, main/features.c, include/asterisk/channel.h: Merged revisions 303549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines Merged revisions 303548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines 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/ ........ ................ ................ * contrib/scripts/install_prereq: Add gsm-devel as a package to install on redhat based systems. 2011-01-24 18:59 +0000 [r303509] Matthew Nicholson * res/res_config_curl.c, include/asterisk/utils.h, funcs/func_curl.c, channels/chan_sip.c, tests/test_utils.c, res/res_agi.c, channels/sip/reqresp_parser.c, main/http.c, main/utils.c, funcs/func_uri.c: According to section 19.1.2 of RFC 3261: For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future. The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs. The unit tests for these functions have also been updated. ABE-2705 Review: https://reviewboard.asterisk.org/r/1081/ 2011-01-24 17:21 +0000 [r303468] Jason Parker * channels/chan_dahdi.c, /: Merged revisions 303467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303467 | qwell | 2011-01-24 11:20:03 -0600 (Mon, 24 Jan 2011) | 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines 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 ........ ................ ................ 2011-01-22 04:13 +0000 [r303418] Russell Bryant * configure, configure.ac: Revert default compiler change. If someone wishes to do so, it is trivial to set your own default when running the configure script. 2011-01-21 23:11 +0000 [r303288-303376] Jason Parker * channels/chan_dahdi.c, /: Temporarily revert r303288 * channels/chan_dahdi.c, /: Merged revisions 303286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303286 | qwell | 2011-01-21 15:50:11 -0600 (Fri, 21 Jan 2011) | 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines 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 09:09 +0000 [r303198-303235] Tilghman Lesher * configure, configure.ac: Really use llvm-gcc, when available. * funcs/func_db.c, CHANGES: Add DB_KEYS. Discussion on #asterisk on 2011-01-19: (02:07:03 PM) boch: i wonder how to cycle all entries in a tree (02:07:11 PM) leifmadsen: use While() (02:07:17 PM) leifmadsen: you need to know the tree structure already though (02:07:36 PM) boch: what you mean? (02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan (02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything (02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script (02:10:13 PM) boch: for example i need to know all entries in the tree (02:10:15 PM) boch: got it (02:10:20 PM) leifmadsen: exactly (02:10:22 PM) leifmadsen: that's the problem (02:10:22 PM) boch: thank you (02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over. (02:15:35 PM) leifmadsen: database shows everything (02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show ' (02:16:41 PM) leifmadsen: guess no one has found that important enough to program :) (02:16:52 PM) leifmadsen: at that point you should probably just use a relational database... (02:17:10 PM) mateu: i dunno (02:17:16 PM) mateu: seems pretty basic to me. (02:17:16 PM) leifmadsen: me either (02:17:19 PM) leifmadsen: sure does (02:17:24 PM) leifmadsen: no one has programmed it though (02:17:28 PM) ***leifmadsen shrugs (02:17:43 PM) mateu: ok, well at least we know how it currently stands. thanks leifmadsen (02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ? (02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT() (02:30:31 PM) leifmadsen: although HASHKEYS() might work (02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS() (02:31:01 PM) leifmadsen: DBKEYS() I guess? (02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family? (02:34:02 PM) leifmadsen: ya (02:34:16 PM) leifmadsen: how would you iterate through layers of them? (02:34:30 PM) leifmadsen: i.e. family/key/key/key ? (02:34:43 PM) Corydon76-home: Essentially, yes 2011-01-20 20:35 +0000 [r303154] Richard Mudgett * /, main/ccss.c: Merged revisions 303153 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303153 | rmudgett | 2011-01-20 14:31:20 -0600 (Thu, 20 Jan 2011) | 22 lines Merged revision 303098 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, 20 Jan 2011) | 15 lines CC_INTERFACES does not get built correctly with local channels. If local channels are used with CCSS, CC_INTERFACES gets garbage prepended to it so the CC recall fails. Also CC_INTERFACES gets "&(null)" appended to it. * Initialize the buffer to eliminate the prepended garbage. * Filter out the empty interface strings to eliminate the latter. * Added a diagnostic message if the CC_INTERFACES is ever empty. JIRA ABE-2740 JIRA SWP-2848 .......... ................ 2011-01-20 19:58 +0000 [r303108] Shaun Ruffell * /, main/features.c: Merged revisions 303107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303107 | sruffell | 2011-01-20 13:57:31 -0600 (Thu, 20 Jan 2011) | 23 lines Merged revisions 303106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r303106 | sruffell | 2011-01-20 13:56:34 -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 #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 17:14 +0000 [r303011] Jeff Peeler * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions 303009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines Merged revisions 303008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines 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 16:12 +0000 [r302922] Russell Bryant * /, apps/app_privacy.c: Merged revisions 302921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302921 | russell | 2011-01-20 10:12:15 -0600 (Thu, 20 Jan 2011) | 9 lines Merged revisions 302920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011) | 2 lines Resolve a compiler warning. ........ ................ 2011-01-20 15:46 +0000 [r302919] Leif Madsen * apps/app_dial.c, /: Merged revisions 302918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302918 | lmadsen | 2011-01-20 09:45:39 -0600 (Thu, 20 Jan 2011) | 16 lines Merged revisions 302917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines 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) ........ ................ 2011-01-20 09:07 +0000 [r302879] Tilghman Lesher * configure, configure.ac: On systems which have LLVM, use that compiler. Should result in a massive speed increase. 2011-01-19 23:57 +0000 [r302838] Russell Bryant * main/manager.c, /: Merged revisions 302837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) | 2 lines Only check container count if it exists. ........ 2011-01-19 23:53 +0000 [r302835-302836] Sean Bright * main/config.c: Clarify a source comment about configuration template categories. (closes issue #18578) Reported by: astmiv Patches: asterisk.main.config.2.patch uploaded by astmiv (license 1189) * /, apps/app_voicemail.c: Merged revisions 302834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302834 | seanbright | 2011-01-19 18:49:00 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed, 19 Jan 2011) | 7 lines 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 23:33 +0000 [r302832] Paul Belanger * /, contrib/scripts/install_prereq: Merged revisions 302831 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302831 | pabelanger | 2011-01-19 18:29:45 -0500 (Wed, 19 Jan 2011) | 2 lines Add binutils-dev for BETTER_BACKTRACES ........ 2011-01-19 23:07 +0000 [r302786-302790] Russell Bryant * main/manager.c, /: Merged revisions 302789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302789 | russell | 2011-01-19 17:06:46 -0600 (Wed, 19 Jan 2011) | 11 lines Merged revisions 302788 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011) | 4 lines Turn a noisy verbose message into a debug message. This can drown your console if you're using the AMI over HTTP. ........ ................ * main/manager.c, /: Merged revisions 302785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15 lines Resolve a memory leak with the manager interface is disabled. The intent of this check as it stands in previous versions of Asterisk was to check if there are any active sessions. If there were no sessions, then the function would return immediately and not bother with queueing up the manager event to be processed. Since the conversion of storing sessions in an astobj2 container, this check will always pass. I changed it to go back to checking what was intended. The side effect of this was that if the AMI is disabled, the manager event queue is populated anyway, but the code that runs to clear out the queue never runs. A producer with no consumer is a bad thing. Reported internally by kmorgan. ........ 2011-01-19 21:35 +0000 [r302732] Richard Mudgett * /, main/features.c: Merged revisions 302713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302713 | rmudgett | 2011-01-19 15:29:22 -0600 (Wed, 19 Jan 2011) | 29 lines Merged revisions 302693 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines 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 21:24 +0000 [r302644-302686] Tilghman Lesher * /, include/asterisk/astdb.h: Merged revisions 302680 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302680 | tilghman | 2011-01-19 15:23:31 -0600 (Wed, 19 Jan 2011) | 16 lines Merged revisions 302675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302675 | tilghman | 2011-01-19 15:22:45 -0600 (Wed, 19 Jan 2011) | 9 lines 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: Merged revisions 302634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302634 | tilghman | 2011-01-19 14:24:57 -0600 (Wed, 19 Jan 2011) | 22 lines Merged revisions 302599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) | 15 lines 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 20:15 +0000 [r302601] Jason Parker * /, res/res_fax.c: Merged revisions 302600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302600 | qwell | 2011-01-19 14:14:40 -0600 (Wed, 19 Jan 2011) | 1 line Fix typo pointed out on asterisk-users list. ........ 2011-01-19 19:04 +0000 [r302507-302556] Sean Bright * /, main/utils.c: Merged revisions 302555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302555 | seanbright | 2011-01-19 14:03:32 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan 2011) | 7 lines 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: Merged revisions 302552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302552 | seanbright | 2011-01-19 13:54:47 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan 2011) | 7 lines 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: Merged revisions 302549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302549 | seanbright | 2011-01-19 13:43:11 -0500 (Wed, 19 Jan 2011) | 17 lines Merged revisions 302548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan 2011) | 10 lines 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: Merged revisions 302505 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302505 | seanbright | 2011-01-19 12:58:11 -0500 (Wed, 19 Jan 2011) | 14 lines Merged revisions 302504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan 2011) | 7 lines 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 17:15 +0000 [r302463] Paul Belanger * /, res/res_timing_timerfd.c: Merged revisions 302462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302462 | pabelanger | 2011-01-19 12:09:35 -0500 (Wed, 19 Jan 2011) | 9 lines Merged revisions 302461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed, 19 Jan 2011) | 2 lines Handle 'Resource temporarily unavailable' error more gracefully. ........ ................ 2011-01-19 15:54 +0000 [r302413-302418] Sean Bright * /, configs/extensions.conf.sample: Merged revisions 302417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302417 | seanbright | 2011-01-19 10:53:20 -0500 (Wed, 19 Jan 2011) | 16 lines Merged revisions 302416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan 2011) | 9 lines 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) ........ ................ * /, channels/chan_sip.c: Merged revisions 302414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302414 | seanbright | 2011-01-19 10:45:17 -0500 (Wed, 19 Jan 2011) | 7 lines Initialize an uninitialized variable. (closes issue #18640) Reported by: jcovert Patches: chan_sip.c.patch uploaded by jcovert (license 551) ........ * channels/chan_local.c, /: Merged revisions 302412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) | 10 lines Use appropriate type for requested format in chan_local. We were passing and storing the requested format as an int instead of format_t resulting in truncation. (closes issue #18238) Reported by: whizemen Patches: 0018238_speex16.patch uploaded by whizemen (license 1143) ........ 2011-01-18 22:06 +0000 [r302319] Richard Mudgett * /, main/features.c: Merged revisions 302318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302318 | rmudgett | 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011) | 1 line Use the expanded format type instead of plain int. ........ 2011-01-18 21:44 +0000 [r302315] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 302314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302314 | mnicholson | 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) | 18 lines Merged revisions 302313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) | 11 lines 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 20:40 +0000 [r302270] Jeff Peeler * main/pbx.c, /: Merged revisions 302266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302266 | jpeeler | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 34 lines Merged revisions 302265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) | 27 lines 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 20:21 +0000 [r302268] Russell Bryant * /, main/astobj2.c: Merged revisions 302267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302267 | russell | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 5 lines Don't enable AO2_DEBUG by default if AST_DEVMODE is on. AO2_DEBUG is not important and is causing a false compiler warning to be generated on my Ubuntu Natty dev box. ........ 2011-01-18 18:17 +0000 [r302178] Richard Mudgett * /, main/features.c: Merged revisions 302174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302174 | rmudgett | 2011-01-18 12:11:43 -0600 (Tue, 18 Jan 2011) | 102 lines Merged revisions 302173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) | 95 lines 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 16:38 +0000 [r302006-302048] 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 #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ ........ * /, configs/sip.conf.sample: Merged revisions 302005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17 Jan 2011) | 2 lines Document "encryption" option in sip.conf.sample ........ 2011-01-14 21:13 +0000 [r301947] Richard Mudgett * /, channels/sig_pri.c: Merged revisions 301946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) | 13 lines Deadlock between dahdi_request() and pri_dchannel() processing an incomming call. The sig_pri_new_ast_channel() is called with the channel private lock held when pri_dchannel() calls it and no channel private lock held when dahdi_request() calls it. The use of pri_grab() in sig_pri_new_ast_channel() could leave the channel private lock held when it returns if the lock was not held before calling it. Make sig_pri_new_ast_channel() just lock the PRI span lock instead of using pri_grab(). It is safe to do this because dahdi_request() does not have the channel private lock and the deadlock potential with the PRI span lock is only between pri_dchannel() and other threads. ........ 2011-01-14 20:18 +0000 [r301858] Brett Bryant * channels/chan_multicast_rtp.c, /: Merged revisions 301851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301851 | bbryant | 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) | 6 lines Changing previous revisions 301845/301847 to use ast_sockaddr_setnull() instead of setting the field manually to avoid uninitialized data. Review: https://reviewboard.asterisk.org/r/1076/ ........ 2011-01-14 20:07 +0000 [r301850] Andrew Latham * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to function documentation. Fix amatuer type mistake 2011-01-14 19:44 +0000 [r301847] Brett Bryant * channels/chan_multicast_rtp.c, /: Merged revisions 301845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301845 | bbryant | 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) | 9 lines Fix for a consistent MulticastRTP channel driver crash due to use of unitilized data. (closes issue #18290) (closes issue #18602) Reported by: voipgate, wybecom Review: https://reviewboard.asterisk.org/r/1076/ ........ 2011-01-14 19:39 +0000 [r301846] Andrew Latham * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to function documentation. 2011-01-14 17:34 +0000 [r301791] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 301790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301790 | jpeeler | 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) | 42 lines Resolve deadlock involving REFER. Two fixes: 1) One must always have the private unlocked before calling pbx_builtin_setvar_helper to not invalidate locking order since it locks the channel. 2) Unlock the channel before calling pbx_find_extension, which starts and stops autoservice during the lookup. The problem scenario as illustrated by the reporter: Thread: do_monitor ----------------------- handle_request_do handle_incoming handle_request_refer ast_parking_ext_valid pbx_find_extension ast_autoservice_stop while (chan_list_state == as_chan_list_state) { usleep(1000); } Thread: autoservice_run ----------------------- autoservice_run chan = ast_waitfor_n ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple / complex (depending on your system) ast_channel_lock(c[x]); handle_request_do and schedule_process_request_queue locks the owner if it exists. The autoservice thread is waiting for the channel lock, which wasn't ever released since the do_monitor thread was waiting for autoservice operations to complete. Solved by unlocking the channel but keeping a reference to guarantee safety. (closes issue #18403) Reported by: jthurman Patches: 20110103-blind_deadlock.diff uploaded by jthurman (license 614) issue18403.patch uploaded by jpeeler (license 325) Tested by: jthurman ........ 2011-01-13 17:02 +0000 [r301732] Leif Madsen * /, configs/phoneprov.conf.sample: Merged revisions 301731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301731 | lmadsen | 2011-01-13 11:01:43 -0600 (Thu, 13 Jan 2011) | 15 lines Merged revisions 301730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011) | 7 lines 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 ........ ................ 2011-01-13 16:27 +0000 [r301729] Paul Belanger * main/pbx.c, CHANGES: Add dialplan variables for asterisk.conf directories Review: https://reviewboard.asterisk.org/r/1075/ 2011-01-12 21:24 +0000 [r301684] Terry Wilson * /, channels/chan_sip.c: Merged revisions 301683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301683 | twilson | 2011-01-12 15:19:48 -0600 (Wed, 12 Jan 2011) | 15 lines Merged revisions 301682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) | 9 lines 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 18:52 +0000 [r301596] Matthew Nicholson * main/manager.c, /: Merged revisions 301595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301595 | mnicholson | 2011-01-12 12:51:37 -0600 (Wed, 12 Jan 2011) | 22 lines Merged revisions 301594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600 (Wed, 12 Jan 2011) | 15 lines 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 18:12 +0000 [r301505] Jeff Peeler * main/channel.c, /: Merged revisions 301504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301504 | jpeeler | 2011-01-12 12:12:08 -0600 (Wed, 12 Jan 2011) | 26 lines Merged revisions 301503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600 (Wed, 12 Jan 2011) | 19 lines 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 ........ ................ ................ 2011-01-12 16:05 +0000 [r301447] David Vossel * /, main/file.c: Merged revisions 301446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301446 | dvossel | 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011) | 2 lines Removal of unused variables so Asterisk will compile. ........ 2011-01-12 15:59 +0000 [r301445] Stefan Schmidt * Makefile: fix wrong text of rerun menuselect after user interface warning the warning, if no user interface for menuselect warning was found is not right. you have to rerun configure before make menuselect after installing a proper user interface. (closes issue 0018594) Reported by: Dovid 2011-01-12 00:27 +0000 [r301403] Tilghman Lesher * /, main/file.c: Merged revisions 301402 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301402 | tilghman | 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011) | 7 lines Call execl() directly for a better solution for paths with spaces. (closes issue #18600) Reported by: ebroad Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman (license 14) ........ 2011-01-11 19:19 +0000 [r301319] Paul Belanger * /, configs/extensions.conf.sample: Merged revisions 301311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301311 | pabelanger | 2011-01-11 14:16:06 -0500 (Tue, 11 Jan 2011) | 9 lines Merged revisions 301310 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301310 | pabelanger | 2011-01-11 14:14:31 -