2010-08-10 Leif Madsen * Asterisk 1.6.2.11 Released. 2010-07-26 Leif Madsen * Asterisk 1.6.2.11-rc2 Released. 2010-07-26 Leif Madsen * qwell, asterisk, branch-1.6.2, r279657 *** Really fix sounds Makefile (and make it readableish). There was a rather large syntax error that should have caused ALL versions of GNU make to fail. I don't know how it worked. (Closes issue #17716) 2010-07-22 Leif Madsen * Asterisk 1.6.2.11-rc1 Released. 2010-07-22 15:00 +0000 [r278621] Mark Michelson * main/channel.c, /: Merged revisions 278620 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul 2010) | 19 lines Merged revisions 278618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul 2010) | 13 lines Allow PLC to function properly when channels use SLIN for audio. If a channel involved in a bridge was using SLIN audio, then translation paths were not guaranteed to be set up properly since in all likelihood the number of translation steps was only 1. This patch enforces the transcode_via_slin behavior if transcode_via_slin or generic_plc is enabled and one of the formats to make compatible is SLIN. AST-352 ........ ................ 2010-07-21 18:22 +0000 [r278524] Tzafrir Cohen * channels/chan_dahdi.c: Fix invalid test for rxisoffhook in FXO channels This fixes some cases of no outgoing calls on FXO before an incoming call. Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO (KS/GS) channels.In some cases the bit would not be initialized properly before the first inbound call and thus prevent an outgoing call. If those tests are actually required by anybody, they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c . (closes issue #14577) Reported by: jkroon Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd (license 610) Tested by: frawd Review: https://reviewboard.asterisk.org/r/699/ 2010-07-21 16:20 +0000 [r278479] Russell Bryant * /, res/res_timing_pthread.c: Merged revisions 278465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines Use poll() instead of select() in res_timing_pthread to avoid stack corruption. This code did not properly check FD_SETSIZE to ensure that it did not try to select() on fds that were too large. Switching to poll() removes the limitation on the maximum fd value. (closes issue #15915) Reported by: keiron (closes issue #17187) Reported by: Eddie Edwards (closes issue #16494) Reported by: Hubguru (closes issue #15731) Reported by: flop (closes issue #12917) Reported by: falves11 (closes issue #14920) Reported by: vrban (closes issue #17199) Reported by: aleksey2000 (closes issue #15406) Reported by: kowalma (closes issue #17438) Reported by: dcabot (closes issue #17325) Reported by: glwgoes (closes issue #17118) Reported by: erikje possibly other issues, too ... ........ 2010-07-21 15:58 +0000 [r278025-278464] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 278463 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 | tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11 lines Ensure realtime conferences are treated the same as static conferences when trying to find an empty one. Also, parse the useropts properly, when retrieving from realtime, and add them to the existing flags. (closes issue #17502) Reported by: kenji Patches: 20100720__issue17502.diff.txt uploaded by tilghman (license 14) Tested by: kenji ........ * apps/app_voicemail.c, /: Merged revisions 278275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500 (Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) | 7 lines Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message. (closes issue #16350) Reported by: noahisaac Patches: 20100623__issue16350.diff.txt uploaded by tilghman (license 14) ........ ................ * main/autoservice.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 278272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278272 | tilghman | 2010-07-20 17:26:23 -0500 (Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines Do not queue up DTMF frames while a call is on hold. (Fixes ABE-2110) ........ ................ * main/manager.c, /: Merged revisions 278024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010) | 14 lines Merged revisions 278023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) | 7 lines Off-by-one error (closes issue #16506) Reported by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by tilghman (license 14) ........ ................ 2010-07-19 21:21 +0000 [r277966] Jean Galarneau * /, main/features.c: Merged revisions 277945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) | 15 lines Merged revisions 277906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | 7 lines Avoid trying to pickup a parked extension before the park operation is completed. A crash could occur if the extension is picked up while the parking extension is being announced. Testing pu->notquiteyet while searching for a parked extension resolves this crash. (ABE-2418) ........ ................ 2010-07-17 17:52 +0000 [r277774-277777] Tilghman Lesher * res/res_config_pgsql.c: Merge issues... * /, autoconf/ast_func_fork.m4, configure, include/asterisk/autoconfig.h.in: Merged revisions 277775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277775 | tilghman | 2010-07-17 12:42:32 -0500 (Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) | 5 lines Remove uclibc cross-compile triplet, as uclibc has a working fork()... it's only uclinux that does not. (closes issue #17616) Reported by: pprindeville ........ ................ * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged revisions 277773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010) | 15 lines Merged revisions 277568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines Since we split values at the semicolon, we should store values with a semicolon as an encoded value. (closes issue #17369) Reported by: gkservice Patches: 20100625__issue17369.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ 2010-07-16 23:37 +0000 [r277666] Tim Ringenbach * /, main/features.c: Merged revisions 277657 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul 2010) | 16 lines Merged revisions 277625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer. ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer, ast_bridge_call() is called for a second bridge on the same channel, and it clears that flag, which still needs to get set for when the original ast_bridge_call() gets control back and checks it. Review: https://reviewboard.asterisk.org/r/741 ........ ................ 2010-07-16 21:31 +0000 [r277563] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 277530 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul 2010) | 11 lines Merged revisions 277497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested. FAX-128 ........ ................ 2010-07-16 21:16 +0000 [r277489] Jeff Peeler * apps/app_queue.c, /: Merged revisions 277488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 | jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines Fix reporting estimated queue hold time. Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes. (closes issue #17498) Reported by: corruptor Patches: holdesecs_bug.diff uploaded by corruptor (license 253) ........ 2010-07-16 20:35 +0000 [r277485] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 277467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500 (Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when reloading dahdi module During a reload, the priexclusive and outsignalling parameters are not read in from the config file as intended. Unfortunately, they get set to defaults as a result. This patch makes sure that they do not get set to defaults during a reload. (closes issue #17441) Reported by: mtryfoss Patches: issue17441_v1.4.patch uploaded by rmudgett (license 664) issue17441_v1.6.2.patch uploaded by rmudgett (license 664) issue17441_trunk.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ ................ 2010-07-16 20:30 +0000 [r277478] Tilghman Lesher * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql (added), /: Merged revisions 277452 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r277452 | tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines Add documentation for MOH realtime fields ........ 2010-07-16 19:24 +0000 [r277377] Jeff Peeler * apps/app_queue.c, /: Merged revisions 277366 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 | jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines Add missing handling for ringing state for use with queue empty options. (closes issue #17471) Reported by: jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056) ........ 2010-07-16 18:33 +0000 [r277338] Matthew Nicholson * main/pbx.c, /: Merged revisions 277331 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul 2010) | 15 lines Merged revisions 277327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035) Reported by: francesco_r Patches: pbx.c.patch uploaded by viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar ........ ................ 2010-07-16 18:14 +0000 [r277264] Tilghman Lesher * main/manager.c, /: Merged revisions 277263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010) | 12 lines Merged revisions 277261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines If variable gotten is not set, will segfault on Solaris. (closes issue #17636) Reported by: bklang ........ ................ 2010-07-16 17:31 +0000 [r277256] Matthew Nicholson * main/channel.c, /: Merged revisions 277250 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277250 | mnicholson | 2010-07-16 12:30:39 -0500 (Fri, 16 Jul 2010) | 11 lines Merged revisions 277247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul 2010) | 4 lines For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation. AST-362 ........ ................ 2010-07-16 17:18 +0000 [r277188] Paul Belanger * /, apps/app_amd.c: Merged revisions 277183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul 2010) | 15 lines Merged revisions 277182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines Total analysis time error with SIP and silence suppression When using app_amd with SIP providers that have silence suppression on, the iTotalTime count increases exponentially. (closes issue #17656) Reported by: juls ........ ................ 2010-07-16 15:21 +0000 [r277144] Sean Bright * /, main/translate.c: Merged revisions 277143 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r277143 | seanbright | 2010-07-16 11:20:40 -0400 (Fri, 16 Jul 2010) | 8 lines Avoid crashing when installing a duplicate translation path with a lower cost. (closes issue #17092) Reported by: moy Patches: translate.rev254273.patch uploaded by moy (license 222) Tested by: moy ........ 2010-07-15 20:42 +0000 [r276572-276809] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 276788 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r276788 | jpeeler | 2010-07-15 15:21:03 -0500 (Thu, 15 Jul 2010) | 6 lines Correct not setting the bindport before attempting to open the socket. Related to changes from 276571, I was accidentally testing with a port set in my configuration causing me to miss this. Also moved the TCP handling as well to occur before build_peer is called. ........ * main/channel.c, /: Merged revisions 276653 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r276653 | jpeeler | 2010-07-15 08:51:11 -0500 (Thu, 15 Jul 2010) | 9 lines Merged revisions 276652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is. ........ ................ * /, channels/chan_sip.c: Merged revisions 276571 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r276571 | jpeeler | 2010-07-14 17:58:24 -0500 (Wed, 14 Jul 2010) | 21 lines Fix MWI notification transmission problems over SIP. MWI updates were not being sent if no messages were found in the event cache. This was corrected since a phone may need to clear its MWI status configured previously from another mailbox. Upon module or sip reload, MWI updates could not be sent due to the sipsock socket not being set early enough in reload_config. The code handling the descriptor assignment and such has simply been moved before the call to build_peer. Issuing a sip reload cleared the IP address of the peer, but skipped checking the database for registration information. The database is now checked both for sip reload and actually reloading the module. If a transmission occurs before the do_monitor thread has started, do not attempt to send a signal to it. (closes issue #17398) Reported by: ip-rob ........ 2010-07-14 20:16 +0000 [r276442] Kevin P. Fleming * main/loader.c, /: Merged revisions 276441 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r276441 | kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4 lines Don't try to call an embedded module's backup_globals() function until after confirming it exists. ........ 2010-07-14 11:52 +0000 [r276269] Leif Madsen * /, configs/voicemail.conf.sample: Merged revisions 276268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r276268 | lmadsen | 2010-07-14 06:51:48 -0500 (Wed, 14 Jul 2010) | 9 lines Merged revisions 276267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010) | 1 line Update documentation for voicemail.conf externpass option. ........ ................ 2010-07-13 19:11 +0000 [r276125] Russell Bryant * /, main/features.c: Merged revisions 276124 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r276124 | russell | 2010-07-13 14:09:42 -0500 (Tue, 13 Jul 2010) | 9 lines Merged revisions 276123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr instead of peer_cdr in the last commit). ........ ................ 2010-07-13 19:01 +0000 [r276121] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 276074 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010) | 19 lines Merged revisions 275773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines Make user removals and traversals thread safe in meetme. Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ ........ ................ 2010-07-13 16:55 +0000 [r275996] Russell Bryant * /, main/features.c: Merged revisions 275995 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275995 | russell | 2010-07-13 11:53:44 -0500 (Tue, 13 Jul 2010) | 21 lines Merged revisions 275994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) | 14 lines Access peer->cdr directly instead of through a saved off reference. At this point in the code, it is possible that peer_cdr may be invalid. Specifically, in the blind transfer code, CDRs are swapped between channels. So, peer_cdr is no longer == peer->cdr. The scenario that exposed a crash in this code was a blind transfer that hit the system call limit, causing the transferee channel to get destroyed after the transfer attempt failed. Even if it succeeds and this code doesn't crash, this code was still trying to reset a CDR on a channel that was now owned by a different thread, which is a BadThing(tm). (ABE-2417) ........ ................ 2010-07-13 14:49 +0000 [r275911] Tilghman Lesher * contrib/realtime/mysql, contrib/realtime/oracle, contrib/scripts/sip-friends.sql (removed), contrib/realtime/mysql/sipfriends.sql, contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql (removed), contrib/realtime/mysql/meetme.sql, contrib/realtime/sqlserver, contrib/scripts/realtime_pgsql.sql (removed), contrib/scripts/iax-friends.sql (removed), /, contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql (removed), contrib/realtime (added), contrib/realtime/postgresql, contrib/realtime/postgresql/realtime.sql: Merged revisions 275910 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275910 | tilghman | 2010-07-13 09:48:40 -0500 (Tue, 13 Jul 2010) | 9 lines Merged revisions 275909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 Jul 2010) | 2 lines Move SQL scripts into their own database-specific directories. ........ ................ 2010-07-12 17:26 +0000 [r275706] Jeff Peeler * main/channel.c, /: Merged revisions 275682 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275682 | jpeeler | 2010-07-12 12:21:01 -0500 (Mon, 12 Jul 2010) | 18 lines Merged revisions 275665 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) | 11 lines Change ast_write to not stop generator when called from ast_prod. For SIP channels configured with the progressinband option on, the ringback was being immediately stopped. This problem was due to ast_prod being moved for a deadlock fix in 259858. Prodding the channel after setting up the generator triggered the check in ast_write to stop the generator. The fix here should write the frame the same as was done before the call to ast_prod was moved. (closes issue #17372) Reported by: tech_admin ........ ................ 2010-07-12 15:38 +0000 [r275627] Leif Madsen * cdr/cdr_pgsql.c, /: Merged revisions 275626 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r275626 | lmadsen | 2010-07-12 10:37:01 -0500 (Mon, 12 Jul 2010) | 11 lines cdr_pgsql does not detect when a table is found. This change adds an ERROR message to let you know when a failure exists to get the columns from the pgsql database, which typically means that the table does not exist. (closes issue #17478) Reported by: kobaz Patches: cdr_pgsql.patch uploaded by kobaz (license 834) Tested by: kobaz, russell, lmadsen ........ 2010-07-10 15:11 +0000 [r275311-275469] Russell Bryant * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 245192 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245192 | mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21 lines Remove useless sip options related to hash table size. First off, these options weren't actually doing anything. By the time the options were parsed, the peer and dialog containers had already been allocated with their default values. Second, hash table size is something that doesn't really make sense to change in a config file. If a user is that interested in changing the hashtable size, he can modify the source itself. I have removed the parsing of the hash_peer, hash_user, and hash_dialog options. I have removed the hash_user_size variable altogether since it is not used at all. I also changed hash_peer_size and hash_dialog_size to be constant, and have changed the symbols to be in all caps as constants typically are. I have also removed the entire section in sip.conf.sample regarding configurable hashtable sizes. ........ (merge to 1.6.2 inspired by issue #17553) * /: unblock a rev * configs/features.conf.sample, /, main/features.c: Merged revisions 275424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r275424 | russell | 2010-07-09 16:57:21 -0500 (Fri, 09 Jul 2010) | 27 lines Fix some issues related to dynamic feature groups in features.conf. The bridge handling code did not properly consider feature groups when setting parameters that would affect whether or not a native bridge would be attempted. If DYNAMIC_FEATURES only include a feature group, a native bridge would occur that may prevent features from working. Fix a bug in verbose output that would show the key mapping as empty if it was using the default mapping and not a custom mapping in the feature group. Add feature groups to the output of "features show". Adjust the feature execution logic to match that of the logic when executing a feature that was not configured through a feature group. Update features.conf.sample to show that an '=' is still required if using the default key mapping from [applicationmap]. Finally, clean up a little bit of formatting to better coform to coding guidelines while in the area. (closes issue #17589) Reported by: lmadsen Patches: issue_17589.rev4.txt uploaded by russell (license 2) Tested by: russell, lmadsen ........ * /, main/features.c: Merged revisions 275310 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r275310 | russell | 2010-07-09 14:58:06 -0500 (Fri, 09 Jul 2010) | 2 lines Add missing ao2_iterator_destroy(). ........ 2010-07-09 19:23 +0000 [r275260] Paul Belanger * /, channels/chan_sip.c: Merged revisions 275249 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275249 | pabelanger | 2010-07-09 15:21:27 -0400 (Fri, 09 Jul 2010) | 15 lines Merged revisions 275241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul 2010) | 8 lines Fix logging message for stale nonce. (closes issue #17582) Reported by: kenner Patches: chan_sip.c.diff uploaded by kenner (license 1040) Tested by: lmadsen ........ ................ 2010-07-09 18:24 +0000 [r275191] Matthew Nicholson * main/loader.c, /: Merged revisions 275186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275186 | mnicholson | 2010-07-09 13:24:03 -0500 (Fri, 09 Jul 2010) | 9 lines Merged revisions 275182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul 2010) | 2 lines give a better error message when attempting to unload a module that is not loaded ........ ................ 2010-07-09 18:11 +0000 [r275148] Russell Bryant * configs/features.conf.sample, /: Merged revisions 275147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r275147 | russell | 2010-07-09 13:11:13 -0500 (Fri, 09 Jul 2010) | 2 lines Move parking lot sample config out from the middle of dynamic features sample config. ........ 2010-07-09 17:51 +0000 [r275029-275145] Matthew Nicholson * main/loader.c, /: Merged revisions 275144 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul 2010) | 9 lines Merged revisions 275143 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul 2010) | 2 lines don't unload modules that returned AST_MODULE_LOAD_DECLINE when they were loaded ........ ................ * apps/app_dial.c, /: Merged revisions 275028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul 2010) | 15 lines Merged revisions 275027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial (closes issue #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson ........ ................ 2010-07-09 15:39 +0000 [r275023] Russell Bryant * include/asterisk/test.h, /, main/test.c: Merged revisions 275022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r275022 | russell | 2010-07-09 10:35:53 -0500 (Fri, 09 Jul 2010) | 11 lines Merged revisions 275021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines Document that a leading and trailing slash is expected for test categories. Also, emit a warning if a test is registered without one of these. ........ ................ 2010-07-07 18:34 +0000 [r274627-274640] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 274639 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r274639 | rmudgett | 2010-07-07 13:32:35 -0500 (Wed, 07 Jul 2010) | 1 line Add missing conditional around chan_dahdi mfcr2_skip_category config parameter. ........ * channels/chan_dahdi.c, /: Merged revisions 274595 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274595 | rmudgett | 2010-07-07 13:20:00 -0500 (Wed, 07 Jul 2010) | 9 lines Merged revisions 274579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07 Jul 2010) | 1 line Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter. ........ ................ 2010-07-07 06:16 +0000 [r274419] Tilghman Lesher * configs/say.conf.sample, /: Merged revisions 274418 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274418 | tilghman | 2010-07-07 01:15:43 -0500 (Wed, 07 Jul 2010) | 15 lines Merged revisions 274417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010) | 8 lines Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers. (closes issue #16102) Reported by: Delvar Patches: say.conf.fix.patch uploaded by Delvar (license 908) (plus a few additional fixes and simplifications by me) ........ ................ 2010-07-06 23:06 +0000 [r274360] Terry Wilson * configs/sip.conf.sample, channels/chan_sip.c: Merged revisions 274284 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010) | 18 lines Merged revisions 274280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines Add option to not do a call forward on 482 Loop Detected Asterisk has always set up a forwarded call when receiving a 482 Loop Detected. This prevents handling the call failure by just continuing on in the dialplan. Since this would be a change in behavior, the new option to disable this behavior is forwardloopdetected which defaults to 'yes'. Review: https://reviewboard.asterisk.org/r/764/ ........ ................ 2010-07-06 22:30 +0000 [r274347] Jeff Peeler * configs/sip.conf.sample, /: Merged revisions 274316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274316 | jpeeler | 2010-07-06 17:23:35 -0500 (Tue, 06 Jul 2010) | 14 lines Merged revisions 274283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines Correct sip.conf.sample comments for prematuremedia option. (closes issue #17513) Reported by: festr Patches: patch uploaded by festr (license 443) ........ ................ 2010-07-06 22:10 +0000 [r274282] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 274281 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r274281 | tilghman | 2010-07-06 17:09:23 -0500 (Tue, 06 Jul 2010) | 2 lines Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the bitfield. ........ 2010-07-06 14:33 +0000 [r274168] Mark Michelson * main/rtp.c, /: Merged revisions 274164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r274164 | mmichelson | 2010-07-06 09:31:13 -0500 (Tue, 06 Jul 2010) | 22 lines Merged revisions 274157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines Fix problem with RFC 2833 DTMF not being accepted. A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order. The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the seqno rolling over will not cause us to stop accepting DTMF. (closes issue #17571) Reported by: mdeneen Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license 60) Tested by: richardf, maxochoa, JJCinAZ ........ ................ 2010-07-05 13:55 +0000 [r273888] Paul Belanger * main/config.c, /: Merged revisions 273886 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273886 | pabelanger | 2010-07-05 09:53:44 -0400 (Mon, 05 Jul 2010) | 15 lines Merged revisions 273884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul 2010) | 8 lines Remove extra line breaks from 'core show config mappings' (closes issue #17583) Reported by: pabelanger Patches: issue17583.patch uploaded by pabelanger (license 224) Tested by: lmadsen ........ ................ 2010-07-03 02:43 +0000 [r273716-273831] Tilghman Lesher * channels/chan_local.c, /, channels/chan_agent.c, channels/chan_h323.c, include/asterisk/lock.h: Merged revisions 273830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010) | 16 lines Merged revisions 273793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ ................ * main/autoservice.c, /: Merged revisions 273718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273718 | tilghman | 2010-07-02 12:10:59 -0500 (Fri, 02 Jul 2010) | 15 lines Merged revisions 273717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup. (closes issue #17564) Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ ................ * apps/app_queue.c, /: Merged revisions 273714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273714 | tilghman | 2010-07-02 11:57:28 -0500 (Fri, 02 Jul 2010) | 2 lines The switch fallthrough could create some errorneous situations, so best to force directly to the default case. ........ 2010-07-02 15:59 +0000 [r273642] Tzafrir Cohen * channels/chan_iax2.c, apps/app_voicemail.c, channels/chan_dahdi.c, channels/chan_sip.c, res/res_agi.c: Fix typos reported by Lintian 2010-07-01 22:17 +0000 [r273571] Russell Bryant * main/datastore.c, /: Merged revisions 273566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273566 | russell | 2010-07-01 17:16:23 -0500 (Thu, 01 Jul 2010) | 14 lines Merged revisions 273565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines Don't return a partially initialized datastore. If memory allocation fails in ast_strdup(), don't return a partially initialized datastore. Bad things may happen. (related to ABE-2415) ........ ................ 2010-07-01 20:29 +0000 [r273356-273529] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 273522 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273522 | jpeeler | 2010-07-01 15:28:15 -0500 (Thu, 01 Jul 2010) | 21 lines Merged revisions 273474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines Allow admin user to join conference without using admin mode and no user pin. Configuring the conference in meetme.conf like the following: conf => 2345,,6666 did not prompt for pin when used without admin mode. This meant that the conference could not be joined as an admin even if the user knew the correct pin. The original bug report was submitted claiming that the blank user pin should deny entry into the conference. I think a better way to handle this would be with a feature enhancement that used the following syntax: conf => 2345,X,6666 - where X denotes no acceptable pin allowed (closes issue #15704) Reported by: modelnine ........ ................ * /, apps/app_meetme.c: Merged revisions 273355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273355 | jpeeler | 2010-07-01 10:12:31 -0500 (Thu, 01 Jul 2010) | 19 lines Merged revisions 273354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. Specifically checking for this scenario by looking at the appropriate control frames resolves the issue. (closes issue #15871) Reported by: Ivan Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229) ........ ................ 2010-07-01 14:39 +0000 [r273271-273353] Matthew Nicholson * main/manager.c, /: Merged revisions 273352 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273352 | mnicholson | 2010-07-01 09:37:37 -0500 (Thu, 01 Jul 2010) | 2 lines Fixed whitespace problems ........ * main/manager.c, /: Merged revisions 273350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273350 | mnicholson | 2010-07-01 09:34:31 -0500 (Thu, 01 Jul 2010) | 2 lines Altered my comment about TCP_NODELAY ........ * main/manager.c, /: Merged revisions 273270 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273270 | mnicholson | 2010-06-30 13:48:21 -0500 (Wed, 30 Jun 2010) | 2 lines Set TCP_NODELAY on manager TCP sockets to prevent delays on outgoing packets. This regression was introduced in r48338. AST-359 ........ 2010-06-30 17:32 +0000 [r273193-273234] Paul Belanger * main/rtp.c, /: Merged revisions 273233 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273233 | pabelanger | 2010-06-30 13:28:04 -0400 (Wed, 30 Jun 2010) | 11 lines Fix rt(c)p set debug ip taking wrong argument Also clean up some coding errors. (closes issue #17469) Reported by: wdoekes Patches: astsvn-rtp-set-debug-ip.patch uploaded by wdoekes (license 717) Tested by: wdoekes, pabelanger ........ * /: Revert previous commit; res_rtp_asterisk.c does not exist. * /: Unblock revisions 218107 ........ r218107 | mvanbaak | 2009-09-12 15:08:16 +0200 (Sat, 12 Sep 2009) | 8 lines use the actual given ip address for 'rtp set debug ip ' instead of the word 'ip' (closes issue 0015711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) Tested by: davidw ........ 2010-06-30 01:07 +0000 [r273056-273145] Tilghman Lesher * main/manager.c, /: Merged revisions 273144 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273144 | tilghman | 2010-06-29 20:07:02 -0500 (Tue, 29 Jun 2010) | 8 lines Permission checking for the system application is backwards. (closes issue #17550) Reported by: kenner Patches: manager.c.diff uploaded by kenner (license 1040) Tested by: kenner ........ * main/config.c, /: Merged revisions 273142 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273142 | tilghman | 2010-06-29 20:01:14 -0500 (Tue, 29 Jun 2010) | 5 lines Don't attempt to proceed if our internal parser indicates an invalid file. (closes issue #17560) Reported by: Nick_Lewis ........ * /, channels/chan_sip.c: Merged revisions 273078 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273078 | tilghman | 2010-06-29 18:20:40 -0500 (Tue, 29 Jun 2010) | 17 lines Merged revisions 273060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines Allow the "useragent" value to be restored into memory from the realtime backend. This value is purely informational. It does not alter configuration at all. (closes issue #16029) Reported by: Guggemand Patches: realtime-useragent.patch uploaded by Guggemand (license 897) Tested by: Guggemand ........ ................ * main/channel.c, /: Merged revisions 273058 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r273058 | tilghman | 2010-06-29 17:59:51 -0500 (Tue, 29 Jun 2010) | 11 lines Recorded merge of revisions 273057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) | 4 lines _Really_ skip the channel... don't just retry for another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........ ................ * main/pbx.c, /: Merged revisions 273054 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r273054 | tilghman | 2010-06-29 17:39:22 -0500 (Tue, 29 Jun 2010) | 11 lines Send DialPlanComplete as a response, not as a separate event. Otherwise, it goes to all manager sessions and may exclude the current session, if the Events mask excludes it. (closes issue #17504) Reported by: rrb3942 Patches: showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested by: rrb3942 ........ 2010-06-29 16:43 +0000 [r272972] Russell Bryant * main/asterisk.c, /: Merged revisions 253357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253357 | russell | 2010-03-18 13:18:43 -0500 (Thu, 18 Mar 2010) | 8 lines Increase CLI command output timeout for asterisk -rx to 60 seconds. (closes issue #17049) Reported by: russell Tested by: russell Review: https://reviewboard.asterisk.org/r/573/ ........ 2010-07-22 Leif Madsen * Release Asterisk 1.6.2.10 * Included a fix for res_timing_pthread per the description below: r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines Use poll() instead of select() in res_timing_pthread to avoid stack corruption. This code did not properly check FD_SETSIZE to ensure that it did not try to select() on fds that were too large. Switching to poll() removes the limitation on the maximum fd value. 2010-07-07 Leif Madsen * Release Asterisk 1.6.2.10-rc2 * Fix problem with RFC 2833 DTMF not being accepted. A recent check was added to ensure that we did not erroneously detect duplicate DTMF when we received packets out of order. The problem was that the check did not account for the fact that the seqno of an RTP stream will roll over back to 0 after hitting 65535. Now, we have a secondary check that will ensure that the seqno rolling over will not cause us to stop accepting DTMF. (closes issue 0017571) Reported by: mdeneen Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license 60) Tested by: richardf, maxochoa, JJCinAZ * Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial (closes issue 0017592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson 2010-06-29 Leif Madsen * Release Asterisk 1.6.2.10-rc1 2010-06-28 21:51 +0000 [r272924-272927] Tilghman Lesher * main/asterisk.c, /: Merged revisions 272926 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272926 | tilghman | 2010-06-28 16:50:57 -0500 (Mon, 28 Jun 2010) | 15 lines Merged revisions 272925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) | 8 lines Don't change ownership/group/permissions on run directory, if it already exists. (closes issue #17076) Reported by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by tilghman (license 14) Tested by: stuarth ........ ................ * main/config.c, /: Merged revisions 272923 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272923 | tilghman | 2010-06-28 16:42:52 -0500 (Mon, 28 Jun 2010) | 19 lines Merged revisions 272921-272922 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010) | 8 lines Change the way that we read include files, to accommodate for changes in GCC 4.4. (closes issue #17472) Reported by: seandarcy Patches: config2.patch uploaded by nivan (license 1066) Tested by: nivan ........ r272922 | tilghman | 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim trailing blanks on #includes ........ ................ 2010-06-28 18:50 +0000 [r272882] Russell Bryant * tests/test_astobj2.c (added): Backport applicable parts of test_astobj2 from trunk. 2010-06-28 17:37 +0000 [r272806] Mark Michelson * /, channels/chan_sip.c: Merged revisions 272805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272805 | mmichelson | 2010-06-28 12:33:12 -0500 (Mon, 28 Jun 2010) | 11 lines Merged revisions 272804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun 2010) | 5 lines Decode URI in contact header of 302 response. ABE-2352 ........ ................ 2010-06-28 15:36 +0000 [r272685-272686] Russell Bryant * doc/tex/chan-mobile.tex (removed): remove accidentally added file. * doc/tex/cdrdriver.tex, doc/tex/asterisk.tex, /, doc/tex/chan-mobile.tex (added): Merged revisions 272684 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272684 | russell | 2010-06-28 10:33:32 -0500 (Mon, 28 Jun 2010) | 2 lines Use the underscore package so that underscores do not need to be escaped. ........ 2010-06-25 20:20 +0000 [r272556-272577] Tilghman Lesher * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272568 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272568 | tilghman | 2010-06-25 15:18:47 -0500 (Fri, 25 Jun 2010) | 12 lines Merged revisions 272562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines Make the structure of the table specified before match the queries and results. (closes issue #17557) Reported by: cmaj ........ ................ * sounds/Makefile, /: Merged revisions 272533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272533 | tilghman | 2010-06-25 14:17:16 -0500 (Fri, 25 Jun 2010) | 2 lines Symlink sounds files, to save disk space, when multiple tarballs/checkouts are on the same system. ........ 2010-06-25 18:58 +0000 [r272531] Russell Bryant * include/asterisk/_private.h, tests/test_sched.c, main/asterisk.c, include/asterisk/test.h (added), build_tools/cflags-devmode.xml, tests/test_heap.c, tests/test_skel.c, main/Makefile, main/test.c (added): Backport unit test API from trunk. Also, update existing test modules that were already in this branch but had been converted to the unit test API in trunk. Review: https://reviewboard.asterisk.org/r/748/ 2010-06-24 22:19 +0000 [r272459] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 272447 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272447 | rmudgett | 2010-06-24 17:11:26 -0500 (Thu, 24 Jun 2010) | 17 lines Merged revisions 272446 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines ss_thread calls pri_grab without lock during overlap dial Recent changes to chan_dahdi with relation to overlap dialing call pri_grab without first obtaining a lock. (closes issue #17414) Reported by: pdf Patches: bug17414.patch uploaded by jpeeler (license 325) ........ ................ 2010-06-23 23:40 +0000 [r272440] Terry Wilson * autoconf/ast_ext_tool_check.m4, /, configure: Merged revisions 272254,272256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272254 | twilson | 2010-06-23 15:53:48 -0500 (Wed, 23 Jun 2010) | 10 lines Honor the --with-${library}=path for AST_EXT_TOOL_CHECK (closes issue #16991) Reported by: pprindeville Patches: with_netsnmp.patch.txt uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/739/ ........ r272256 | twilson | 2010-06-23 15:59:17 -0500 (Wed, 23 Jun 2010) | 2 lines Update configure when changing autconf m4 files... ........ 2010-06-23 23:14 +0000 [r272371] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 272370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272370 | russell | 2010-06-23 18:09:28 -0500 (Wed, 23 Jun 2010) | 23 lines Resolve some errors produced during module unload of chan_iax2. The external test suite stops Asterisk using the "core stop gracefully" command. The logs from the tests show that there are a number of problems with Asterisk trying to cleanly shut down. This patch addresses the following type of error that comes from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy: chan_iax2.c line 11371 (iax2_process_thread_cleanup): Error destroying mutex &thread->lock: Device or resource busy For an example in the context of a build, see: http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary purpose of this patch is to change the thread pool shutdown procedure to be more explicit to ensure that the thread exits from a point where it is not holding a lock. While testing that, I encountered various crashes due to the order of operations in unload_module() being problematic. I reordered some things there, as well. Review: https://reviewboard.asterisk.org/r/736/ ........ 2010-06-23 22:37 +0000 [r272369] Matthew Nicholson * apps/app_queue.c, /: Merged revisions 272368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272368 | mnicholson | 2010-06-23 17:36:49 -0500 (Wed, 23 Jun 2010) | 16 lines Merged revisions 272367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers. ........ r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines Send AgentComplete manager events in the event of blind and attended transfers. (closes issue #16819) Reported by: elbriga Patches: app_queue.diff uploaded by elbriga (license 482) ........ ................ 2010-06-23 21:54 +0000 [r272333] Tilghman Lesher * res/res_musiconhold.c, /: Merged revisions 272332 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272332 | tilghman | 2010-06-23 16:53:49 -0500 (Wed, 23 Jun 2010) | 8 lines If there is realtime configuration, it does not get re-read on reload unless the config file also changes. (closes issue #16982) Reported by: dmitri Patches: res_musiconhold.patch uploaded by dmitri (license 1001) Tested by: atis ........ 2010-06-23 21:15 +0000 [r272263] Paul Belanger * apps/app_meetme.c: Revert previous commit, ast_test_flag64 does not exist in 1.6.2 2010-06-23 21:09 +0000 [r272262] Tilghman Lesher * res/ael/ael.flex, /, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael_lex.c: Merged revisions 272260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272260 | tilghman | 2010-06-23 16:06:40 -0500 (Wed, 23 Jun 2010) | 8 lines Ensure a NULL file while debugging cannot crash AEL. (closes issue #17215) Reported by: vazir Patches: 20100518__issue17215.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ 2010-06-23 21:07 +0000 [r272253-272261] Paul Belanger * /, apps/app_meetme.c: Merged revisions 272259 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272259 | pabelanger | 2010-06-23 17:06:15 -0400 (Wed, 23 Jun 2010) | 2 lines Fix previous merge. ast_test_flag != ast_test_flag64 ........ * /, apps/app_meetme.c: Merged revisions 272257 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r272257 | pabelanger | 2010-06-23 17:00:00 -0400 (Wed, 23 Jun 2010) | 19 lines Merged revisions 272255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines First caller into a dynamic conference now enter pin once. If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ ................ * main/manager.c, /: Merged revisions 272252 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272252 | pabelanger | 2010-06-23 16:35:45 -0400 (Wed, 23 Jun 2010) | 8 lines Correct manager variable 'EventList' case. (closes issue #17520) Reported by: kobaz Patches: manager.patch uploaded by kobaz (license 834) Tested by: lmadsen ........ 2010-06-23 18:41 +0000 [r272124-272149] Terry Wilson * /, apps/app_meetme.c: Merged revisions 272146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272146 | twilson | 2010-06-23 13:39:20 -0500 (Wed, 23 Jun 2010) | 2 lines Don't start the sla thread unless we realy need it ........ * /, apps/app_meetme.c: Merged revisions 272109 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272109 | twilson | 2010-06-23 12:21:40 -0500 (Wed, 23 Jun 2010) | 12 lines Make sure reload updates SLA config Even if there are no stations or trunks defined, we need to start the sla thread to make sure we get the reload event. Also, when doing a reload we need to remove the existing trunks and stations or they end up hanging around. (closes issue #16818) Reported by: mbonin Patches: sla_reload.patch uploaded by twilson (license 396) Tested by: twilson ........ 2010-06-22 22:14 +0000 [r272015] David Vossel * pbx/pbx_config.c, /: Merged revisions 272014 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r272014 | dvossel | 2010-06-22 17:11:50 -0500 (Tue, 22 Jun 2010) | 5 lines fixes issue with 'dialplan remove extension blah' segfaulting with tab completion (closes issue #17440) Reported by: kobaz ........ 2010-06-22 17:37 +0000 [r271904] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 271903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun 2010) | 15 lines Merged revisions 271902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct. (closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain ........ ................ 2010-06-22 16:30 +0000 [r271869] Russell Bryant * /, res/ais/clm.c, res/ais/evt.c: Merged revisions 271867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r271867 | russell | 2010-06-22 11:28:03 -0500 (Tue, 22 Jun 2010) | 7 lines Resolve some errors that occur on a graceful shutdown. Don't Finalize() if Initialize() did not succeed. This resulted in an error about trying to Finalize() an invalid handle. Also trim some trailing whitespace while in the area. ........ 2010-06-22 15:49 +0000 [r271832] David Vossel * /, main/features.c: Merged revisions 271831 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r271831 | dvossel | 2010-06-22 10:46:22 -0500 (Tue, 22 Jun 2010) | 10 lines fixes attended transfer behavior when both transferee and transferer hung up If both the transferer and transferee of a attended transfer hangup before the new channel picks up, the new channel should be hung up as well as it has no endpoint to talk to. This mirrors the expected behavior used in 1.4. (closes issue #17444) Reported by: corruptor ........ 2010-06-22 15:00 +0000 [r271691-271763] Matthew Nicholson * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions 271762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271762 | mnicholson | 2010-06-22 09:54:58 -0500 (Tue, 22 Jun 2010) | 15 lines Merged revisions 271761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines Allow users to specify a port for dundi peers. (closes issue #17056) Reported by: klaus3000 Patches: dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ * include/asterisk/strings.h, configs/sip_notify.conf.sample, /, channels/chan_sip.c: Merged revisions 271690 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun 2010) | 18 lines Merged revisions 271689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct. (closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ ........ This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls. ................ 2010-06-21 20:48 +0000 [r271555] Jeff Peeler * res/ael/pval.c, /: Merged revisions 271554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271554 | jpeeler | 2010-06-21 15:46:53 -0500 (Mon, 21 Jun 2010) | 14 lines Merged revisions 271552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines Do not use sizeof to calculate size of a heap allocated character array. Change left out from 271399. (closes issue #16053) Reported by: diLLec ........ ................ 2010-06-18 21:33 +0000 [r271338-271484] Jeff Peeler * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged revisions 271483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010) | 18 lines Merged revisions 271399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines Fix crash when parsing some heavily nested statements in AEL on reload. Due to the recursion used when compiling AEL in gen_prios, all the stack space was being consumed when parsing some AEL that contained nesting 13 levels deep. Changing a few large buffers to be heap allocated fixed the crash, although I did not test how many more levels can now be safely used. (closes issue #16053) Reported by: diLLec Tested by: jpeeler ........ ................ * channels/chan_dahdi.c, /: Merged revisions 269307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010) | 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are in a potentially inconsistent state is a potentially bad thing in principle. However, calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. ........ 2010-06-17 Leif Madsen * Asterisk 1.6.2.9 Released. 2010-06-10 Leif Madsen * Asterisk 1.6.2.9-rc3 Released. 2010-06-10 Tilghman Lesher * Ensure signals are not blocked inside other signal handlers. This eliminates the annoying on the console. (closes issue 0017477) Reported by: jvandal Patches: 20100610__issue17477.diff.txt uploaded by tilghman (license 14 2010-06-09 Paul Belanger * Fix Debian init script to not use -c. When using the init script as-is currently, it could cause issues on Debian such as high CPU usage. This fix has worked for several people so I'm implementing the change. We now handle color displays properly. (closes issue 0016784) Reported by: pabelanger Patches: 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14) Tested by: pabelanger, tilghman 2010-06-07 Leif Madsen * Asterisk 1.6.2.9-rc2 Released. 2010-06-07 Tilghman Lesher * Fix crash in DTMF detection. What I did not originally see in my previous commit was that even though the next digit could be detected before the previous was considered ended, the detection of the next digit effectively ends the detection of the previous. Therefore, the length moves in lockstep with the digit, and no separate counter is needed for the length alone. (closes issue 0017371) Reported by: alecdavis (closes issue 0017474) Reported by: kenner 2010-06-01 Leif Madsen * Asterisk 1.6.2.9-rc1 Released. 2010-06-01 15:20 +0000 [r266598] Tilghman Lesher * main/asterisk.c, /: Merged revisions 266592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010) | 18 lines Merged revisions 266585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) | 11 lines Prevent CLI prompt from distorting output of lines shorter than the prompt. Uses the VT100 method of clearing the line from the cursor position to the end of the line: Esc-0K (closes issue #17160) Reported by: coolmig Patches: 20100531__issue17160.diff.txt uploaded by tilghman (license 14) Tested by: coolmig ........ ................ 2010-05-31 16:07 +0000 [r266570] Paul Belanger * res/res_agi.c: Fix typo in documentation (closes issue #17395) Reported by: pabelanger Patches: res_agi.c.patch uploaded by pabelanger (license 224) 2010-05-30 04:45 +0000 [r266439] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r266438 | tilghman | 2010-05-29 23:44:28 -0500 (Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 May 2010) | 2 lines Reverting patch and reopening issue #16784, as patch breaks color display. ........ ................ 2010-05-28 20:55 +0000 [r266338] Tilghman Lesher * main/asterisk.c, /: Merged revisions 266337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 | tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line Only report swap on platforms which can examine those statistics ........ 2010-05-28 17:57 +0000 [r266293] David Vossel * /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 | dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines fixes crash when creation of UDPTL fails (closes issue #17264) Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff uploaded by dvossel (license 671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel (license 671) Tested by: falves11 ........ 2010-05-26 21:19 +0000 [r266154] Tilghman Lesher * utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged revisions 266146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010) | 21 lines Merged revisions 266142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) | 14 lines Use sigaction for signals which should persist past the initial trigger, not signal. If you call signal() in a Solaris signal handler, instead of just resetting the signal handler, it causes the signal to refire, because the signal is not marked as handled prior to the signal handler being called. This effectively causes Solaris to immediately exceed the threadstack in recursive signal handlers and crash. (closes issue #17000) Reported by: rmcgilvr Patches: 20100526__issue17000.diff.txt uploaded by tilghman (license 14) Tested by: rmcgilvr ........ ................ 2010-05-26 18:37 +0000 [r266007] David Vossel * /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 | dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines fixes failed SIP Directed pickup resulting in dead channel (closes issue #17339) Reported by: one47 Patches: sip_magic_pickup2 uploaded by one47 (license 23) Tested by: one47, dvossel ........ 2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher * res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r265923 | tilghman | 2010-05-26 11:23:28 -0500 (Wed, 26 May 2010) | 14 lines Merged revisions 265910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010) | 7 lines Not finding rows in the DB does not rise to the level of a warning. (closes issue #17062) Reported by: drookie Patches: 20100525__issue17062.diff.txt uploaded by tilghman (license 14) ........ ................ * configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged revisions 265894 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 | tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines Construct socket name, according to the Postgres docs, and document as such. (closes issue #17392) Reported by: dps Patches: 20100525__issue17392.diff.txt uploaded by tilghman (license 14) Tested by: dps ........ 2010-05-26 15:52 +0000 [r265890] Mark Michelson * /, channels/chan_sip.c: Recorded merge of revisions 265842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed, 26 May 2010) | 9 lines Re-enable "always" option for videosupport option in sip.conf. (closes issue #17016) Reported by: twilson Patches: 17016.patch uploaded by mmichelson (license 60) Tested by: devmod ........ 2010-05-26 00:33 +0000 [r265748] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 | tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines Use configure to determine the prefixes and include directories properly. This ensures cross-platform compatibility, even among Linux distributions, which don't always put headers in the same place. (closes issue #17391) Reported by: loloski ........ 2010-05-25 21:05 +0000 [r265699] Mark Michelson * /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 | mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12 lines Properly use peer's outboundproxy for outbound REGISTERs. The logic used in transmit_register to get the outboundproxy for a peer was flawed since this value would be overridden shortly afterwards when create_addr was called. In addition, this also fixes some logic used when parsing users.conf so that the peer name is placed in the internally-generated register string so that an outboundproxy set in the Asterisk GUI will be used for outbound REGISTERs. ........ 2010-05-25 17:15 +0000 [r265615] David Vossel * channels/chan_dahdi.c: fixes build issue with zaptel (closes issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff uploaded by dvossel (license 671) Tested by: aragon 2010-05-25 17:06 +0000 [r265612] Matthew Nicholson * apps/app_queue.c, /: Merged revisions 265611 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May 2010) | 15 lines Merged revisions 265610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal ........ ................ 2010-05-24 23:52 +0000 [r265521] Terry Wilson * include/asterisk/options.h, main/asterisk.c, Makefile, doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged revisions 265320,265467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines Add the FullyBooted AMI event It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ ........ r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line Merge the rest of the FullyBooted patch ........ 2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson * /, channels/h323/ast_h323.cxx: Merged revisions 265451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon, 24 May 2010) | 8 lines Print openh323 log to the Asterisk console. (closes issue #17109) Reported by: under Patches: logstream.diff uploaded by under (license 914) ........ * /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 | mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11 lines Allow type=user SIP endpoints to be loaded properly from realtime. (closes issue #16021) Reported by: Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand (license 897) (altered by me slightly to avoid ref leaks) Tested by: Guggemand ........ 2010-05-24 19:30 +0000 [r265364] David Vossel * main/channel.c, /: Merged revisions 265273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 | dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines fixes segfault when using generic plc ........ 2010-05-24 18:30 +0000 [r265318] Tilghman Lesher * main/asterisk.c, /: Merged revisions 265316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 | tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines On systems with a LOT of RAM, a signed integer sometimes printed negative. (closes issue #16837) Reported by: jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by tilghman (license 14) ........ 2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson * apps/app_queue.c: Fix memory hogging behavior of app_queue. From reviewboard: This review request is for the patch on issue 17081. A user reported that he saw increasing numbers of allocations stemming from app_queue.c when he would run the "queue show" CLI command. The user reported that he was using approximately 40 realtime queues and as he ran the CLI command more and more, the memory usage would shoot up. As it turns out, there was a memory leak and a separate usage of memory that, while not really a leak, was very irresponsible. Both memory problems can be attributed to the function init_queue(). When the "queue show" command is run, all realtime queues have the init_queue() function called on the in-memory queue. The idea is to place the queue in its default state and then overwrite options specified in the realtime backend as we read them. The first problem, the memory leak, had to do with the fact that the string field for the name of the first periodic announcement file was being re-created every time init_queue was called. This patch corrects the behavior by only calling ast_str_create if the memory has not already been allocated. The other problem is a bit more complicated. The majority of the strings in the call_queue structure were changed to use the ast_string_fields API for 1.6.0 and beyond. init_queue resets all string fields on the queue to their default values. Then, later in the realtime queue loading process, these string fields are set to their configured values. For those unfamiliar with string fields, frequent resizing of a string like this is not what the string fields API is designed for. The result of this constant resizing is that as the queue gets loaded, eventually space for the string runs out and so a new memory pool, at twice the size of the previously allocated one, is created for the string fields. The reporter of issue 17081 wrote a script that ran the "queue show" CLI command 2100 times. By the end, each of his 40 queues was taking about a megabyte of memory apiece just for their string fields. My fix for this problem is to revert the call_queue structure from using string fields. In my patch here, I have moved the queue back to using fixed-sized buffers. I ran the script provided by the reporter of 17081 and determined that I no longer saw the steadily-increasing memory usage that I had seen before applying the patch. (closes issue #17081) Reported by: wliegel Patches: 17081v2.patch uploaded by mmichelson (license 60) Tested by: wliegel, mmichelson Review: https://reviewboard.asterisk.org/r/651/ * apps/app_queue.c, include/asterisk/file.h, /: Merged revisions 265090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May 2010) | 15 lines Merged revisions 265089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines Don't hang up on a queue caller if the file we attempt to play does not exist. This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill ........ ................ * /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 | mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7 lines Be sure to set the sin_family on the proxy when allocating. (closes issue #17157) Reported by: stuarth ........ * /, include/asterisk/channel.h: Merged revisions 265000 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500 (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May 2010) | 3 lines Fix grammatical error in comment. ........ ................ * main/channel.c, main/autoservice.c, /, include/asterisk/channel.h: Merged revisions 264997 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500 (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific frames until after the sleep has concluded. From reviewboard Background: A Digium customer discovered a somewhat odd bug. The setup is that parties A and B are bridged, and party A places party B on hold. While party B is listening to hold music, he mashes a bunch of DTMF. Party A takes party B off hold while this is happening, but party B continues to hear hold music. I could reproduce this about 1 in 5 times. The issue: When DTMF features are enabled and a user presses keys, the channel that the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read from the channel during the sleep, the frame is dropped. Thus the unhold indication is never made to the channel that was originally placed on hold. The fix: Originally, I discussed with Kevin possible ways of fixing the specific problem reported. However, we determined that the same type of problem could happen in other situations where ast_safe_sleep() is used. Using autoservice as a model, I modified ast_safe_sleep_conditional() to defer specific frame types so they can be re-queued once the sleep has finished. I made a common function for determining if a frame should be deferred so that there are not two identical switch blocks to maintain. Review: https://reviewboard.asterisk.org/r/674/ ........ ................ 2010-05-20 23:34 +0000 [r264829] Richard Mudgett * /, main/callerid.c: Merged revisions 264828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010) | 13 lines Merged revisions 264820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010) | 6 lines ast_callerid_parse() had a path that left name uninitialized. Several callers of ast_callerid_parse() do not initialize the name parameter before calling thus there is the potential to use an uninitialized pointer. ........ ................ 2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher * main/pbx.c, /: Merged revisions 264779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 | tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines Let ExtensionState resolve dynamic hints. (closes issue #16623) Reported by: tilghman Patches: 20100116__issue16623.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ * apps/app_stack.c, /: Merged revisions 264752 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 | tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines Error message fix. (closes issue #17356) Reported by: kenner Patches: app_stack.c.diff uploaded by kenner (license 1040) ........ 2010-05-19 22:10 +0000 [r264453] Mark Michelson * include/asterisk/_private.h, include/asterisk/options.h, main/asterisk.c, main/loader.c, main/channel.c, /, channels/chan_sip.c: Merged revisions 264452 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 | mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86 lines Fix transcode_via_sln option with SIP calls and improve PLC usage. From reviewboard: The problem here is a bit complex, so try to bear with me... It was noticed by a Digium customer that generic PLC (as configured in codecs.conf) did not appear to actually be having any sort of benefit when packet loss was introduced on an RTP stream. I reproduced this issue myself by streaming a file across an RTP stream and dropping approx. 5% of the RTP packets. I saw no real difference between when PLC was enabled or disabled when using wireshark to analyze the RTP streams. After analyzing what was going on, it became clear that one of the problems faced was that when running my tests, the translation paths were being set up in such a way that PLC could not possibly work as expected. To illustrate, if packets are lost on channel A's read stream, then we expect that PLC will be applied to channel B's write stream. The problem is that generic PLC can only be done when there is a translation path that moves from some codec to SLINEAR. When I would run my tests, I found that every single time, read and write translation paths would be set up on channel A instead of channel B. There appeared to be no real way to predict which channel the translation paths would be set up on. This is where Kevin swooped in to let me know about the transcode_via_sln option in asterisk.conf. It is supposed to work by placing a read translation path on both channels from the channel's rawreadformat to SLINEAR. It also will place a write translation path on both channels from SLINEAR to the channel's rawwriteformat. Using this option allows one to predictably set up translation paths on all channels. There are two problems with this, though. First and foremost, the transcode_via_sln option did not appear to be working properly when I was placing a SIP call between two endpoints which did not share any common formats. Second, even if this option were to work, for PLC to be applied, there had to be a write translation path that would go from some format to SLINEAR. It would not work properly if the starting format of translation was SLINEAR. The one-line change presented in this review request in chan_sip.c fixed the first issue for me. The problem was that in sip_request_call, the jointcapability of the outbound channel was being set to the format passed to sip_request_call. This is nativeformats of the inbound channel. Because of this, when ast_channel_make_compatible was called by app_dial, both channels already had compatibly read and write formats. Thus, no translation path was set up at the time. My change is to set the jointcapability of the sip_pvt created during sip_request_call to the intersection of the inbound channel's nativeformats and the configured peer capability that we determined during the earlier call to create_addr. Doing this got the translation paths set up as expected when using transcode_via_sln. The changes presented in channel.c fixed the second issue for me. First and foremost, when Asterisk is started, we'll read codecs.conf to see the value of the genericplc option. If this option is set, and ast_write is called for a frame with no data, then we will attempt to fill in the missing samples for the frame. The implementation uses a channel datastore for maintaining the PLC state and for creating a buffer to store PLC samples in. Even when we receive a frame with data, we'll call plc_rx so that the PLC state will have knowledge of the previous voice frame, which it can use as a basis for when it comes time to actually do a PLC fill-in. So, reviewers, now I ask for your help. First off, there's the one line change in chan_sip that I have put in. Is it right? By my logic it seems correct, but I'm sure someone can tell me why it is not going to work. This is probably the change I'm least concerned about, though. What concerns me much more is the set of changes in channel.c. First off, am I even doing it right? When I run tests, I can clearly see that when PLC is activated, I see a significant increase in RTP traffic where I would expect it to be. However, in my humble opinion, the audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to me than when no PLC is used at all. I need someone to review the logic I have used to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm sure someone can point out somewhere where I've done something incorrectly. As I was writing this review request up, I decided to give the code a test run under valgrind, and I find that for some reason, calls to plc_rx are causing some invalid reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around a bit to see why that is the case. If it's obvious to someone reviewing, speak up! Finally, I have one other proposal that is not reflected in my code review. Since without transcode_via_sln set, one cannot predict or control where a translation path will be up, it seems to me that the current practice of using PLC only when transcoding to SLINEAR is not useful. I recommend that once it has been determined that the method used in this code review is correct and works as expected, then the code in translate.c that invokes PLC should be removed. Review: https://reviewboard.asterisk.org/r/622/ ........ 2010-05-19 20:31 +0000 [r264405] David Vossel * main/udptl.c, /: Merged revisions 264400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 | dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines fixes infinite loop during udptl.c's decode_open_type When decode_length returns the length there is a check to see if that length is negative, if so the decode loop breaks as this means the limit has been reached. The problem here is that length is an unsigned int, so length can never be negative. This resulted in an infinite loop. (issue #17352) ........ 2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson * main/udptl.c, /: Merged revisions 264379 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 | mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4 lines Cast an unsigned int to a signed int when comparing it with 0. (AST-377) ........ * apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500 (Wed, 19 May 2010) | 12 lines Merged revisions 264334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May 2010) | 5 lines Set quieted flag when receiving a dtmf tone during playback in speechbackground. (closes issue #16966) Reported by: asackheim ........ ................ 2010-05-19 19:25 +0000 [r264332] David Vossel * /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 | dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines fixes crash in check_rtp_timeout During deadlock avoidance the sip dialog pvt is locked and unlocked. When this occurs we have no guarantee the pvt's owner is still valid. We were trying to access the pvt's owner after this without checking to see if it still existed first. (closes issue #17271) Reported by: under Patches: check_rtp_timeout.diff uploaded by under (license 914) Tested by: dvossel ........ 2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher * include/asterisk/options.h, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 264249 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010) | 24 lines Merged revisions 264248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) | 17 lines Internal timing is now on by default, if you're using DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is that this version ensures that a timer is always available, whereas in previous versions, it was possible for DAHDI to be loaded, but have no drivers to actually generate timing. If internal_timing was turned on in this circumstance, a complete lack of audio would result. This is the reason why internal_timing was not on by default. However, now that DAHDI ensures the availability of a timer, there is no reason for this setting to be off (and in fact, it solves a great many initial user problems). (closes issue #15932) Reported by: dimas Patches: 20100519__issue15932.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ * main/dsp.c, /: Merged revisions 264204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 | tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines Keep track of digit duration, when we're decoding inband to pass DTMF frames. (closes issue #17235) Reported by: frawd Patches: new_dtmf_dsp_len.patch uploaded by frawd (license 610) 20100518__issue17235.diff.txt uploaded by tilghman (license 14) Tested by: frawd ........ 2010-05-19 14:47 +0000 [r264115] David Vossel * main/rtp.c, /: Merged revisions 264114 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 | dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines fixes crash during dtmf During the processing of Cisco dtmf the dtmf samples were not being calculated correctly. In an attempt to determine what sample rate was being used, a NULL frame was processed which caused a crash. This patch resolves this. (closes issue #17248) Reported by: falves11 Patches: issue_17248.diff uploaded by dvossel (license 671) ........ 2010-05-19 08:15 +0000 [r264032] Alec L Davis * /, configs/indications.conf.sample: Merged revisions 264031 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19 May 2010) | 8 lines fix incorrectly typed indications for [nz] stutter and dialrecall (closes issue #17359) Reported by: alecdavis Patches: bug17359.diff.txt uploaded by alecdavis (license 585) ........ 2010-05-19 06:41 +0000 [r263951] Tilghman Lesher * main/dsp.c, /: Merged revisions 263950 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010) | 15 lines Merged revisions 263949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) | 8 lines Because progress is called multiple times, across several frames, we must persist states when detecting multitone sequences. (closes issue #16749) Reported by: dant Patches: dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by: dant ........ ................ 2010-05-18 22:49 +0000 [r263906] David Vossel * main/strings.c, /: Merged revisions 263904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 | dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines fixes segfault on logging (closes issue #17331) Reported by: under Patches: utils.diff uploaded by under (license 914) segfault_on_logging.diff uploaded by dvossel (license 671) Tested by: under, dvossel ........ 2010-05-18 19:41 +0000 [r263809] Jeff Peeler * apps/app_directory.c, /: Merged revisions 263807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500 (Tue, 18 May 2010) | 17 lines Merged revisions 263769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines Modify directory name reading to be interrupted with operator or pound escape. In the case of accidentally entering the wrong first three letters for the reading, users could be very frustrated if the name listing is very long. This allows interrupting the reading by pressing 0 or #. 0 will attempt to execute a configured operator (o) extension and # will exit and proceed in the dialplan. ABE-2200 ........ ................ 2010-05-17 22:10 +0000 [r263642] Mark Michelson * /, main/devicestate.c: Merged revisions 263640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May 2010) | 16 lines Merged revisions 263639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May 2010) | 10 lines Fix logic error when checking for a devstate provider. When using strsep, if one of the list of specified separators is not found, it is the first parameter to strsep which is now NULL, not the pointer returned by strsep. This issue isn't especially severe in that the worst it is likely to do is waste some cycles when a device with no '/' and no ':' is passed to ast_device_state. ........ ................ 2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010) | 9 lines With IMAP backend, messages in INBOX were counted twice for MWI. (closes issue #17135) Reported by: edhorton Patches: 20100513__issue17135.diff.txt uploaded by tilghman (license 14) 17135_2.diff uploaded by ebroad (license 878) Tested by: edhorton, ebroad ........ * main/app.c: Don't close 'n', just close 'above_n'. (closes issue #17345) Reported by: wdoekes 2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen * main/manager.c, /: Merged revisions 263457 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010) | 19 lines Recorded merge of revisions 263456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) | 11 lines Manager cookies are not compatible with RFC2109. The Version field in the cookies we're setting contain quotes around the version number which is not compatible with RFC2109 and breaks some implementations. (closes issue #17231) Reported by: ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by ecarruda (license 559) Tested by: ecarruda, russell ........ ................ * sounds/Makefile, /: Merged revisions 263375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010) | 16 lines Merged revisions 263374 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010) | 8 lines Update link to new version of core sounds. The latest version of the core sounds files 1.4.19 now includes the missing queue-minute sound file which is called by app_queue but which has been missing. (closes issue #17123) Reported by: n8ideas ........ ................ 2010-05-17 13:03 +0000 [r263293] David Vossel * CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer policy dialstring option 2010-05-15 23:41 +0000 [r263202] Paul Belanger * /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 | tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines Make the Makefile logic more explicit and move the Snow Leopard logic down to where it's not executed on non-Darwin systems. (closes issue #17028) Reported by: pabelanger Patches: issue17028_20100315.patch uploaded by seanbright (license 71) 20100315__issue17028.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, pabelanger ........ 2010-05-13 22:13 +0000 [r263070] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010) | 1 line Fix inverted logic in cli command: ss7 set debug on/off ........ 2010-05-13 15:36 +0000 [r262898] Russell Bryant * channels/chan_console.c, /: Merged revisions 262897 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010) | 4 lines Fix an off by one error that causes a crash. Thanks to Raymond Burke for pointing it out. ........ 2010-05-12 20:01 +0000 [r262801] Paul Belanger * main/loader.c, main/cli.c, /: Merged revisions 262800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed, 12 May 2010) | 8 lines Notify CLI when modules is loaded / unloaded (closes issue #17308) Reported by: pabelanger Patches: cli.modules.patch uploaded by pabelanger (license 224) Tested by: pabelanger, russell ........ 2010-05-12 19:53 +0000 [r262797-262799] Leif Madsen * res/ael/pval.c, /: Merged revisions 262798 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 | lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines Revert previous WARNING message removal. Marquis42 suggested a better method of doing what I wanted because I ended up removing the WARNING message for all instances when really I just wanted to remove it for the 'return' keyword, not everything. (issue #17145) ........ * res/ael/pval.c, /: Merged revisions 262796 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 | lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines Remove unnecessary WARNING message in ael/pval.c (closes issue #17145) Reported by: okrief ........ 2010-05-12 18:03 +0000 [r262746] David Vossel * /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010) | 17 lines Merged revisions 262662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines fixes app_meetme dsp error We attempted to detect silence after translating a frame from signed linear. This caused a flooding of errors. To resolve this the code to detect silence was moved before the translation. (closes issue #17133) Reported by: jsdyer ........ ................ 2010-05-12 16:29 +0000 [r262516-262659] Tilghman Lesher * /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 | tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines Ensure the arguments are initialized. Also miscellaneous CG cleanup. (closes issue #16576) Reported by: uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman (license 14) Tested by: uxbod ........ * /, include/asterisk/causes.h: Merged revisions 262513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11 May 2010) | 7 lines Move cause 200 to cause 26, as specified in Q.850. Also cleanup the formatting and add a few more that seem like good candidates. (closes issue #16157) Reported by: wimpy ........ 2010-05-11 19:58 +0000 [r262425] Jason Parker * /, res/Makefile: Merged revisions 262422 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) | 18 lines Merged revisions 262421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines Use a less silly method for modifying a flex-generated file. The sed syntax that was used wasn't actually valid, causing some versions to choke. This is the method that is used in 1.6.x+ for similar changes. (closes issue #16696) Reported by: bklang Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested by: qwell ........ ................ 2010-05-11 19:41 +0000 [r262415-262420] Paul Belanger * pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 | pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8 lines Improve logging by displaying line number (closes issue #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger ........ * /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 | pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8 lines Improve logging information for misconfigured contexts (closes issue #17238) Reported by: pprindeville Patches: chan_sip-bug17238.patch uploaded by pprindeville (license 347) Tested by: pprindeville ........ 2010-05-11 17:25 +0000 [r262340] Tilghman Lesher * apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262330 | tilghman | 2010-05-11 12:23:51 -0500 (Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010) | 2 lines Fix issue #17302 a slightly different way (mad props to Qwell) ........ ................ 2010-05-10 19:06 +0000 [r262237-262241] David Vossel * /, apps/app_directed_pickup.c: Merged revisions 262240 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10 May 2010) | 9 lines fixes PickupChan application (closes issue #16863) Reported by: schern Patches: app_directed_pickup.c.patch uploaded by schern (license 995) for_trunk.diff uploaded by cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama, rickead2000, dvossel ........ * channels/chan_console.c, /: Merged revisions 262236 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010) | 11 lines fixes crash in chan_console There is a race condition between console_hangup() and start_stream(). It is possible for console_hangup() to be called and then the stream thread to begin after the hangup. To avoid this a check in start_stream() to make sure the pvt-owner still exists while the pvt lock is held is made. If the owner is gone that means the channel hung up and start_stream should be aborted. ........ 2010-05-10 16:39 +0000 [r262155] Tilghman Lesher * /, Makefile.rules: Merged revisions 262152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010) | 17 lines Merged revisions 262151 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010) | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes issue #17297) Reported by: jcovert Patches: 20100506__issue17297.diff.txt uploaded by tilghman (license 14) (closes issue #17302) Reported by: jcovert ........ ................ 2010-05-09 02:17 +0000 [r261916-262105] Tilghman Lesher * autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4, autoconf/ast_c_define_check.m4, /, configure, include/asterisk/autoconfig.h.in: Merged revisions 262102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08 May 2010) | 5 lines Cleanup a bit more by getting rid of useless version defines. Also make library detection use passed CFLAGS. (closes issue #17309) Reported by: stuarth ........ * /, configure, configure.ac: Merged revisions 262048 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010) | 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only ........ * /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 | tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines Double free crash (closes issue #17245) Reported by: thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by tilghman (license 14) Tested by: murraytm ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 261913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 | tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14 lines Use the detected pthread building flags in every place, instead of hardcoding -lpthread. We nicely detect the right flags on each system for building Asterisk with pthreads, then ignore it for every other build option that requires us to build with pthreads. This caused some items to return a false negative. Also cleanup some minor naming issues that caused "library library" redundancy in the output. (closes issue #17303) Reported by: stuarth Patches: 20100507__issue17303.diff.txt uploaded by tilghman (license 14) Tested by: stuarth ........ 2010-05-07 16:08 +0000 [r261868] Leif Madsen * UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 | lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines Update UPGRADE-1.6.txt stating insecure=very has been removed. (closes issue #17282) Reported by: stuarth Tested by: stuarth ........ 2010-05-06 20:13 +0000 [r261739] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500 (Thu, 06 May 2010) | 15 lines Merged revisions 261735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines Only allow the operator key to be accepted after leaving a voicemail. Or rather disallow the operator key from being accepted when not offered, such as after finishing a recording from within the mailbox options menu. ABE-2121 SWP-1267 ........ ................ 2010-05-06 17:08 +0000 [r261612] Jason Parker * sounds/Makefile, /: Merged revisions 261609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) | 11 lines Merged revisions 261608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) | 4 lines Use the versioned MOH tarballs, now that we have them. This makes for more reproducibility. Prompted by a discussion in #asterisk-dev ........ ................ 2010-05-06 15:43 +0000 [r261563] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 | tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines Permit more lines within a SIP body to be parsed. The example given within the related issue showed 120 lines, which was mostly a result of the body being XML. (closes issue #17179) Reported by: khw ........ 2010-06-01 Leif Madsen * Asterisk 1.6.2.8 Released. 2010-05-26 Leif Madsen * Asterisk 1.6.2.8-rc2 Released. 2010-05-26 10:56 -0500 [r265891] Matt Nicholson * Merged r265610 from 1.4: Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal 2010-05-06 Leif Madsen * Asterisk 1.6.2.8-rc1 Released 2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant * tests/test_heap.c: Add test case that ensures the heap handles arbitrary removals properly. (issue #17277) Reported by: cappucinoking Patches: test_heap.diff uploaded by cappucinoking (license 1036) Tested by: cappucinoking, russell * /, main/heap.c: Merged revisions 261496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 | russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines Fix handling of removing nodes from the middle of a heap. This bug surfaced in 1.6.2 and does not affect code in any other released version of Asterisk. It manifested itself as SIP qualify not happening when it should, causing peers to go unreachable. This was debugged down to scheduler entries sometimes not getting executed when they were supposed to, which was in turn caused by an error in the heap code. The problem only sometimes occurs, and it is due to the logic for removing an entry in the heap from an arbitrary location (not just popping off the top). The scheduler performs this operation frequently when entries are removed before they run (when ast_sched_del() is used). In a normal pop off of the top of the heap, a node is taken off the bottom, placed at the top, and then bubbled down until the max heap property is restored (see max_heapify()). This same logic was used for removing an arbitrary node from the middle of the heap. Unfortunately, that logic is full of fail. This patch fixes that by fully restoring the max heap property when a node is thrown into the middle of the heap. Instead of just pushing it down as appropriate, it first pushes it up as high as it will go, and _then_ pushes it down. Lastly, fix a minor problem in ast_heap_verify(), which is only used for debugging. If a parent and child node have the same value, that is not an error. The only error is if a parent's value is less than its children. A huge thanks goes out to cappucinoking for debugging this down to the scheduler, and then producing an ast_heap test case that demonstrated the breakage. That made it very easy for me to focus on the heap logic and produce a fix. Open source projects are awesome. (closes issue #16936) Reported by: ib2 Tested by: cappucinoking, crjw (closes issue #17277) Reported by: cappucinoking Patches: heap-fix.rev2.diff uploaded by russell (license 2) Tested by: cappucinoking, russell ........ 2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen * channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) | 4 lines When failing to configure, don't destroy 'cfg' twice Fixes a crash when some config section had an incorrect channel config. ........ 2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May 2010) | 19 lines Merged revisions 261274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May 2010) | 12 lines Registration fix for SIP realtime. Make sure realtime fields are not empty. (closes issue #17266) Reported by: Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis, sberney Review: https://reviewboard.asterisk.org/r/643/ ........ ................ * apps/app_queue.c, /: Merged revisions 261232 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 | pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10 lines 'queue reset stats' erroneously clears wrapuptime configuration. Resets each member's lastcall to 0 now. (closes issue #17262, #16519) Reported by: rain Patches: wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested by: rain ........ 2010-05-04 23:55 +0000 [r261098] Tilghman Lesher * main/channel.c, /: Merged revisions 261095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010) | 18 lines Merged revisions 261093-261094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010) | 7 lines Protect against overflow, when calculating how long to wait for a frame. (closes issue #17128) Reported by: under Patches: d.diff uploaded by under (license 914) ........ r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2 lines Add a tiny corner case to the previous commit ........ ................ 2010-05-04 19:01 +0000 [r260927] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500 (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) | 12 lines Voicemail transfer to operator should occur immediately, not after main menu. There were two scenarios in the advanced options that while using the operator=yes and review=yes options, the transfer occurred only after exiting the main menu (after sending a reply or leaving a message for an extension). Now after the audio is processed for the reply or message the transfer occurs immediately as expected. ABE-2107 ABE-2108 ........ ................ 2010-05-04 16:58 +0000 [r260884] Matthew Nicholson * configs/sip.conf.sample, include/asterisk/frame.h, main/channel.c, /, channels/chan_sip.c: Merged revisions 254450 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25 Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that arrive before a T.38-capable application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ ........ 2010-05-04 15:51 +0000 [r260746-260805] Jason Parker * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260802 | qwell | 2010-05-04 10:49:57 -0500 (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May 2010) | 1 line Fix fallout from removing from configure script. Pointed out by philipp64 on #asterisk-dev ........ ................ * /: Fix merge props 2010-05-03 17:42 +0000 [r260743] Paul Belanger * Makefile, /: Merged revisions 260661-260662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend libdir when executing mkpkgconfig allowing non-root installs to work. (closes issue #17268) Reported by: pabelanger Patches: issue17268.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41 -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/ part. Thanks Qwell. ........ 2010-05-03 14:59 +0000 [r260571] Leif Madsen * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500 (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010) | 1 line Minor typo pointed out by pabelanger on IRC. ........ ................ 2010-04-30 22:48 +0000 [r260441] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500 (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines Ensure channel state is not incorrectly set in the case of a very early answer. The needringing bit was being read in dahdi_read after answering thereby setting the state to ringing from up. This clears needringing upon answering so that is no longer possible. (closes issue #17067) Reported by: tzafrir Patches: needringing.diff uploaded by tzafrir (license 46) ........ ................ 2010-04-30 20:22 +0000 [r260373] Mark Michelson * res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500 (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan->stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 ........ ................ 2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher * /, main/app.c: Merged revisions 260292 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 | tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13 lines Don't allow file descriptors to go above 64k, when we're closing them in a fork(2). This saves time, when, even though the system allows the process limit to be that high, the practical limit is much lower. (closes issue #17223) Reported by: dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by tilghman (license 14) Tested by: dbackeberg ........ * configs/extensions.conf.sample, /: Merged revisions 260280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30 Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan context. (closes issue #17263) Reported by: pprindeville Patches: freenum-dialplan.patch#3 uploaded by pprindeville (license 347) ........ 2010-04-29 23:13 +0000 [r260234] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500 (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines DTMF CallerID detection problems. The code handling DTMF CallerID drops digits on long CallerID numbers and may timeout waiting for the first ring with shorter numbers. The DTMF emulation mode was not turned off when processing DTMF CallerID. When the emulation code gets behind in processing the DTMF digits it can skip a digit. For shorter numbers, the timeout may have been too short. I increased it from 2 seconds to 4 seconds. Four seconds is a typical time between rings for many countries. (closes issue #16460) Reported by: sum Patches: issue16460.patch uploaded by rmudgett (license 664) issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA AST-334 JIRA SWP-901 ........ ................ 2010-04-29 18:18 +0000 [r260156] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 260148 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29 Apr 2010) | 2 lines Pattern match fail. ........ 2010-04-29 15:35 +0000 [r260051] David Vossel * main/audiohook.c, /, include/asterisk/audiohook.h: Merged revisions 260050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010) | 21 lines Merged revisions 260049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) | 14 lines Fixes crash in audiohook_write_list The middle_frame in the audiohook_write_list function was being freed if a audiohook manipulator returned a failure. This is incorrect logic. This patch resolves this and adds detailed descriptions of how this function should work and why manipulator failures must be ignored. (closes issue #17052) Reported by: dvossel Tested by: dvossel (closes issue #16196) Reported by: atis Review: https://reviewboard.asterisk.org/r/623/ ........ ................ 2010-04-28 22:36 +0000 [r259959] Mark Michelson * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 | mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11 lines Don't override peer context with domain context. (closes issue #17040) Reported by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347) Tested by: pprindeville Review: https://reviewboard.asterisk.org/r/565/ ........ 2010-04-28 21:26 +0000 [r259899] David Vossel * main/channel.c, channels/chan_local.c, /: Merged revisions 259870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500 (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) | 33 lines resolves deadlocks in chan_local Issue_1. In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock avoidance is done when the channel to hangup is the outbound chan_local channel, but when it is not the outbound channel we have an issue... We attempt to do deadlock avoidance only on the tech pvt, when both the tech pvt and the pvt->owner are locked coming into that loop. By never giving up the pvt->owner channel deadlock avoidance is not entirely possible. This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is used in ast_activate_generator() to queue a frame on the channel and make the channel's read function get called. This function is used in ast_activate_generator() while the channel is locked, which mean's the channel will have a lock both from the generator code and the frame_queue code by the time it gets to chan_local.c's local_queue_frame code... local_queue_frame contains some of the same crazy deadlock avoidance that local_hangup requires, and this recursive lock prevents that deadlock avoidance from happening correctly. This patch removes ast_prod() from the channel lock so only one lock is held during the local_queue_frame function. (closes issue #17185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ ........ ................ 2010-04-28 21:09 +0000 [r259854] Leif Madsen * config.guess: Merged revisions 259853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010) | 14 lines Merged revisions 259852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010) | 6 lines Update config.guess. Updating config.guess because after installing Ubuntu Server 9.10 and running all the update scripts, running ./configure would not continue because it was unable to determine what kind of system I had. After updating config.guess things started working again. ........ ................ 2010-04-28 20:34 +0000 [r259781-259851] Jason Parker * /, configure, configure.ac: Merged revisions 259848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259848 | qwell | 2010-04-28 15:32:14 -0500 (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so systems without install can use install-sh from our source dir. ........ ................ * makeopts.in, /: Merged revisions 259837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) | 9 lines Merged revisions 259833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) | 1 line Missed this when removing $ID ........ ................ * Makefile, /, configure, configure.ac: Merged revisions 259760 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259760 | qwell | 2010-04-28 14:19:54 -0500 (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) | 7 lines Remove usage of `id` since it isn't useful and was causing breakge. Solaris `id` doesn't support the -u argument. Instead of figuring out how to fix this to work on Solaris, I decided to check why it was necessary and where else it was used. It was only used in one place, and it hasn't been needed for a very long time (I question whether it was ever needed). ........ ................ 2010-04-28 17:19 +0000 [r259681] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500 (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010) | 4 lines Do not play goodbye prompt after timeout of message review. ABE-2124 ........ ................ 2010-04-27 22:46 +0000 [r259616] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500 (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success" Changed the warning to "Failed to decode CallerID on channel 'name'". The message before it is likely more specific about why the CallerID decode failed. SWP-501 AST-283 ........ ................ 2010-04-27 21:50 +0000 [r259528] Leif Madsen * sounds/Makefile: Merged revisions 259527 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010) | 23 lines Merged revisions 259526 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010) | 15 lines Update sounds files. * Add additional sounds prompts for say_enumeration * Update the English conference sounds prompts so they are better quality and all sound more consistent * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files to include all present sound files Both core (en, fr, es) and extra (en, fr) sounds files have been updated. (closes issue #16200) Reported by: murf (closes issue #17137) Reported by: lmadsen ........ ................ 2010-04-27 21:25 +0000 [r259356-259486] Jason Parker * main/editline/configure.in, /, main/editline/configure, main/editline/Makefile.in: Merged revisions 259439 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) | 5 lines Add gar to the check for AR for those silly OSes (Solaris) that don't have ar. autoconf2.13 couldn't handle AC_PROG_GREP, so I removed it. This is fine, since we don't need to use anything that the configure script doesn't. ........ * /: Unblock revision 259439. * /, configure, configure.ac: Merged revisions 259353 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259353 | qwell | 2010-04-27 14:31:55 -0500 (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) | 5 lines Support the silly OSes that don't have ar and strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to AC_CHECK_TOOLS. ........ ................ 2010-04-27 19:03 +0000 [r259310] Richard Mudgett * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 259307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010) | 21 lines Merged revisions 259270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ ................ 2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson * main/channel.c, /: Merged revisions 259105 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr 2010) | 9 lines Merged revisions 259104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr 2010) | 3 lines Let compilation succeed warning-free when DONT_OPTIMIZE is turned off. ........ ................ * main/channel.c, /: Merged revisions 259023 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr 2010) | 19 lines Merged revisions 259018 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr 2010) | 13 lines Prevent Newchannel manager events for dummy channels. No Newchannel manager event will be fired for channels that are allocated to not match a registered technology type. Thus bogus channels allocated solely for variable substitution or CDR operations do not result in a Newchannel event. (closes issue #16957) Reported by: atis Review: https://reviewboard.asterisk.org/r/601 ........ ................ 2010-04-26 16:00 +0000 [r258935] Leif Madsen * /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 | lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines Small error in the T.140 RTP port verbose log. (closes issue #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610) Tested by: russell ........ 2010-04-25 18:14 +0000 [r258779] Tilghman Lesher * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010) | 13 lines Merged revisions 258775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) | 6 lines When StopMonitor is called, ensure that it will not be restarted by a channel event. (closes issue #16590) Reported by: kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888) ........ ................ 2010-04-22 22:15 +0000 [r258676] Matthew Nicholson * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions 258671,258675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May 2009) | 8 lines Set the proper disposition on originated calls. (closes issue #14167) Reported by: jpt Patches: call-file-missing-cdr2.diff uploaded by mnicholson (license 96) Tested by: dlotina, rmartinez, mnicholson ........ r258670 | mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11 lines Fix broken CDR behavior. This change allows a CDR record previously marked with disposition ANSWERED to be set as BUSY or NO ANSWER. Additionally this change partially reverts r235635 and does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call(). To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in ast_bridge_call(). (closes issue #16797) Reported by: VarnishedOtter Tested by: mnicholson ........ (closes issue #16222) Reported by: telles Tested by: mnicholson ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500 (Thu, 22 Apr 2010) | 2 lines Fix previous commit. ................ 2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant * /, main/event.c: Merged revisions 258632 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only merge the bug fixes, not the unit test. ........ r258632 | russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines Add ast_event subscription unit test and fix some ast_event API bugs. This patch introduces another test in test_event.c that exercises most of the subscription related ast_event API calls. I made some minor additions to the existing event allocation test to increase API coverage by the test code. Finally, I made a list in a comment of API calls not yet touched by the test module as a to-do list for future test development. During the development of this test code, I discovered a number of bugs in the event API. 1) subscriptions to AST_EVENT_ALL were not handled appropriately in a couple of different places. The API allows a subscription to all event types, but with IE parameters, just as if it was a subscription to a specific event type. However, the parameters were being ignored. This affected ast_event_check_subscriber() and event distribution to subscribers. 2) Some of the logic in ast_event_check_subscriber() for checking subscriptions against query parameters was wrong. Review: https://reviewboard.asterisk.org/r/617/ ........ * /, doc/tex/channelvariables.tex: Merged revisions 258515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22 Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........ 2010-04-21 22:11 +0000 [r258436] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500 (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) | 8 lines Fix looping forever when no input received in certain voicemail menu scenarios. Specifically, prompting for an extension (when leaving or forwarding a message) or when prompting for a digit (when saving a message or changing folders). ABE-2122 SWP-1268 ........ ................ 2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen * doc/tex/asterisk.tex: Remove missed line in previous merge. (issue #17220) * configure: Forgot to merge the updated configure script. (issue #17220) * doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in, doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile, configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex, build_tools/prep_tarball: Merged revisions 258351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010) | 20 lines Add ability to generate ASCII documentation from the TeX files. These changes add the ability to run 'make asterisk.txt' just like the existing 'make asterisk.pdf' commands to generate a text document from the TeX files we have in the doc/tex/ directory. I've also updated a few of the .tex files because they weren't properly escaping certain characters so they would show up as Unicode characters (like [U+021C]). Made changes to the configure scripts so it would detect the catdvi program which is required to convert the .dvi file generated by latex. I've also added a few lines to the build_tools/prep_tarball script so that the text documentation gets generated and added to future tarballs of Asterisk releases. (closes issue #17220) Reported by: lmadsen Patches: asterisk.txt.patch uploaded by lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger (license 224) Tested by: lmadsen, pabelanger ........ 2010-04-21 18:19 +0000 [r258314] David Vossel * /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 | dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines fixes issue with double "sip:" in header field This is a clear mistake in logic. Future discussions about how to avoid having to handle uri's like this should take place in the future, but this fix needs to go in for now. (closes issue #15847) Reported by: ebroad Patches: doublesip.patch uploaded by ebroad (license 878) ........ 2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen * /, configs/cli_aliases.conf.sample: Merged revisions 258149 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20 Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on asterisk-users. ........ * configs/extensions.conf.sample, /: Merged revisions 258147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20 Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers (http://www.freenum.org). Minor tweaks and documentation added by me. (closes issue #17058) Reported by: pprindeville Patches: freenum.patch#5 uploaded by pprindeville (license 347) Tested by: lmadsen ........ 2010-04-20 18:04 +0000 [r258108] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500 (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) | 11 lines Play correct prompt when voicemail store failure occurs after attempted forward. If a user's mailbox was full and a message was attempted to be forwarded to said box, warnings on the console would indicate failure. However, the played prompt was that of success (vm-msgsaved). Now storage failure is taken into account and the correct prompt (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262 ........ ................ 2010-04-20 18:02 +0000 [r258107] Leif Madsen * contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20 Apr 2010) | 7 lines Add missing 'useragent' field to sip-friends.sql file. (closes issue #17171) Reported by: thehar Patches: sip-friends.patch uploaded by thehar (license 831) Tested by: pabelanger, thehar ........ 2010-04-19 21:58 +0000 [r257948-257950] Jason Parker * main/indications.c, /: Merged revisions 257949 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 | qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line Change log message to match severity. ........ * main/indications.c, /: Merged revisions 257947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 | qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines Don't consider a missing indications.conf to be a critical error. There were many changes in revision 176627 which would avoid the error that a missing config would have caused. Other than this, there are no other config files (including asterisk.conf, surprisingly) that are required. ........ 2010-04-19 18:30 +0000 [r257850] Terry Wilson * /, main/features.c: Merged revisions 257810 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 | twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines Fix incomplete CDR merge from r195881 Because res/res_features.c was removed and main/cdr.c added, these changes didn't make it to trunk and the 1.6.x branches ........ 2010-04-18 17:28 +0000 [r257771] Tilghman Lesher * configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18 Apr 2010) | 2 lines Removing unused configuration parameters ........ 2010-04-16 21:47 +0000 [r257740] Dwayne M. Hubbard * apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500 (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines Make the mixmonitor thread process audio frames faster Mantis issue 17078 reports MixMonitor recordings have shorter durations than the call duration. This was because the mixmonitor thread was not processing frames from the audiohook fast enough. The mixmonitor thread would slowly fall behind the most recent audio frame and when the channel hangs up, the mixmonitor thread would exit without processing the same number of frames as the channel; leaving the mixmonitor recording shorter than actual call duration. This revision fixes this issue by moving the ast_audiohook_trigger_wait() and the subsequent audiohook.status check into the block where the ast_audiohook_read_frame() function returns NULL. (closes issue #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review: https://reviewboard.asterisk.org/r/611/ ........ ................ 2010-04-15 21:34 +0000 [r257510-257597] Tilghman Lesher * include/asterisk/app.h, /, main/app.c: Merged revisions 257560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500 (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) | 6 lines Allow application options with arguments to contain parentheses, through a variety of escaping techniques. Fixes SWP-1194 (ABE-2143). Review: https://reviewboard.asterisk.org/r/604/ ........ ................ * /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010) | 20 lines Merged revisions 257467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines Don't recreate peer, when responding to a repeated deregistration attempt. When a reply to a deregistration is lost in transmit, the client retries the deregistration. Previously, this would cause a realtime/autocreate peer to be loaded back into memory, after it had already been correctly purged. Instead, we just want to resend the reply without loading the peer. (closes issue #16908) Reported by: kkm Patches: 20100412__issue16908.diff.txt uploaded by tilghman (license 14) Tested by: kkm ........ ................ 2010-04-15 19:42 +0000 [r257344-257428] Leif Madsen * doc/backtrace.txt: Merged revisions 257427 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010) | 21 lines Merged revisions 257426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) | 13 lines Update backtrace.txt documentation. Update the backtrace.txt documentation so it conforms to the same layout as other documents we've been working on recently. Additionally, add a bunch of new information about gathering backtraces for crashes and deadlocks, along with ways of verifying your file before uploading it. Create a couple of one line commands for people to generate the files we need. (closes issue #17190) Reported by: lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen (license 10) Tested by: lmadsen, pabelanger ........ ................ * doc/backtrace.txt: Merged revisions 257343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010) | 9 lines Merged revisions 257342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) | 1 line Update address of the bug tracker. ........ ................ 2010-04-14 23:00 +0000 [r257265] Tilghman Lesher * configs/features.conf.sample, /, main/features.c: Merged revisions 257262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 | tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15 lines Yet another issue where the conversion of the application delimiter to comma caused an issue. Application arguments within the feature map could possibly contain a comma, which conflicts with the syntax of the features.conf configuration file. This patch allows the argument to be wrapped in parentheses or quoted, to allow the application arguments to be interpreted as a single configuration parameter. (closes issue #16646) Reported by: pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/547/ ........ 2010-04-13 19:20 +0000 [r257210] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 | tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10 lines Also unref the pvt when we delete the provisional keepalive job. (closes issue #16774) Reported by: kowalma Patches: 20100315__issue16774.diff.txt uploaded by tilghman (license 14) Tested by: falves11, jamicque Review: https://reviewboard.asterisk.org/r/591/ ........ 2010-04-13 18:43 +0000 [r257184] Matthew Nicholson * main/manager.c, /, configs/manager.conf.sample: Merged revisions 257146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr 2010) | 16 lines Merged revisions 257070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines Add an option to restore past broken behavor of the Events manager action Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested. (closes issue #17023) Reported by: nblasgen Review: https://reviewboard.asterisk.org/r/602/ ........ ................ 2010-04-13 16:38 +0000 [r257068] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010) | 8 lines Ensure that we can have commas within cdr values. (closes issue #17001) Reported by: snuffy Patches: 20100412__issue17001.diff.txt uploaded by tilghman (license 14) Tested by: snuffy ........ 2010-04-12 17:30 +0000 [r256822-256902] Leif Madsen * doc/HOWTO_collect_debug_information.txt (added): Merged revisions 256901 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010) | 23 lines Merged revisions 256900 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) | 15 lines Add How-To document on collecting debugging info for issues.asterisk.org Paul Belanger has been helping a lot with bug tracking recently and created this document that we can now point to when additional debugging information is required. This document will help those filing issues to know how to get the information required when filing their issues. This will make things easier on the developers. Initial text and changes by pabelanger. Tweaks and editing by myself. (closes issue #17159) Reported by: pabelanger Patches: HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10) Tested by: tzafrir, pabelanger, lmadsen ........ ................ * apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010) | 3 lines Remove silly debug message that is not useful. (issue #17159) ........ * /, main/logger.c: Merged revisions 256821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 | lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines CLI command logger set level auto complete. A simple patch to enable auto tab complete. (closes issue #17152) Reported by: pabelanger Patches: 0017152.patch uploaded by pabelanger (license 224) ........ 2010-04-08 22:03 +0000 [r256483] Tilghman Lesher * main/app.c: Backport /proc/%d/fd method of closing file descriptors to 1.6.2. 2010-04-06 19:40 +0000 [r256373] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/lock.h: Merged revisions 256370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010) | 2 lines Mac OS X does not support comparing a mutex to its initializer. Create a test for this. ........ 2010-04-06 18:53 +0000 [r256268-256368] Richard Mudgett * channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty after the first call. The bug is exposed if MFC/R2 support is built into asterisk (i.e., openr2.h is present in the include path). Code that unconditionally clears the CallerID name and number is included. Also fixed a malformed if test in mkintf() added by issue 15883. Converted the if statement to a switch statement for clarity. Regression of the issue 15883 fix. (closes issue #16968) Reported by: grecco Patches: issue16968.patch uploaded by rmudgett (license 664) (closes issue #16747) Reported by: viniciusfontes * channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500 (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock. SWP-1231 ABE-2163 ........ ................ 2010-05-03 Leif Madsen * Asterisk 1.6.2.7 Released 2010-04-29 Leif Madsen * Asterisk 1.6.2.7-rc3 Released 2010-04-29 10:31 +0000 [r260053] David Vossel * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in audiohook_write_list. (closes issue 0017052) Reported by: dvossel Tested by: dvossel. (closes issue 0016196) Reported by: atis. Review: https://reviewboard.asterisk.org/r/623/ 2010-04-28 10:31 +0000 [r259899] David Vossel * channels/chan_local.c, main/channel.c: Resolves deadlocks in chan_local. (closes issue 0017185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ 2010-04-13 Leif Madsen * Asterisk 1.6.2.7-rc2 Released 2010-04-13 [r257210] Tilghman Lesher Also unref the pvt when we delete the provisional keepalive job. (closes issue #16774) Reported by: kowalma Patches: 20100315__issue16774.diff.txt uploaded by tilghman (license 14) Tested by: falves11, jamicque Review: https://reviewboard.asterisk.org/r/591/ 2010-04-05 Leif Madsen * Asterisk 1.6.2.7-rc1 Released 2010-04-05 15:15 +0000 [r256162] Leif Madsen * doc/tex/localchannel.tex, /: Merged revisions 256161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010) | 1 line Fix for localchannel.tex to allow PDFs to be generated again. ........ 2010-04-02 23:56 +0000 [r256013-256020] Russell Bryant * /, apps/app_meetme.c: Merged revisions 256019 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256019 | russell | 2010-04-02 18:55:57 -0500 (Fri, 02 Apr 2010) | 10 lines Export MEETMEBOOKID and fix pin-less conferences with realtime conferences (closes issue #16866) Reported by: DEA Patches: rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA Review: https://reviewboard.asterisk.org/r/582/ ........ * channels/chan_local.c, /: Merged revisions 256015 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256015 | russell | 2010-04-02 18:46:45 -0500 (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) | 9 lines Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel() (closes issue #16840) Reported by: bzing2 Patches: patch.txt uploaded by bzing2 (license 902) issue_16840.rev1.diff uploaded by russell (license 2) Tested by: bzing2, russell ........ ................ * main/channel.c, /: Merged revisions 256010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010) | 9 lines Merged revisions 256009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010) | 2 lines Remove extremely verbose debug message. ........ ................ 2010-04-02 20:20 +0000 [r255955] Tilghman Lesher * main/asterisk.c, /: Merged revisions 255952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 | tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines Pass the PID of the Asterisk process, not the PID of the canary. (closes issue #17065) Reported by: globalnetinc Patches: astcanary.patch uploaded by makoto (license 38) Tested by: frawd, globalnetinc ........ 2010-04-01 18:21 +0000 [r255676-255816] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010) | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt uploaded by tilghman (license 14) ........ * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500 (Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) | 15 lines Ensure line terminators in email are consistent. Fixes an issue with certain Mail Transport Agents, where attachments are not interpreted correctly. (closes issue #16557) Reported by: jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14) Tested by: ebroad, zktech Reviewboard: https://reviewboard.asterisk.org/r/544/ ........ ................ 2010-03-31 17:49 +0000 [r255505] Leif Madsen * configs/sip.conf.sample, apps/app_dial.c: Merged revisions 255504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31 Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T' can be used. (closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad ........ 2010-03-30 20:58 +0000 [r255326-255413] Russell Bryant * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r255410 | russell | 2010-03-30 15:56:26 -0500 (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does not start. ........ ................ * /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010) | 9 lines Merged revisions 255322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010) | 2 lines Don't make Asterisk not start if pbx_dundi fails to initialize. ........ ................ 2010-03-26 19:28 +0000 [r255023-255067] Leif Madsen * configs/sip.conf.sample, /: Merged revisions 255066 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010) | 6 lines Replace some documentation from 1.6.x back into trunk. This documentation associated wth tlsbindaddr is still useful so lets synchronize it between trunk and 1.6.x branches. (issue #17054) ........ * configs/sip.conf.sample, /: Merged revisions 255021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010) | 8 lines Update confusing documentation for tlsbindaddr. Update some confusing documentation for the tlsbindaddr option in sip.conf.sample. Point at a link instead which has better documentation. (closes issue #17054) Reported by: klaus3000 ........ 2010-03-25 20:43 +0000 [r254770-254805] Jason Parker * utils/Makefile, /: Merged revisions 254802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) | 9 lines Merged revisions 254800 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) | 1 line Don't remove local copies of utils in uninstall. ........ ................ * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS issue with out-of-tree modules. Take 2, without ABI breakage this time. Review: https://reviewboard.asterisk.org/r/588/ 2010-03-25 20:09 +0000 [r254721] Russell Bryant * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010) | 2 lines chan_usbradio depends on alsa. ........ 2010-03-25 17:47 +0000 [r254556] Mark Michelson * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500 (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar 2010) | 5 lines Add doxygen for acl.h Review: https://reviewboard.asterisk.org/r/528 ........ ................ 2010-03-25 17:21 +0000 [r254548] Sean Bright * channels/chan_sip.c: Initialize stream to avoid a compilation error. 2010-03-25 17:12 +0000 [r254542] Mark Michelson * channels/chan_sip.c: Fix potential crashes from trying to reference nonexistent RTP streams. 2010-03-25 16:26 +0000 [r254499] Terry Wilson * /, main/file.c: Merged revisions 254453 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010) | 9 lines Merged revisions 254451 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) | 2 lines Handle new SRCCHANGE control message here too ........ ................ 2010-03-25 16:22 +0000 [r254482] Mark Michelson * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500 (Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection. Here is a copy and paste of the details from my request on reviewboard that dealt with these changes: Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too: seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF. Fix 2. The second change in place is to fix an issue like the following: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end) *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list. Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem ........ ................ 2010-03-25 15:21 +0000 [r254447] Leif Madsen * /, res/res_agi.c: Merged revisions 254446 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 | lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines handle_speechset has 4 arguments. Update code to reflect that handle_speechset has 4 arguments. (closes issue #17093) Reported by: gpatri Patches: res_agi.patch uploaded by gpatri (license 1014) Tested by: pabelanger, mmichelson ........ 2010-03-24 17:19 +0000 [r254288] Jeff Peeler * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010) | 78 lines Merged revisions 254235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) | 72 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248860. As such the dialplan test has been extended: ; non absolute path, not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test) exten => 5040, n, dial(sip/5001) ; absolute path, not combined exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten => 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1, monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ; combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m) exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n, dial(sip/5001) ; combined: changemonitor from no path to non absolute path exten => 5044, 1, monitor(wav,monitor_test6,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before exten => 5044, n, dial(sip/5001) ; non absolute path, combined exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n, dial(sip/5001) ; absolute path, combined exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n, dial(sip/5001) ; no path, combined exten => 5047, 1, monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ; combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n, dial(sip/5001) ; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n, changemonitor(tmp/jeff/monitor_test14) exten => 5049, n, dial(sip/5001) ; combined: changemonitor from no path to absolute exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n, dial(sip/5001) ; combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n, changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ; not combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19) exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n, dial(sip/5001) ; not combined: changemonitor from no path to non absolute exten => 5053, 1, monitor(wav,monitor_test21) exten => 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n, dial(sip/5001) ; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n, dial(sip/5001) ; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n, changemonitor(tmp/jeff/monitor_test25) exten => 5055, n, dial(sip/5001) ; not combined: changemonitor from no path to absolute exten => 5056, 1, monitor(wav,monitor_test26) exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056, n, dial(sip/5001) ; not combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n, changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001) ........ ................ 2010-03-23 22:05 +0000 [r254131] Tzafrir Cohen * tests/Makefile, /: Merged revisions 254001 via svnmerge from http://svn.digium.com/svn/asterisk/trunk ........ r254001 | tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines Change the name of the category 'TEST' to match the name of the subdir ........ 2010-03-23 21:20 +0000 [r254068] Jeff Peeler * main/channel.c, /: Merged revisions 254050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 | jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines Exit native bridging early for greater timing accuracy with warnings This changes native bridging to break one millisecond early so that the more accurate timeval calculations done in the generic bridge can be performed using the bridge config. Currently the time between exiting native bridging slightly late can sometimes cause a large enough discrepancy for warnings to be missed. For the record, 1.4 does not attempt to native bridge at all when warnings are enabled. (closes issue #15815) Reported by: adomjan Review: https://reviewboard.asterisk.org/r/577/ ........ 2010-03-22 19:55 +0000 [r253801] Matthew Nicholson * /, main/features.c: Merged revisions 253800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar 2010) | 11 lines Merged revisions 253799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar 2010) | 4 lines Unconditionally copy the caller's account code to the called party. (related to issue #16331) ........ ................ 2010-03-22 19:06 +0000 [r253714-253760] Tilghman Lesher * /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22 Mar 2010) | 2 lines Update query should be an UPDATE, not a SELECT. ........ * /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22 Mar 2010) | 4 lines Return the list for later manipulation. This fixes an issue with the update procedure. Debugging with mmichelson. ........ * configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged revisions 253712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 | tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines Accomodate equal signs in DSNs and add documentation, based upon mmichelson's feedback. ........ 2010-03-20 17:33 +0000 [r253595-253620] Russell Bryant * cdr/cdr_pgsql.c, main/stdtime/localtime.c, main/tcptls.c, /, main/features.c: Merged revisions 253540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253540 | russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve more compiler warnings on FreeBSD. ........ * apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD. ........ * /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 | russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve a compiler warning on FreeBSD. ........ * channels/chan_dahdi.c, /: Merged revisions 253536 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010) | 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix build issues I had with this module on FreeBSD. ........ 2010-03-19 08:05 +0000 [r253492] Alec L Davis * main/astobj2.c, /: Merged revisions 253490 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 | alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19 lines prevent segfault if bad magic number is encountered. internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic number', but internal_ao2_ref continues on, causing segfault. Although AO2_MAGIC number is checked by INTERNAL_OBJ before internal_ao2_ref is called, A02_MAGIC is being destroyed (or a wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ. internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad magic number. (issue #17037) Reported by: alecdavis Patches: bug17037.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ 2010-03-18 17:54 +0000 [r253257-253346] Leif Madsen * /, apps/app_userevent.c: Merged revisions 253345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253345 | lmadsen | 2010-03-18 12:52:35 -0500 (Thu, 18 Mar 2010) | 7 lines Change usage of pipe to comma in UserEvent docs. Change the example usage of pipe as a separator to comma in the UserEvent documentation. (closes issue #16961) Reported by: jlpedrosa ........ * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010) | 9 lines Update to new Local channel documentation. Add same changes as commit to 1.4, but convert to TeX. (issue #16963) Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz (license 834) ........ 2010-03-17 16:25 +0000 [r253158] Terry Wilson * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_mgcp.c, channels/chan_sip.c, include/asterisk/rtp.h: Revert API change in release branches This re-renames ast_rtp_update_source to ast_rtp_new_source 2010-03-17 00:41 +0000 [r253029-253033] Leif Madsen * main/xmldoc.c, /: Merged revisions 253032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253032 | lmadsen | 2010-03-16 19:40:51 -0500 (Tue, 16 Mar 2010) | 1 line Fix a typo. ........ * configs/say.conf.sample, /: Merged revisions 253028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500 (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010) | 6 lines Add french snipset to say.conf. Add the french snipset to say.conf. (Closes issue #15799) ........ ................ 2010-03-16 23:54 +0000 [r252978] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 | tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines Mask out previous arguments on each nested invocation of Gosub. (closes issue #16758) Reported by: wdoekes Patches: 20100316__issue16758.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/561/ ........ 2010-03-16 19:38 +0000 [r252850] Sean Bright * res/res_clialiases.c, /: Merged revisions 252848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252848 | seanbright | 2010-03-16 15:36:24 -0400 (Tue, 16 Mar 2010) | 10 lines Include an extra newline after "Aliased CLI command" to get back the prompt. The other issue mentioned in this bug will be more difficult to resolve since we have no idea (right now) of knowing if the command that is aliased has been installed yet. (issue #16978) Reported by: jw-asterisk Tested by: seanbright ........ 2010-03-16 19:02 +0000 [r252770] Russell Bryant * utils/Makefile, /: Merged revisions 252767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010) | 13 lines Merged revisions 252766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010) | 6 lines Don't treat warnings as errors for muted. muted supports OS X, but uses functions marked as deprecated in 10.6. However, the functions are still supported, so just ignore the warnings for now and allow the build to proceed. ........ ................ 2010-03-16 18:49 +0000 [r252763] Leif Madsen * configs/extensions.ael.sample, /: Merged revisions 252762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500 (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines Additional extensions.ael global variable fixes. Fixing up a couple more overlapping global variable namespaces shared with extensions.conf.sample. Also noticed a few of the lines that were commented out didn't have the closing semi-colon so I added that as well. (issue #17035) ........ ................ 2010-03-15 21:59 +0000 [r252626] Sean Bright * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 | seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4 lines Resolve a crash in SLATrunk when the specified trunk doesn't exist. Reported by philipp64 in #asterisk-dev. ........ 2010-03-15 21:54 +0000 [r252622] Tilghman Lesher * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions 252619 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010) | 9 lines Merged revisions 252617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010) | 2 lines Uh, yeah. Umask. I'm stupid. ........ ................ 2010-03-15 20:53 +0000 [r252535] Leif Madsen * configs/extensions.ael.sample: Merged revisions 252534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500 (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines Update extensions.ael file to not overlap extensions.conf. Updated the extensions.ael file so the global variables don't overlap those that we have in extensions.conf (sample files). This way unexpected things won't happed hopefully if both pbx_ael and res_config are loaded. (closes issue #17035) Reported by: pprindeville ........ ................ 2010-03-15 05:04 +0000 [r252365-252444] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 | tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines THIS IS NOT PYTHON. Indentation doesn't matter, only braces do. (closes issue #17025) Reported by: smurfix Patches: sip.patch uploaded by smurfix (license 547) ........ * main/asterisk.c, Makefile, contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged revisions 252362 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010) | 11 lines Merged revisions 252361 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010) | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard: https://reviewboard.asterisk.org/r/551/ ........ ................ 2010-03-14 17:48 +0000 [r252317] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar 2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They added a sqlite3_log() function which was conflicting with our function names. (closes issue #17017) Reported by: alephlg ........ 2010-03-13 00:32 +0000 [r252137-252178] Terry Wilson * main/rtp.c: Remove unusued field * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c, channels/chan_mgcp.c, main/channel.c, /, channels/chan_sip.c, channels/chan_skinny.c, include/asterisk/rtp.h, channels/chan_h323.c: Merged revisions 252089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ 2010-03-12 22:05 +0000 [r252090] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 252088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252088 | moy | 2010-03-12 16:57:40 -0500 (Fri, 12 Mar 2010) | 1 line add missing mfcr2_skip_category setting ........ 2010-03-12 19:50 +0000 [r251994] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010) | 8 lines Don't override a user option with the global option. (closes issue #16849) Reported by: ip-rob Patches: 20100311__issue16849.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob ........ 2010-03-12 19:49 +0000 [r251991] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 251946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251946 | rmudgett | 2010-03-12 13:05:40 -0600 (Fri, 12 Mar 2010) | 1 line Doxegen this chan_dahdi lock. ........ 2010-03-11 21:08 +0000 [r251879-251887] Tilghman Lesher * apps/app_exec.c, /: Merged revisions 251884 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 | tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines Because ExecIf needs to reprocess arguments, it's best if we don't remove quotes during parsing. (closes issue #16905) Reported by: ip-rob Patches: 20100303__issue16905.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob ........ * apps/app_system.c, /: Merged revisions 251877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 | tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines If the argument to the system application is quoted, ensure we remove the quotes before trying to execute. (closes issue #16842) Reported by: ip-rob Patches: 20100310__issue16842.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob ........ 2010-03-11 Leif Madsen * Asterisk 1.6.2.6 released 2010-03-05 Leif Madsen * Asterisk 1.6.2.6-rc2 released 2010-03-05 Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman | 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in ODBC query. (closes issue #16953) Reported by: elguero Patches: app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37) ........ 2010-03-04 Leif Madsen * Asterisk 1.6.2.6-rc1 released 2010-03-03 21:24 +0000 [r250610] Leif Madsen * doc/tex/localchannel.tex, /: Merged revisions 250609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010) | 11 lines Update existing Local channel documentation. A complete re-write of the Local channel documentation has been performed, with the existing information from localchannel.txt and localchannel.tex merged in. (closes issue #16637) Reported by: kobaz Patches: localchannel.tex uploaded by lmadsen (license 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson ........ 2010-03-03 19:13 +0000 [r250484] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines Make sure to clear red alarm after polarity reversal. From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ ................ 2010-03-03 18:05 +0000 [r250253-250396] David Vossel * channels/chan_iax2.c, /: Merged revisions 250395 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets When Asterisk receives an IAX2 TXREQ packet, try_transfer() will call store_by_transfercallno() to link the chan_iax2_pvt struct into iax_transfercallno_pvts. If a duplicate TXREQ packet is received for the same call, the pvt struct will be linked into iax_transfercallno_pvts multiple times. This patch fixes this. Thanks rain for debugging this and providing a patch! (closes issue #16904) Reported by: rain Patches: iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested by: rain, dvossel ........ ................ * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 | dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines fixes signed to unsigned int comparision issue for FaxMaxDatagram value. ........ 2010-03-02 21:10 +0000 [r249953-250052] Leif Madsen * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010) | 8 lines Update IMAP documentation. Update the IMAP documentation to make it clear that storing voicemails in the same folder as a large number of emails could potentially cause significant slow downs when writing or retrieving voicemails. (issue #16704) Reported by: TimeHider Tested by: lmadsen, TimeHider ........ * configs/cdr.conf.sample: Merged revisions 250045 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500 (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines Update documentation to clarify purpose of unanswered option. (closes issue #16267) Reported by: elsto Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested by: davidw, elsto ........ ................ * doc/tex/configuration.tex, /: Merged revisions 250037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 Mar 2010) | 4 lines Update documentation to not imply we support overriding options. (closes issue #16855) Reported by: davidw ........ * apps/app_directory.c, /: Merged revisions 249950 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010) | 4 lines Fix literal values wrapped in documentation. (closes issue #16145) Reported by: tilghman ........ 2010-03-02 19:50 +0000 [r249952] Alec L Davis * UPGRADE-1.6.txt, main/editline/makelist.in, apps/app_echo.c, UPGRADE.txt: revert ability to exit echo app caused a regression, as only supported VOICE, not VIDEO etc. (issue #16880) 2010-03-02 19:26 +0000 [r249916-249933] Leif Madsen * /, main/features.c: Merged revisions 249925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 | lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines Add missing description of the PARKINGLOT variable in XML documentation. (closes issue #16743) Reported by: snuffy Patches: parkingdoc.diff uploaded by snuffy (license 35) ........ * /, pbx/pbx_dundi.c: Merged revisions 249912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 | lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines Convert some DUNDI functions to XML documentation. (closes issue #16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by snuffy (license 35) ........ 2010-03-02 19:12 +0000 [r249895] David Vossel * channels/chan_console.c, channels/chan_gtalk.c, channels/chan_oss.c, channels/misdn_config.c, include/asterisk/abstract_jb.h, configs/alsa.conf.sample, channels/chan_jingle.c, channels/chan_usbradio.c, channels/chan_dahdi.c, channels/chan_skinny.c, configs/mgcp.conf.sample, main/abstract_jb.c, channels/chan_h323.c, channels/chan_alsa.c, configs/sip.conf.sample, channels/chan_mgcp.c, channels/chan_unistim.c, configs/console.conf.sample, configs/chan_dahdi.conf.sample, channels/chan_local.c, configs/oss.conf.sample, channels/chan_sip.c, /, configs/usbradio.conf.sample, configs/misdn.conf.sample: Merged revisions 249893 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ 2010-03-02 19:09 +0000 [r249894] Leif Madsen * /, apps/app_confbridge.c: Merged revisions 249892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010) | 1 line Fix several XML documentation validate errors. ........ 2010-03-02 09:05 +0000 [r249844] Alec L Davis * apps/app_echo.c: fixes ability to exit echo app when called from a ISDN channel, null frames prevent '#' exit. Now only echo back VOICE and DTMF frames (issue #16880) Reported by: alecdavis Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis 2010-03-01 19:40 +0000 [r249675] Sean Bright * apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar 2010) | 11 lines Fix crash in app_voicemail related to message counting. We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *' causing a segfault. (closes issue #16921) Reported by: whardier Patches: 20100301_issue16921.patch uploaded by seanbright (license 71) Tested by: whardier ........ ................ 2010-03-01 18:47 +0000 [r249625] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010) | 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP compile once again. ........ 2010-03-01 17:25 +0000 [r249580] Jeff Peeler * channels/chan_local.c, /: Merged revisions 249538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) | 11 lines Modify queued frames from local channels to not set the other side to up In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor ........ ................ 2010-02-28 20:51 +0000 [r249407-249493] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010) | 5 lines Fix unit test that Alec Davis broke. (closes issue #16927) Reported by: alecdavis ........ * apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions 249405 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 | tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines Properly document voicemail API documents. Also fix a crash reported via the -dev list. ........ 2010-02-27 23:04 +0000 [r249321] Alec L Davis * channels/chan_dahdi.c: overlap receiving: automatically send CALL PROCEEDING when dialplan starts Following Q.931 5.2.4 When the user has determined that sufficient call information has been received the user shall stop T302 and send CALL PROCEEDING to the network. Previously timeouts were possible if the dialplan took a long time to issue any response back to the network. Verified that our local TELCO also does the same. (issue #16789) Reported by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis 2010-02-27 14:10 +0000 [r249238] Kevin P. Fleming * channels/chan_iax2.c, /: Merged revisions 249235 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500 (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 Feb 2010) | 1 line add a reference to the now-published IAX2 RFC ........ ................ 2010-02-26 18:49 +0000 [r249190] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010) | 18 lines Cleanups to fix bugs in the VM count API functions. - Urgent voicemails were not attached, because the attachment code looked in the wrong folder. - Urgent voicemails were sometimes counted twice when displaying the count of new messages. - Backends were inconsistent as to which voicemails each API counted. (closes issue #15654) Reported by: tomo1657 Patches: 20100225__issue15654.diff.txt uploaded by tilghman (license 14) Tested by: tilghman (closes issue #16448) Reported by: hevad Review: https://reviewboard.asterisk.org/r/525/ ........ 2010-02-26 17:06 +0000 [r249104] Mark Michelson * /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb 2010) | 14 lines Merged revisions 249100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. (closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) ........ ................ 2010-02-25 23:12 +0000 [r248955] Jeff Peeler * res/res_monitor.c, /: Merged revisions 248952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010) | 24 lines Merged revisions 248860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248757. As such the dialplan test has been extended: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning exten => 5044, n, dial(sip/5001) ........ ................ 2010-02-25 22:42 +0000 [r248949] Mark Michelson * /, main/acl.c: Merged revisions 248946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 | mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5 lines Fix incorrect ACL behavior when CIDR notation of "/0" is used. AST-2010-003 ........ 2010-02-25 21:25 +0000 [r248864] Tilghman Lesher * main/asterisk.c, /: Merged revisions 248861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) | 15 lines Some platforms clear /var/run at boot, which makes connecting a remote console... difficult. Previously, we only created the default /var/run/asterisk directory at install time. While we could create it in the init script, that would not work for those who start asterisk manually from the command line. So the safest thing to do is to create it as part of the Asterisk boot process. This also changes the ownership of the directory, because the pid and ctl files are created after we setuid/setgid. (closes issue #16802) Reported by: Brian Patches: 20100224__issue16802.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir ........ ................ 2010-02-25 18:52 +0000 [r248797] Jeff Peeler * res/res_monitor.c, /: Merged revisions 248793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines Ensure that monitor recordings are written to the correct location. Recordings should be placed in the monitor directory when a non-absolute path is used. Exact dialplan used for testing: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) ABE-2101 ........ ................ 2010-02-24 21:29 +0000 [r248643] Tilghman Lesher * /, main/logger.c: Merged revisions 248584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010) | 14 lines Merged revisions 248582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) | 7 lines Remove color code sequences from verbose messages that go to logfiles. (closes issue #16786) Reported by: dodo Patches: logger2.patch uploaded by dodo (license 989) Tested by: tilghman ........ ................ 2010-02-23 16:37 +0000 [r248398] David Vossel * /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines Merged revisions 248396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines fixes invite with replaces deadlock (closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel ........ ................ 2010-02-19 19:07 +0000 [r248011] Tilghman Lesher * channels/chan_console.c, main/loader.c, /: Merged revisions 228798 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228798 | tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix various problems detected with Valgrind. * chan_console accessed pvts after deallocation. * The module loader did not check usecount on shutdown, which led to chan_iax2 reading a timer that was already unloaded. (closes issue #16062) Reported by: alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ 2010-02-19 19:00 +0000 [r248005] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 248003 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1 line mfcr2 issue 0016844 - Fix portability bit fields and make mfcr2_immediate_accept work again, reported and patched by korihor ........ 2010-02-19 18:45 +0000 [r248004] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600 (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ ................ 2010-02-19 17:41 +0000 [r247916] David Vossel * /, channels/chan_sip.c: Merged revisions 247915 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 | dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines handle_request_invite revise comment, fix coding guideline issues I'm working with this code right now trying to analyze a deadlock. This change is just to clean up a few things before I make a more complex patch. ........ 2010-02-18 23:15 +0000 [r247792-247845] Tilghman Lesher * res/res_speech.c, /: Merged revisions 247841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 | tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines Revert an errant part of a previous cleanup, to fix a memory corruption issue. (closes issue #16368) Reported by: thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf (license 955) ........ * /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 | tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 lines If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. NULL means the value is not specified for the column, which normally means the driver uses whatever is the default value. However, on MySQL, placing a NULL in either a float or integer column results in a retrieval of the 0 value. Hence, users get an errant error on load. This patch suppresses that error and makes the value as if it was not there. Note that this cannot be done in the realtime driver, because the lack of difference between NULL and 0 can only be intepreted correctly by the driver itself. If we did it in the realtime driver, then it would be effectively impossible to set any realtime field to 0, because it would act as if the field were unspecified and possibly take on a different value. (closes issue #16683) Reported by: wdoekes ........ 2010-02-18 21:25 +0000 [r247737-247776] David Vossel * /, bridges/bridge_softmix.c: Merged revisions 247770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010) | 9 lines fixes confbridge crash when no timing module is loaded. (closes issue #16471) Reported by: kjotte Patches: M16471.diff uploaded by junky (license 177) Tested by: kjotte, junky ........ * apps/app_queue.c, /: Merged revisions 247736 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 | dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines fixes Queue with C option crash (closes issue #16475) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) ........ 2010-02-18 19:41 +0000 [r247653] Matthew Nicholson * /, main/features.c: Merged revisions 247652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb 2010) | 13 lines Merged revisions 247651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb 2010) | 6 lines Copy the calling party's account code to the called party if they don't already have one. (closes issue #16331) Reported by: bluefox Tested by: mnicholson ........ ................ 2010-02-18 16:58 +0000 [r247506-247512] Leif Madsen * README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500 (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010) | 1 line Add additional link to best practices document per jsmith. ........ ................ * README-SERIOUSLY.bestpractices.txt (added): Merged revisions 247503 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010) | 18 lines Merged revisions 247502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) | 10 lines Add best practices documentation. (issue #16808) Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/507/ ........ ................ 2010-02-18 04:21 +0000 [r247426] Russell Bryant * sounds/Makefile, Makefile, /: Merged revisions 247423 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247423 | russell | 2010-02-17 22:20:11 -0600 (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) | 10 lines Tweak argument handling for wget in the sounds Makefile. 1) Fix the check to see if we are using wget to not be full of fail. The configure script populates this variable with the absolute path to wget if it is found, so it didn't work. 2) Allow some extra arguments to be passed in for wget. This is just a simple change to allow our Bamboo build script to tell wget to be quiet and not fill up our logs with download status output. ........ ................ 2010-02-17 21:32 +0000 [r246989-247337] Mark Michelson * include/asterisk/strings.h, main/strings.c, /: Merged revisions 247335 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 | mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20 lines Fix two problems in ast_str functions found while writing a unit test. 1. The documentation for ast_str_set and ast_str_append state that the max_len parameter may be -1 in order to limit the size of the ast_str to its current allocated size. The problem was that the max_len parameter in all cases was a size_t, which is unsigned. Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the max_len parameter to be ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an off-by-one error in the case where we attempted to write a string larger than the current allotted size to a string when -1 was passed as the max_len parameter. When trying to write more than the allotted size, the ast_str's __AST_STR_USED was set to 1 higher than it should have been. Thanks to Tilghman for quickly spotting the offending line of code. Oh, and the unit test that I referenced in the top line of this commit will be added to reviewboard shortly. Sit tight... ........ * apps/app_queue.c, /: Merged revisions 247169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb 2010) | 9 lines Merged revisions 247168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls. ........ ................ * main/strings.c, /: Merged revisions 247076 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247076 | mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12 lines Add va_end calls to __ast_str_helper. According to the man page for stdarg(3), "Each invocation of va_copy() must be matched by a corresponding invocation of va_end() in the same function." There were several cases in __ast_str_helper where va_copy was not matched with a corresponding call to va_end. ........ * include/asterisk/strings.h, /: Merged revisions 246985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue, 16 Feb 2010) | 3 lines Add some clarifying documentation to the ast_str_set and ast_str_append functions. ........ 2010-02-16 21:03 +0000 [r246900-246982] David Vossel * main/tcptls.c, /: Merged revisions 246980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 | dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines warning message if openssl support is missing while attempting tls connection (closes issue #16673) Reported by: michaesc Patches: tls_error_msg.diff uploaded by dvossel (license 671) ........ * main/channel.c: fixes merge error with Monitor calculation fix * main/channel.c, /: Merged revisions 246899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 | dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines fixes sample rate conversion issue with Monitor application When using ast_seekstream with the read/write streams of a monitor, the number of samples we are seeking must be of the same rate as the stream or the jump calculation will be incorrect. This patch adds logic to correctly convert the number of samples to jump to the sample rate the read/write stream is using. For example, if the call is G722 (16khz) and the read/write stream is recording a 8khz wav, seeking 320 samples of 16khz audio is not the same as seeking 320 samples of 8khz audio when performing the ast_seekstream on the stream. ABE-2044 ........ 2010-02-15 23:45 +0000 [r246713] Tilghman Lesher * Makefile, /: Merged revisions 246710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010) | 12 lines Merged revisions 246709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) | 5 lines Make the menuselect instructions correct by allowing 'make menuselect' to actually solve dependency problems. (Previously, it would fail out again with the same message about running 'make menuselect', which was NOT at all helpful.) ........ ................ 2010-02-12 23:33 +0000 [r246547] David Vossel * main/channel.c, /: Merged revisions 246546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010) | 21 lines Merged revisions 246545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) | 16 lines lock channel during datastore removal On channel destruction the channel's datastores are removed and destroyed. Since there are public API calls to find and remove datastores on a channel, a lock should be held whenever datastores are removed and destroyed. This resolves a crash caused by a race condition in app_chanspy.c. (closes issue #16678) Reported by: tim_ringenbach Patches: datastore_destroy_race.diff uploaded by tim ringenbach (license 540) Tested by: dvossel ........ ................ 2010-02-12 19:08 +0000 [r246464] Jason Parker * main/channel.c: Fix some silly formatting that made my head hurt. 2010-02-10 21:28 +0000 [r246199-246207] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 246204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010) | 2 lines Fussy compiler on another machine... ........ * /, funcs/func_strings.c: Merged revisions 246200 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010) | 2 lines Fix weird issue with unit tests on optimized build - turned out to be a signing issue. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, res/res_agi.c: Merged revisions 246030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246030 | tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12 lines Solaris doesn't like outputting a NULL to a %s in format strings. Detect all platforms that don't like that, either, and ensure that when documentation is missing, we pass a non-NULL pointer when outputting the corresponding documentation. (closes issue #16689) Reported by: bklang Patches: 20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/497/ ........ 2010-02-10 17:51 +0000 [r246117] David Vossel * apps/app_queue.c, /: Merged revisions 246116 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010) | 14 lines Merged revisions 246115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) | 8 lines fixes random deadlock in app_queue with use_weight during reload (closes issue #16677) Reported by: tim_ringenbach Patches: app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540) ........ ................ 2010-02-10 16:58 +0000 [r246073] Jeff Peeler * channels/chan_local.c, /: Merged revisions 246070 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010) | 22 lines Change channel state on local channels for busy,answer,ring. Previously local channels channel state never changed. This became problematic when the state of the other side of the local channel was lost, for example during a masquerade. Changing the state of the local channel allows for the scenario to be detected when the channel state is set to ringing, but the peer isn't ringing. The specific problem scenario is described in 164201. Although this was noted on one of the issues, here is the tested dialplan verified to work: exten => 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) exten => *9700,n,wait(3) ;3 works, 1 did not exten => *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did not exten => 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes issue #14992) Reported by: davidw ........ 2010-02-10 15:38 +0000 [r245948-246025] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 246022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010) | 2 lines Enable warnings on atypical conditions for the FILTER function (suggested by mmichelson on the -dev list). ........ * configs/extensions.conf.sample, /, funcs/func_strings.c: Merged revisions 245945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010) | 9 lines Merged revisions 245944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) | 2 lines Include examples of FILTER usage in extension patterns where a "." may be a risk. ........ ................ 2010-02-09 23:11 +0000 [r245794] David Vossel * channels/chan_iax2.c, /: Merged revisions 245793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600 (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 = 32768 which is the maximum allowed iax2 callnumber. Creating the iaxs and iaxsl array of size 32768 means the maximum callnumber is actually out of bounds. This causes a nasty crash. (closes issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded by dvossel (license 671) ........ ................ 2010-02-09 18:09 +0000 [r245732] Tilghman Lesher * /, apps/app_fax.c: Merged revisions 245729 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 | tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines Ensure frames are only freed once. (closes issue #16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt uploaded by tilghman (license 14) Tested by: kenny, bloodoff, misaksen ........ 2010-02-09 17:43 +0000 [r245728] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 245727 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245727 | mnicholson | 2010-02-09 11:40:04 -0600 (Tue, 09 Feb 2010) | 2 lines This commit removes an extra newline in T.38 generated SDP packets. This bug was caused by the fix introduced in r243860. (closes issue #16766) Reported by: raivisr Patches: t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96) Tested by: raivisr ........ 2010-02-09 16:26 +0000 [r245683] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 245680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 | kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8 lines Don't offer MMR or JBIG transcoding during T.38 negotiation. After further discussion with Steve Underwood, we should not (yet) be offering to receive MMR or JBIG transcoded streams from T.38 endpoints. A future spandsp release will support those features, and then they can be enabled during negotiation ........ 2010-02-08 23:47 +0000 [r245626] Russell Bryant * /, main/event.c: Merged revisions 245624 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245624 | russell | 2010-02-08 17:43:00 -0600 (Mon, 08 Feb 2010) | 5 lines Fix return value of get_ie_str() and get_ie_str_hash() for non-existent IE. I found this bug while developing a unit test for event allocation. Testing is awesome. ........ 2010-02-08 22:46 +0000 [r245581] Tilghman Lesher * channels/Makefile, /, main/Makefile: Merged revisions 245578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08 Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and channels/ Makefiles. They were previously passed correctly, but they simply weren't used. This caused issues with various platforms whose builds needed to pass special linker flags via the configure script. (closes issue #16596) Reported by: pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by pprindeville (license 347) Tested by: tilghman ........ 2010-02-08 20:43 +0000 [r245500] Jason Parker * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245497 | qwell | 2010-02-08 14:41:05 -0600 (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) | 4 lines Remove reference of documentation in source directory. People don't always build Asterisk from source (distro packages, anybody?). ........ ................ 2010-02-05 19:27 +0000 [r245097] Jeff Peeler * contrib/firmware (removed), /, LICENSE: Merged revisions 245090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600 (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb 2010) | 5 lines Remove contrib/firmware directory as it is empty Remove explicit license for IAXy firmware as it is no longer included in the tree ........ ................ 2010-02-05 17:10 +0000 [r244930] Sean Bright * main/asterisk.c, /: Merged revisions 244927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb 2010) | 9 lines Merged revisions 244926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb 2010) | 1 line Update main copyright date. ........ ................ 2010-02-03 19:28 +0000 [r244555] Mark Michelson * main/sched.c, /: Merged revisions 244547 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244547 | mmichelson | 2010-02-03 13:26:53 -0600 (Wed, 03 Feb 2010) | 3 lines Initialize counters in ast_sched_report so that resulting data is not bogus. ........ 2010-02-03 18:47 +0000 [r244508] Tilghman Lesher * channels/chan_dahdi.c, /, main/ast_expr2f.c: Merged revisions 244505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244505 | tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines The chanvar= setting should inherit the entire list of variables, not just the first one. (closes issue #16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded by raarts (license 937) Tested by: raarts ........ 2010-02-02 22:29 +0000 [r244445] David Vossel * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 244443 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 | dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported by: krn (closes issue #16724) Reported by: barthpbx (closes issue #16517) Reported by: bklang (closes issue #16485) Reported by: elsto ........ 2010-02-02 20:35 +0000 [r244395] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 244393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244393 | tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18 lines Properly respect GOSUB_RESULT as to what to do with the master channel. Previously, we would parse GOSUB_RESULT, but not actually do anything with it. (closes issue #16686) Reported by: bklang Patches: app_dial-respect-gosub_result.patch uploaded by bklang (license 919) (with modifications) ........ 2010-02-02 Leif Madsen * Release Asterisk 1.6.2.2 * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash occurs when the FaxMaxDatagram field is omitted from the SDP as well. 2010-01-14 Leif Madsen * Release Asterisk 1.6.2.1 2010-01-08 Leif Madsen * Release Asterisk 1.6.2.1-rc1 2010-01-07 21:17 +0000 [r238499] Tilghman Lesher * channels/chan_console.c, channels/chan_oss.c, main/poll.c, channels/chan_usbradio.c, include/asterisk/utils.h, /, channels/chan_sip.c, channels/chan_alsa.c: Merged revisions 209400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. (closes issue #16251) Reported by: asgaroth ........ 2010-01-07 20:17 +0000 [r238362-238416] David Vossel * channels/chan_iax2.c, /: Merged revisions 238412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600 (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in chan_iax A signed short was used to represent a callnumber. This is makes it possible to attempt to access the iaxs array with a negative index. (closes issue #16565) Reported by: jensvb ........ ................ * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 | dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines Change in sip show channels display format allowing more digits for CID (closes issue #16459) Reported by: Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins (license 953) ........ * apps/app_queue.c, /: Merged revisions 238361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 | dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines cli 'queue show' formatting fix. queue name was truncated over 12 characters (closes issue #16078) Reported by: RoadKill Patches: quequename_limit.patch uploaded by ppyy (license 906) Tested by: dvossel ........ 2010-01-07 09:49 +0000 [r238349] Tzafrir Cohen * configs/sip.conf.sample, /: Merged revisions 238313 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) | 2 lines Document the usefulness of explicit udp:// in the register string ........ 2010-01-06 21:48 +0000 [r238234] Tilghman Lesher * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010) | 11 lines Merged revisions 238230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010) | 4 lines Revise documentation on disposition values to the actual values used. (closes issue #16289) Reported by: wdoekes ........ ................ 2010-01-06 20:40 +0000 [r238137-238185] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 238181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 | jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines Fix misreverting from 177158. (closes issue #15725) Reported by: shanermn Patches: v1-15725.patch uploaded by dimas (license 88) Tested by: shanermn ........ * /, main/features.c: Merged revisions 238134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 | jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines Fix channel name comparison for bridge application. The channel name comparison was not comparing the whole string and therefore if one channel name was a substring of the other, the bridge would fail. (closes issue #16528) Reported by: telecos82 Patches: res_features_r236843.diff uploaded by telecos82 (license 687) ........ 2010-01-06 15:22 +0000 [r238013] Russell Bryant * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010) | 14 lines Merged revisions 238009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) | 7 lines Resolve a crash due to an ast_frame not being fully initialized. (closes issue #16531) Reported by: john8675309 (closes SWP-615) ........ ................ 2010-01-06 06:54 +0000 [r237969] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 237968 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237968 | tilghman | 2010-01-06 00:53:23 -0600 (Wed, 06 Jan 2010) | 2 lines Whoa, duplicate setting (dead code). ........ 2010-01-05 23:10 +0000 [r237924] Kinsey Moore * apps/app_test.c: Add a wait to ensure TestServer thinks it has finished sending the final digit. This was previously committed to 1.4, 1.6.0, 1.6.1, and trunk just after 1.6.2 was created (and missed). 1.6.2 also needs this patch to resolve the bug. (closes issue #16550) Reported by: opticron Patches: apptest.diff uploaded by opticron (license 267) 2010-01-05 23:09 +0000 [r237840-237921] David Vossel * apps/app_queue.c, /: Merged revisions 237920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 | dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines fixes holdtime playback issue in app_queue When reporting hold time, the number of seconds should be mod 60. Otherwise audio playback could be something like "2 minutes 123 seconds" rather than "2 minutes 3 seconds". Also, the "minute" sound file is missing, so for the moment until that file can be created the "minutes" file is used instead. (closes issue #16168) Reported by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by nickilo (license ) Tested by: nickilo, wonderg ........ * main/pbx.c, /: Merged revisions 237839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 | dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines fixes subscriptions being lost after 'module reload' During a module reload if multiple extension configs are present, such as both extensions.conf and extensions.ael, watchers for one config's hints will be lost during the merging of the other config. This happens because hint watchers are only preserved for the current config being merged. The old context list is destroyed after the merging takes place, meaning any watchers that were not perserved will be removed. Now all hints are preserved during merging regardless of what config file is being merged. These hints are only restored if they are present within the new context list. (closes issue #16093) Reported by: jlaroff ........ 2010-01-05 17:25 +0000 [r237743] Russell Bryant * /, main/utils.c: Merged revisions 237699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010) | 14 lines Merged revisions 237697 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010) | 7 lines Change a NOTICE log message to DEBUG where it belongs. (closes issue #16479) Reported by: alexrecarey (closes SWP-577) ........ ................ 2010-01-05 16:09 +0000 [r237657] Michiel van Baak * apps/app_mixmonitor.c, /: Merged revisions 237656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010) | 6 lines Make CLI command 'mixmonitor start|stop work again. (closes issue #16534) Reported by: jlaguilar Fix as suggested by jlaguilar in the bugreport ........ 2010-01-04 21:52 +0000 [r237409-237577] Tilghman Lesher * /, main/say.c: Merged revisions 237574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010) | 13 lines Merged revisions 237573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010) | 6 lines Bounds checking for input string (closes issue #16407) Reported by: qwell Patches: 20100104__issue16407.diff.txt uploaded by tilghman (license 14) ........ ................ * main/pbx.c, /: Merged revisions 237494 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010) | 15 lines Merged revisions 237493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010) | 8 lines Regression in issue #15421 - Pattern matching (closes issue #16482) Reported by: wdoekes Patches: astsvn-16482-betterfix.diff uploaded by wdoekes (license 717) 20091223__issue16482.diff.txt uploaded by tilghman (license 14) Tested by: wdoekes, tilghman ........ ................ * main/config.c, /: Merged revisions 237414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 | tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines Oops, didn't compile (thanks, kpfleming) ........ * main/config.c, /: Merged revisions 237410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 | tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines Further reduce the encoded blank values back to blank in the realtime API. (closes issue #16533) Reported by: sergee Patches: 200100104__issue16533.diff.txt uploaded by tilghman (license 14) Tested by: sergee ........ * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged revisions 237406 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) | 23 lines Merged revisions 237405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines Add a flag to disable the Background behavior, for AGI users. This is in a section of code that relates to two other issues, namely issue #14011 and issue #14940), one of which was the behavior of Background when called with a context argument that matched the current context. This fix broke FreePBX, however, in a post-Dial situation. Needless to say, this is an extremely difficult collision of several different issues. While the use of an exception flag is ugly, fixing all of the issues linked is rather difficult (although if someone would like to propose a better solution, we're happy to entertain that suggestion). (closes issue #16434) Reported by: rickead2000 Patches: 20091217__issue16434.diff.txt uploaded by tilghman (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: rickead2000 ........ ................ 2010-01-04 16:50 +0000 [r237328] David Vossel * apps/app_queue.c, /: Merged revisions 237327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237327 | dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines app_queue segfaults if realtime field uniqueid is NULL (closes issue #16385) Reported by: haakon Patches: app_queue.c.patch uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by dvossel (license 671) Tested by: haakon ........ 2010-01-04 16:27 +0000 [r237326] Jeff Peeler * /, res/res_agi.c: Merged revisions 237323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 | jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines Fix timeout for AGI command speech recognize. (closes issue #16297) Reported by: semond ........ 2010-01-04 16:21 +0000 [r237322] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 237319 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600 (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) | 3 lines It's also possible for the Local channel to directly execute an Application. Reviewboard: https://reviewboard.asterisk.org/r/452/ ........ ................ 2010-01-02 10:03 +0000 [r237139] Olle Johansson * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10 lines Merged revisions 237135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 lines Release memory of the contact acl before unloading module ........ ................ 2009-12-30 22:00 +0000 [r236985] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 236982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600 (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) | 9 lines Don't queue frames to channels that have no means to process them. (closes issue #15609) Reported by: aragon Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14) Tested by: aragon Review: https://reviewboard.asterisk.org/r/452/ ........ ................ 2009-12-30 21:13 +0000 [r236899-236905] Jeff Peeler * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 | jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines One more LOW_MEMORY compile fix. ........ * main/cli.c, /: Merged revisions 236893 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236893 | jpeeler | 2009-12-30 14:34:41 -0600 (Wed, 30 Dec 2009) | 11 lines Fix compiling with LOW_MEMORY. Modified handle_verbose to be LOW_MEMORY aware. (closes issue #16381) Reported by: michael_iedema Patches: ast_complete_source_filename.patch uploaded by michael iedema (license 942) modified by me ........ 2009-12-30 17:56 +0000 [r236804-236850] Tilghman Lesher * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009) | 4 lines When the field is blank, don't warn about the field being unable to be coerced, just skip the column. (closes http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html) Reported by Nic Colledge on the -dev list, fixed by me. ........ * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 | tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines Shut down the SIP session timers more gracefully, in order to prevent a possible crash. (closes issue #16452) Reported by: corruptor Patches: 20091221__issue16452.diff.txt uploaded by tilghman (license 14) Tested by: corruptor ........ 2009-12-28 22:13 +0000 [r236716] Jason Parker * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec 2009) | 8 lines Allow "REMAINDER" to function properly in expressions. (closes issue #16427) Reported by: wdoekes Patches: ast16-reminder-remainder.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ 2009-12-28 17:40 +0000 [r236670] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009) | 4 lines Use recommended option, not deprecated option. (closes issue #16515) Reported by: ManChicken ........ 2009-12-28 15:31 +0000 [r236513-236635] Sean Bright * include/asterisk/threadstorage.h, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 236613 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec 2009) | 14 lines Merged revisions 236585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces. There was conditional code (based on build platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add a configure-time check for it. ........ ................ * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec 2009) | 19 lines Merged revisions 236509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines Avoid a crash with large numbers of MeetMe conferences. Similar to changes made to Queue(), when we have large numbers of conferences in meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and crash, so instead just use a single fixed buffer. (closes issue #16509) Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ 2009-12-27 18:22 +0000 [r236437] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600 (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 Dec 2009) | 2 lines Turn on colors in the daemon, since there's many requests for it on Ubuntu. ........ ................ 2009-12-26 15:32 +0000 [r236361] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 236358 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec 2009) | 9 lines Merged revisions 236357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec 2009) | 1 line update to latest releases with zero uid/gid ........ ................ 2009-12-23 18:27 +0000 [r236189-236303] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 | tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines AGI may be invoked from outside the dialplan (closes issue #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ * /, res/res_agi.c: Merged revisions 236186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009) | 11 lines Merged revisions 236184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009) | 4 lines If EXEC only gets a single argument, don't crash when the second is used. (closes issue #16504) Reported by: bklang ........ ................ 2009-12-22 17:04 +0000 [r236064] David Vossel * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009) | 18 lines Merged revisions 236062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines fixes issue with p->method incorrectly set to ACK It is possible for a second ACK to come in for a retransmitted message. If an ack does not match an unacked message in our queue, restore the previous p->method as this ACK is completely ignored. (closes issue #16295) Reported by: omolenkamp Patches: issue16295_v2.diff uploaded by dvossel (license 671) ........ ................ 2009-12-21 19:58 +0000 [r235944] Jeff Peeler * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009) | 20 lines Merged revisions 235940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) | 13 lines Change Monitor to not assume file to write to does not contain pathing. 227944 changed the fname_base argument to always append the configured monitor path. This change was necessary to properly compare files for uniqueness. If a full path is given though, nothing needs to be appended and that is handled correctly now. (closes issue #16377) (closes issue #16376) Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch uploaded by dant (license 670) ........ ................ 2009-12-21 17:11 +0000 [r235826] Tilghman Lesher * /, main/features.c: Merged revisions 235822 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009) | 15 lines Merged revisions 235821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009) | 8 lines Send parking lot announcement to the channel which parked the call, not the park-ee. (closes issue #16234) Reported by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded by tilghman (license 14) 20091221__issue16234__1.4.diff.txt uploaded by tilghman (license 14) Tested by: yeshuawatso ........ ................ 2009-12-20 08:58 +0000 [r235775] Alec L Davis * main/dsp.c: restarts busydetector (if enabled) when DTMF is received after call is bridged. (closes issue #16389) Reported by: alecdavis Tested by: alecdavis Patch dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585) 2009-12-18 23:04 +0000 [r235665] Jeff Peeler * main/channel.c, /, include/asterisk/cdr.h: Merged revisions 235660 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009) | 55 lines Merged revisions 235635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is simple in that it reorders the disposition defines so that the fix for issue 12946 works properly (the default CDR disposition was changed to AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all CDR records are written. The side effects of CDR changes are scary, so I'm documenting the test cases performed to attempt to catch any regressions. The following tests were all performed using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A (Both SIP and features) A calls B A blind transfers to C Hangup C (Both SIP and features) A calls B A attended transfers to C Hangup C A calls B A attended transfers to C (SIP) C blind transfers to A (features) Hangup A All of the test scenario CDRs matched. The following tests were performed just with the patch to ensure proper operation (with unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue #16180) Reported by: aatef Patches: bug16180.patch uploaded by jpeeler (license 325) ........ ................ 2009-12-18 22:42 +0000 [r235576-235659] Tilghman Lesher * /, configure, configure.ac: Merged revisions 235656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600 (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18 Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion ........ ................ * /, configure, configure.ac: Merged revisions 235573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235573 | tilghman | 2009-12-18 15:19:43 -0600 (Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009) | 2 lines Point to the typical missing package, not the cryptic "termcap support". ........ ................ 2009-12-17 23:22 +0000 [r235522] Joshua Colp * /, channels/chan_sip.c: Merged revisions 235521 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r235521 | file | 2009-12-17 19:21:07 -0400 (Thu, 17 Dec 2009) | 3 lines Remove some old code for going to the 'fax' extension when a T.38 switchover occurs. This would have already happened when we detected the CNG tone so this was basically a noop. ........ 2009-12-17 Leif Madsen * Release Asterisk 1.6.2.0 2009-12-09 Leif Madsen * Release Asterisk 1.6.2.0-rc8 2009-12-08 18:33 +0000 [r233731] Tilghman Lesher * res/res_musiconhold.c, /: Merged revisions 233718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009) | 8 lines Find another ref leak and change how we manage module references. (closes issue #16388) Reported by: parisioa Patches: 20091208__issue16388.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman Review: https://reviewboard.asterisk.org/r/442/ ........ 2009-12-08 18:04 +0000 [r233694] Russell Bryant * formats/format_sln16.c, formats/format_wav_gsm.c, formats/format_siren7.c, formats/format_ilbc.c, formats/format_vox.c, formats/format_pcm.c, formats/format_h263.c, formats/format_g723.c, formats/format_h264.c, formats/format_siren14.c, formats/format_jpeg.c, formats/format_g726.c, formats/format_gsm.c, formats/format_g729.c, /, formats/format_sln.c, formats/format_wav.c, formats/format_ogg_vorbis.c: Merged revisions 233692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009) | 16 lines Set a module load priority for format modules. A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2. The fix for 1.4 and 1.6.0 will require a different approach since the module load priority functionality is not present in the module API. (issue #16412) Reported by: jiddings ........ 2009-12-08 07:41 +0000 [r233689] TransNexus OSP Development * apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6. 2009-12-07 23:54 +0000 [r233615] Atis Lezdins * contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines Fix compatibility with valgrind 3.3 and older. (noticed in issue #16388) Reported by: parisioa Patches: valgrind.supp uloaded by atis (license 242) Tested by: atis, parisioa ........ 2009-12-07 23:29 +0000 [r233473-233612] David Vossel * /, main/utils.c: Merged revisions 233611 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 | dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines fixes incorrect logic in ast_uri_encode issue #16299 ........ * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009) | 15 lines Merged revisions 233471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) | 9 lines fixes missing Contact header angle brackets (closes issue #16298) Reported by: mgernoth Patches: reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ 2009-12-07 16:16 +0000 [r233396] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 | mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8 lines Do not reject SDP packets describing only non audio streams. (closes issue #16387) Reported by: zalex1953 Patches: media-level-c-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, zalex1953 ........ 2009-12-04 21:55 +0000 [r233281] David Vossel * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600 (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) | 7 lines clarify requirecalltoken option in iax.sample.conf (closes issue #16223) Reported by: bklang Patches: clarify-iax-requirecalltoken.patch uploaded by bklang (license 919) ........ ................ 2009-12-04 21:07 +0000 [r233240] Matthias Nick * pbx/pbx_config.c, /: Merged revisions 233093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 | mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines Parse global variables or expressions in hint extensions Parse global variables or expressions in hint extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166) Reported by: rmudgett Tested by: mnick, rmudgett ........ 2009-12-04 17:36 +0000 [r233165] David Vossel * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600 (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) | 6 lines document and rename strip_control() in app_voicemail (closes issue #16291) Reported by: wdoekes ........ ................ 2009-12-04 17:23 +0000 [r233130] Russell Bryant * main/channel.c, /: Merged revisions 233100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009) | 14 lines Merged revisions 233092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009) | 7 lines Only do frame payload check for HOLD frames. This code was added for helping to debug the source of invalid HOLD frames. However, a side effect of this is that it will incorrectly report errors for frames that have an integer payload. Make the check for this block specific to the HOLD frame case. ........ ................ 2009-12-04 15:57 +0000 [r233049] Matthias Nick * main/dsp.c, /: Merged revisions 233046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) | 17 lines Merged revisions 233014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | 11 lines Warning message gets displayed only once Added additional field 'int display_inband_dtmf_warning', which when set to '1' displays the warning ('Inband DTMF is not supported on codec %s. Use RFC2833'), and when set to '0' doesn't display the warning. Otherwise you would get hundreds of warnings every second. (closes issue #15769) Reported by: falves11 Patches: patch_15769_14.txt uploaded by mnick (license 874) Tested by: mnick, falves11 ........ ................ 2009-12-03 21:03 +0000 [r232866] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600 (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) | 8 lines Deprecate "cz" in favor of "cs". Also, change the use of language codes so that language registers as a prefix, rather than an exact match. (closes issue #16272) Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-12-03 15:14 +0000 [r232813] David Ruggles * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 | diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12 lines Prevent double closing of FDs by EIVR This caused a problem when asterisk was under heavy load and running both AGI and EIVR applications. EIVR would close an FD at which point it would be considered freed and be used by a new AGI instance the second close would then close the FD now in use by AGI. (closes issue #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec Review: https://reviewboard.asterisk.org/r/436/ ........ 2009-12-03 00:20 +0000 [r232675-232678] Tilghman Lesher * res/res_musiconhold.c: Oops, really remove it this time * res/res_musiconhold.c, /: Recorded merge of revisions 232660-232661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232660 | tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 Dec 2009) | 19 lines Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. * Classes are now tracked past removal from the core container, and module removal is actively prevented until all references are freed. * A hanging reference stored in the channel has been removed. This could have caused a mismatch and the music state not properly cleared, if two or more reloads occurred between MOH being stopped and MOH being restarted. * In certain circumstances, duplicate classes were possible. * A race existed at reload time between a process being killed and the thread responsible for reading from the related pipe respawning that process. * Several reference counts have also been corrected. At least one could have caused deleted classes to stick around forever, consuming resources. This originally manifested as MOH external processes that were not killed at reload time. (closes issue #16279, closes issue #16207) Reported by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman ........ r232661 | tilghman | 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove debugging line ........ 2009-12-02 23:28 +0000 [r232658] David Vossel * CHANGES, /, UPGRADE.txt: Merged revisions 232657 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232657 | dvossel | 2009-12-02 17:27:45 -0600 (Wed, 02 Dec 2009) | 6 lines update CHANGES and UPGRADE.txt for early media behavior change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by: miki ........ 2009-12-02 22:05 +0000 [r232579-232585] Jeff Peeler * main/manager.c, /: Merged revisions 232582 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009) | 14 lines Merged revisions 232581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009) | 7 lines Send ack (response/message) after receiving manager action userevent (closes issue #16264) Reported by: dimas Patches: event-ack.patch uploaded by dimas (license 88) ........ ................ * main/manager.c, /: Merged revisions 232576 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 | jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines Make manager response to "Action: events" finish with empty line (closes issue #16275) Reported by: vnovy Patches: manager.c.diff uploaded by vnovy (license 922) ........ 2009-12-02 17:11 +0000 [r232359] Joshua Colp * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) | 12 lines Merged revisions 232355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG. (closes issue #16239) Reported by: CGMChris ........ ................ 2009-12-02 17:01 +0000 [r232352] David Vossel * /, main/acl.c: Merged revisions 232351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009) | 12 lines Merged revisions 232350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009) | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in strace. (closes issue #16290) Reported by: wdoekes ........ ................ 2009-12-02 16:43 +0000 [r232348] Joshua Colp * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 | file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add support for handling the 415 Unsupported media type response like we do for a 488 Not acceptable here response. (closes issue #16186) Reported by: atis Patches: sip_t38_response_415.patch uploaded by atis (license 242) ........ 2009-12-02 15:43 +0000 [r232270] David Vossel * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600 (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) | 9 lines fixes segfault in func_groupcount closes issue #16337) Reported by: Parantido Patches: issue_16337.diff uploaded by dvossel (license 671) Tested by: Parantido, dvossel ........ ................ 2009-12-02 14:55 +0000 [r232232] Joshua Colp * /, channels/chan_sip.c: Merged revisions 232230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 | file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where a scheduled item ID would get retained on registrations in a certain scenario causing code to execute during reload that should not. (issue AST-263) ........ 2009-12-02 00:52 +0000 [r232094] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600 (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines Do not modify the gain settings on data calls. (The digital flag actually represents a data call.) (closes issue #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ 2009-12-01 23:40 +0000 [r232011-232015] Russell Bryant * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 | russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines Fix a build error on FreeBSD. ........ * /, main/file.c: Merged revisions 232008 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 232007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) | 2 lines Fix a warning pointed out by buildbot. ........ ................ 2009-12-01 22:03 +0000 [r231930] Jeff Peeler * main/channel.c, /: Merged revisions 231927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009) | 19 lines Merged revisions 231911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) | 12 lines Fix crash with invalid frame data The crash was happening as a result of a frame containing an invalid data pointer, but was set with data length of zero. The few times the issue was reproduced it _seemed_ that the frame was queued properly, that is the data pointer was set to NULL. I never could reproduce the crash so as a last resort the crash has been fixed, but a check in __ast_read has been added to give as much information about the source of problematic frames in the future. (closes issue #16058) Reported by: atis ........ ................ 2009-12-01 21:21 +0000 [r231870] David Vossel * main/pbx.c, /: Merged revisions 231867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 231853 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009) | 3 lines WaitExten m option with no parameters generates frame with zero datalen but non-null data ptr ........ ................ 2009-12-01 Leif Madsen * Release Asterisk 1.6.2.0-rc7 2009-12-01 15:48 +0000 [r231743] Matthew Nicholson * /, main/file.c: Merged revisions 231741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 231740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce() and return an error if no know formats are found. ........ ................ 2009-11-30 21:59 +0000 [r231695-231696] Kevin P. Fleming * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 231692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines Another round of UDPTL stack fixes/improvements: 1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL session, so that log/error/debug messages generated by the UDPTL stack can be 'connected' to the endpoint that caused them to be generated. 2) Improve comments (and process) of calculating the far end's maximum IFP size when redundancy mode is in use for error correction. 3) When an IFP larger than the calculated 'far max IFP' size is presented for writing, truncate it rather than putting in the buffer and allowing the buffer to overflow; this will cause the ends to retrain to a lower bit rate that produces IFPs of an appropriate size if possible, and if not possible, the FAX transfer will fail completely. In these cases, it is due to the one endpoint supplying a T38FaxMaxDatagram value that is improperly calculated and is too low to be of use; we have configuration options available to override this behavior. 4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer needed. ........ * pbx/pbx_config.c: Backport a tiny fix from trunk that makes GCC 4.4.x happier. 2009-11-30 21:36 +0000 [r231689] Matthew Nicholson * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c, main/app.c: Merged revisions 231688 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov 2009) | 15 lines Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ ................ 2009-11-30 20:47 +0000 [r231605] Joshua Colp * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 | file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines When receiving SDP that matches the version of the last one do not treat it as a fatal error. (closes issue #16238) Reported by: seandarcy ........ 2009-11-30 18:57 +0000 [r231505-231558] David Vossel * apps/app_queue.c, /: Merged revisions 231556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 | dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines app_queue crashes randomly, often during call-transfers This patch adds a ref to the queue_ent object's parent call_queue in queue_exec() so the call_queue won't be destroyed while the the queue_ent still holds a pointer to it. (closes issue 0015686) Tested by: dvossel, aragon ........ * main/rtp.c, /: Merged revisions 231491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009) | 17 lines Merged revisions 231441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009) | 11 lines fixes crash caused by RTP comfort noise payload greater than 24 bytes AST-2009-010 (closes issue #16242) Reported by: amorsen Patches: issue16242.diff uploaded by oej (license 306) Tested by: amorsen, oej, dvossel ........ ................ 2009-11-25 22:34 +0000 [r231302] Tilghman Lesher * main/channel.c, /: Merged revisions 231299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009) | 9 lines Merged revisions 231298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) | 2 lines After a frame duplication failure, unlock the channel before returning. ........ ................ 2009-11-25 15:48 +0000 [r231191] Matthew Nicholson * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 | mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4 lines Load pbx_lua with global symbols to allow linking with other lua libraries. Found by Maxim Litnitskiy. ........ 2009-11-24 20:36 +0000 [r231136] Tilghman Lesher * apps/app_queue.c, /: Merged revisions 231134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 | tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines Found a few places where queue refcounts were counted incorrectly. Also add debug statements. (closes issue #15982, closes issue #15984) Reported by: atis Patches: 20091111__issue15982.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ 2009-11-24 18:54 +0000 [r231098] Jeff Peeler * /, main/features.c: Merged revisions 231095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 | jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines Fix erroneous hangup extension execution ast_spawn_extension behaves differently from 1.4 in that hangups and extensions that do not exist do not return an error, whereas in 1.6 it does. This is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN flag gets set properly. (closes issue #16106) Reported by: ajohnson Tested by: ajohnson ........ 2009-11-23 15:48 +0000 [r230884] Joshua Colp * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 230881 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines Change fax detection in chan_sip so it behaves as one would expect. Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. ........ 2009-11-23 15:38 +0000 [r230796-230880] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov 2009) | 9 lines Merged revisions 230839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov 2009) | 1 line Correct fix for issue #16268... the reporter's original patch was very close to correct. ........ ................ * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov 2009) | 12 lines Merged revisions 230772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov 2009) | 5 lines Ensure that SDP parsing does not ignore the last line of the SDP. (closes issue #16268) Reported by: sgimeno ........ ................ 2009-11-20 22:36 +0000 [r230727] David Vossel * channels/chan_iax2.c, /: Merged revisions 230726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009) | 7 lines fixes iax2 show cache locking error, thanks alecdavis! (closes issue #16094) Reported by: alecdavis Patches: bug16094.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, dvossel ........ 2009-11-20 21:07 +0000 [r230629] Matthew Nicholson * /, main/features.c: Merged revisions 230628 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov 2009) | 15 lines Merged revisions 230627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR if it exists. This is necessary for the recordagentcalls option in chan_agent to store the recorded file name in the bridge CDR. (closes issue #14590) Reported by: msetim Patches: queue_agent_userfield.patch uploaded by Laureano (license 265) Tested by: Laureano, mnicholson ........ ................ 2009-11-20 17:31 +0000 [r230510-230585] David Vossel * main/audiohook.c, /, include/asterisk/audiohook.h: Merged revisions 230583 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 | dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines audiohook signal trigger on every status change (issue #14618) Review: https://reviewboard.asterisk.org/r/434/ ........ * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600 (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) | 10 lines fixes MixMonitor thread not exiting when StopMixMonitor is used (closes issue #16152) Reported by: AlexMS Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, AlexMS Review: https://reviewboard.asterisk.org/r/424/ ........ ................ 2009-11-16 16:41 +0000 [r230250-230384] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 230381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230381 | kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line Fix another buglet in T.38 session teardown at the end of FAX sessions. ........ * /, apps/app_fax.c: Merged revisions 230343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230343 | kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2 lines Ensure that only one end of a T.38 session initiates teardown at completion. ........ * channels/chan_iax2.c, /: Merged revisions 230247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230247 | kpfleming | 2009-11-15 11:23:02 -0600 (Sun, 15 Nov 2009) | 12 lines Merged revisions 230246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov 2009) | 6 lines Correct mistaken option name in error message. The configuration option for allowing hosts to make non-token-based calls is 'calltokenoptional', not 'calltokenignore'. (reported on asterisk-users) ........ ................ 2009-11-13 22:01 +0000 [r229969-230148] Joshua Colp * /, channels/chan_sip.c: Merged revisions 230145 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) | 15 lines Merged revisions 230144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8 lines Respect the maddr parameter in the Via header. (closes issue #14446) Reported by: frawd Patches: via_maddr.patch uploaded by frawd (license 610) Tested by: frawd ........ ................ * channels/chan_local.c, /: Merged revisions 230039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230039 | file | 2009-11-13 13:44:53 -0600 (Fri, 13 Nov 2009) | 16 lines Merged revisions 230038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9 lines Fix a crash caused by two threads thinking they should both free the chan_local private structure when only one should. (closes issue #15314) Reported by: sroberts Patches: Issue15314_Move_Nulling_owner.patch uploaded by davidw (license 780) Tested by: davidw, lottc ........ ................ * configs/extensions.conf.sample, /, apps/app_chanisavail.c: Merged revisions 229966 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) | 13 lines Merged revisions 229965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6 lines Document a limitation in the AVAILSTATUS variable from ChanIsAvail and provide a workaround for it that does not change existing behavior. (closes issue #14426) Reported by: macli ........ ................ 2009-11-13 Leif Madsen * Release Asterisk 1.6.2.0-rc6 2009-11-13 15:57 +0000 [r229915] Joshua Colp * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 | file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix T.38 negotiation regression introduced with the SDP parser changes. ........ 2009-11-12 23:31 +0000 [r229752] Jason Parker * channels/chan_oss.c, /: Merged revisions 229750 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 | qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix mute toggling on OSS channels. ........ 2009-11-12 16:47 +0000 [r229671] David Vossel * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600 (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009) | 6 lines fixes merging error, datastore was being freed in the wrong function. (closes issue #16219) Reported by: aragon ........ ................ 2009-11-11 20:49 +0000 [r229570] David Ruggles * doc/externalivr.txt: Merged revisions 229568 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229568 | diruggles | 2009-11-11 15:47:06 -0500 (Wed, 11 Nov 2009) | 9 lines Remove non-functional feature from ExternalIVR documentation Remove non-functional socket implementation of ExternalIVR from documentation (closes issue #16225) Reported by: thedavidfactor Patches: externalivr.txt.20091111.1542.patch uploaded by thedavidfactor (license 903) ........ 2009-11-11 19:56 +0000 [r229492-229502] David Brooks * main/pbx.c, /: Merged revisions 229499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009) | 15 lines Merged revisions 229498 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009) | 8 lines Solaris doesn't like NULL going to ast_log Solaris will crash if NULL is passed to ast_log. This simple patch simply uses S_OR to get around this. (closes issue #15392) Reported by: yrashk ........ ................ * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009) | 7 lines Flags not initialized in app_softhangup.c, causing undefined behavior Trivial patch [kobaz] to initialize an ast_flags = {0} (closes issue #16129) Reported by: kobaz ........ 2009-11-10 22:17 +0000 [r229366] Tilghman Lesher * main/pbx.c, /: Merged revisions 229361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009) | 19 lines Merged revisions 229360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) | 12 lines If two pattern classes start with the same digit and have the same number of characters, they will compare equal. The example given in the issue report is that of [234] and [246], which have these characteristics, yet they are clearly not equivalent. The code still uses these two characteristics, yet when the two scores compare equal, an additional check will be done to compare all characters within the class to verify equality. (closes issue #15421) Reported by: jsmith Patches: 20091109__issue15421__2.diff.txt uploaded by tilghman (license 14) Tested by: jsmith, thedavidfactor ........ ................ 2009-11-10 22:04 +0000 [r229359] David Ruggles * doc/externalivr.txt: Merged revisions 229356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov 2009) | 16 lines Merged revisions 229355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov 2009) | 9 lines Fix ExternalIVR Documentation Remove documentation for event that doesn't function (closes issue #16220) Reported by: thedavidfactor Patches: externalivr.txt.20091110.1622.patch uploaded by thedavidfactor (license 903) ........ ................ 2009-11-10 21:33 +0000 [r229354] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 | tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines When GOSUB is invoked within an AGI, it may not exit correctly. (closes issue #16216) Reported by: atis Patches: 20091110__atis_work.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ 2009-11-10 20:09 +0000 [r229285] Joshua Colp * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) | 15 lines Merged revisions 229281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8 lines Remove broken support for direct transcoding between G.726 RFC3551 and G.726 AAL2. On some systems the translation core would actually consider g726aal2 -> g726 -> signed linear to be a quicker path then g726aal2 -> signed linear which exposed this problem. (closes issue #15504) Reported by: globalnetinc ........ ................ 2009-11-10 17:52 +0000 [r229232] David Vossel * channels/chan_iax2.c, /: Merged revisions 229168 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600 (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009) | 9 lines don't crash on log message in solaris AST-2009-006 (closes issue #16206) Reported by: bklang Tested by: bklang ........ ................ 2009-11-10 17:39 +0000 [r229231] David Ruggles * doc/externalivr.txt: Merged revisions 229228 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov 2009) | 18 lines Merged revisions 229191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov 2009) | 11 lines Document ExternalIVR event tag collision ExternalIVR uses the D tag for two different event types. This documents that behavior and how to differentiate between the two cases. Also includes a minor spelling fix and clarification (closes issue #16211) Reported by: thedavidfactor Patches: externalivr.txt.20091109.1507.patch uploaded by thedavidfactor (license 903) ........ ................ 2009-11-10 15:47 +0000 [r229101] Matthew Nicholson * UPGRADE-1.6.txt, main/editline/makelist.in, UPGRADE.txt: Reset props that were accidently deleted in 229088. 2009-11-10 15:28 +0000 [r229094] David Vossel * res/res_config_pgsql.c, /: Merged revisions 229093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229093 | dvossel | 2009-11-10 09:27:45 -0600 (Tue, 10 Nov 2009) | 11 lines fixes pgsql double free of threadstorage A thread storage variable was being freed incorrectly, which resulted in a double free if two queries were made in the same thread. (closes issue #16011) Reported by: cristiandimache Patches: issue16011.diff uploaded by dvossel (license 671) ........ 2009-11-10 15:16 +0000 [r229088] Matthew Nicholson * UPGRADE-1.6.txt, main/editline/makelist.in, channels/chan_sip.c, UPGRADE.txt: Reverted revision 202007. (closes issue #16175) Reported by: paul-tg Tested by: paul-tg 2009-11-10 11:25 +0000 [r229078] Gavin Henry * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10 Nov 2009) | 20 lines Schema file additions * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses to allow standalone dialplan, account and mailbox entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir, - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap, - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed redundant IPaddr (there's already IPAddress) - Gives more configuration Flags for SIP-Users available (tested) - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses without extensibleObject (which really should be the last resort); gives also additional possibilities for LDAP-filter (closes issue #15874) Reported by: Medozas Patches: asterisk.ldap-schema.patch uploaded by Medozas (license 41) Tested by: Medozas, suretec ........ 2009-11-09 22:59 +0000 [r229017] Terry Wilson * channels/chan_local.c, /: Merged revisions 229015 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009) | 8 lines Don't crash when bridge->tech_pvt == NULL This is a similar solution to what is in place for chan_agent (closes issue #16003) Reported by: atis Tested by: twilson ........ 2009-11-09 22:17 +0000 [r229012] David Vossel * channels/chan_sip.c: fixes segfault when transferring a queue caller In sip_hangup we attempted to lock p->owner after we set it to NULL. Thanks to fhackenberger for reporting the issue and submitting a patch. (closes issue #15848) Reported by: fhackenberger Patches: digium_bug_0015848 uploaded by fhackenberger (license 592) Tested by: fhackenberger, lmadsen, TomS, shin-shoryuken, dvossel 2009-11-09 Leif Madsen * Release Asterisk 1.6.2.0-rc5 2009-11-09 15:40 +0000 [r228900] Leif Madsen * main/channel.c: Merged revisions 228897 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009) | 14 lines Merged revisions 228896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009) | 6 lines Update WARNING message. Update a WARNING message to give a suggested fix when encountered. (closes issue #16198) Reported by: atis Tested by: atis ........ ................ 2009-11-09 14:48 +0000 [r228859] Matthew Nicholson * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600 (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov 2009) | 8 lines Perform limited bounds checking when destroying ast_mutex_t structures to make sure we don't try to use negative indices. (closes issue #15588) Reported by: zerohalo Patches: 20090820__issue15588.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ 2009-11-06 22:37 +0000 [r228694] David Vossel * main/channel.c, /: Merged revisions 228693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009) | 16 lines Merged revisions 228692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009) | 9 lines fixes audiohook write crash occuring in chan_spy whisper mode. After writing to the audiohook list in ast_write(), frames were being freed incorrectly. Under certain conditions this resulted in a double free crash. (closes issue #16133) Reported by: wetwired ........ ................ 2009-11-06 20:26 +0000 [r228649] Matthew Nicholson * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600 (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov 2009) | 8 lines Properly handle '=' while decoding base64 messages and null terminate strings returned from BASE64_DECODE. (closes issue #15271) Reported by: chappell Patches: base64_fix.patch uploaded by chappell (license 8) Tested by: kobaz ........ ................ 2009-11-06 18:43 +0000 [r228551] Joshua Colp * /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) | 11 lines Merged revisions 228547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4 lines Don't overwrite caller ID name on a trunk with the configured fullname when using users.conf (issue ABE-1989) ........ ................ 2009-11-06 Leif Madsen * Release Asterisk 1.6.2.0-rc4 2009-11-06 17:54 +0000 [r228504] Joshua Colp * doc/tex/localchannel.tex, /: Merged revisions 228499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2 lines Fix the localchannel.tex file. ........ 2009-11-06 17:24 +0000 [r228421-228447] David Vossel * codecs/codec_ilbc.c, /: Merged revisions 228441 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 | dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines Fixes merging issue from 1.4, frame data is held in data.ptr in trunk ........ * codecs/codec_ilbc.c, /: Merged revisions 228420 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009) | 19 lines Merged revisions 228418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009) | 13 lines fixes segfault in iLBC For reasons not yet known, it appears possible for an ast_frame to have a datalen greater than zero while the actual data is NULL during Packet Loss Concealment. Most codecs don't support PLC so this doesn't affect them. This patch catches the malformed frame and prevents the crash from occuring. Additional efforts to determine why it is possible for a frame to look like this are still being investigated. (issue #16979) ........ ................ 2009-11-06 16:44 +0000 [r228413] Joshua Colp * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) | 14 lines Merged revisions 228409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 lines Fix a bug caused by a partially invalid frame (from the jitterbuffer) passing through the Asterisk core. (closes issue #15560) Reported by: jvandal (closes issue #15709) Reported by: covici ........ ................ 2009-11-06 15:43 +0000 [r228269-228340] David Vossel * /, main/astfd.c: Merged revisions 228339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009) | 12 lines Merged revisions 228338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) | 5 lines fixes crash in astfd.c (closes issue #15981) Reported by: slavon ........ ................ * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06 Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c (closes issue #15394) Reported by: boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790) Tested by: dbrooks, boroda ........ 2009-11-05 22:13 +0000 [r228198] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 | tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines Yet another error message in the dialplan (thanks, rmudgett/russellb) ........ 2009-11-05 21:27 +0000 [r228195] Jeff Peeler * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 | jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines Fix the fix for chanspy option o In 224178, I assumed the uploaded patch was correct as it had received positive feedback. The flags were being checked in the incorrect location. Upon testing the fix this time it was also found that the flags from the dialplan weren't being copied to the chanspy_translation_helper. (closes issue #16167) Reported by: marhbere ........ 2009-11-05 21:27 +0000 [r228194] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 | tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines MEETME_INFO should not return a literal error message to the dialplan. (closes issue #15450) Reported by: JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested by: JimVanM ........ 2009-11-05 19:42 +0000 [r228148] David Brooks * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600 (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r ........ ................ 2009-11-05 19:20 +0000 [r228093] Jason Parker * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228080 | qwell | 2009-11-05 13:16:29 -0600 (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) | 8 lines Fix crash on VPB exception when no hardware is present. (closes issue #14970) Reported by: tzafrir Patches: vpb_exception.diff uploaded by tzafrir (license 46) Tested by: markwaters ........ ................ 2009-11-05 17:14 +0000 [r228017] Tilghman Lesher * apps/app_externalivr.c, /: Merged revisions 228015 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009) | 4 lines Don't crash if no arguments are passed. (closes issue #16119) Reported by: thedavidfactor ........ 2009-11-04 23:53 +0000 [r227947] Jeff Peeler * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009) | 21 lines Merged revisions 227944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) | 14 lines Fix incorrect filename comparsion after monitor file change The logic to detect if a requested file is indeed a different file from the current file was incorrect. The main issue being confusion of the use of filename_base which was previously set without pathing information and then compared to another full path. Robust file comparison logic has been added to properly check if two files are the same even if symlinks are used. (closes issue #15313) Reported by: caspy Patches: 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325) but mostly tilghman's work ........ ................ 2009-11-04 21:09 +0000 [r227760-227831] Matthew Nicholson * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17 lines Merged revisions 227827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements. (closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ ........ ................ * channels/chan_sip.c: Modify the SDP parsing code to parse session and media level items separately. With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future. (closes issue #14994) Reported by: frawd 2009-11-04 19:28 +0000 [r227733-227748] Joshua Colp * /, static-http/prototype.js: Merged revisions 227739 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where it may be possible for someone to execute a cross-site AJAX request exploit. (AST-2009-009) ........ ................ * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where sending a REGISTER with a differing username in the From URI and Authorization header would reveal whether it was valid or not. (AST-2009-008) ........ ................ 2009-11-03 20:01 +0000 [r227375] Jason Parker * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) | 9 lines Fix some build issues on Solaris. (closes issue #14517) (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell ........ 2009-11-03 19:49 +0000 [r227364-227371] Leif Madsen * apps/app_controlplayback.c, /: Merged revisions 227368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03 Nov 2009) | 8 lines Change warning message to debug message. app_controlplayback outputs a warning, when in fact it is normal. (closes issue #16071) Reported by: atis Patches: controlplayback_warning.patch uploaded by atis (license 242) ........ * configs/extensions.conf.sample, /: Merged revisions 227361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines Additional fixes to the extensions.conf.sample file. Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville ........ 2009-11-03 18:15 +0000 [r227280] Richard Mudgett * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) | 4 lines Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. This is the relevant portion of asterisk/trunk -r226648 ........ 2009-11-03 17:14 +0000 [r227239] David Vossel * /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 | dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines user.conf entries in SIP were not having their peer type set. (closes issue #16120) Reported by: jsmith ........ 2009-11-03 15:40 +0000 [r227170] Joshua Colp * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) | 12 lines Merged revisions 227166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines Fix a bug where an RPID header could be generated with a blank username in the URI. (closes issue #15909) Reported by: kobaz ........ ................ 2009-11-03 15:25 +0000 [r227165] Leif Madsen * configs/extensions.conf.sample, /: Merged revisions 227162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03 Nov 2009) | 7 lines Update extensions.conf.sample file to fix incorrect extensions. (closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville ........ 2009-11-03 13:51 +0000 [r227156] Olle Johansson * Makefile, /, channels/chan_sip.c: Merged revisions 227091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines Use proper response code when violating Contact ACL's. https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) ........ ................ 2009-11-02 21:06 +0000 [r226978] David Brooks * channels/chan_sip.c: SIP channel name uniqueness SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ 2009-11-02 18:12 +0000 [r226893] Joshua Colp * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines Merged revisions 226889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) ........ ................ 2009-11-02 17:17 +0000 [r226815] Tilghman Lesher * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600 (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) | 8 lines Don't allow two separate instances of safe_asterisk when restarting from the init script. (closes issue #14562) Reported by: davidw Patches: Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14) Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780) Tested by: davidw ........ ................ 2009-10-29 18:18 +0000 [r226540] Joshua Colp * doc/tex/localchannel.tex, channels/chan_local.c, /: Merged revisions 226532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | 13 lines Merged revisions 226531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ ................ 2009-10-28 21:32 +0000 [r226486] Tzafrir Cohen * build_tools/get_documentation, /: remove empty awk pattern (//) Solaris 10 nawk doesn't like the empty pattern such as '//' for 'always'. Just remove that. No pattern at all always matches. Merged revisions 226453 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-10-28 20:13 +0000 [r226379-226385] Leif Madsen * configs/sip.conf.sample: Merged revisions 226384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ ................ * doc/tex/channelvariables.tex: Merged revisions 226378 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500 (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines Update CALLINGSUBADDR channel variable documentation. (closes issue #15734) Reported by: alecdavis Patches: channelvariables.tex.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ 2009-10-28 18:06 +0000 [r226170-226308] Tilghman Lesher * /, include/asterisk/linkedlists.h: Merged revisions 226305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500 (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines Fix documentation (pointed out by TheDavidFactor on #-dev) ........ ................ * main/manager.c, /: Merged revisions 226159 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009) | 14 lines Merged revisions 226138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines Manager output is not always NULL-terminated, so force a NULL at the end of the filestream. (closes issue #15495) Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded by tilghman (license 14) Tested by: pdf ........ ................ 2009-10-27 17:12 +0000 [r226101] Terry Wilson * res/res_http_post.c, /: Merged revisions 226099 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 | twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines Don't prepend the URI prefix to the post directory ........ 2009-10-27 00:16 +0000 [r226055] Tzafrir Cohen * /, configure, configure.ac: detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even if host_os is linux-gnueabi * When checking if we are Linux, check OSARCH rather than host_os The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is tested for the value of 'linux-gnu' in one or two places in the tree. This patch also fixes the check libcap to check for $OSARCH rather than $host_os . See also: http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4 Merged revisions 226018 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-10-26 19:42 +0000 [r225914] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 | jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines ACL check not present for verifying SIP INVITEs The ACL check in check_peer_ok was missing and has now been restored. The missing check allowed for calls to be made on prohibited networks where an ACL was defined in sip.conf and the allowguest option was set to off. See the AST security advisory below for more information. Merge code associated with AST-2009-007. (closes issue #16091) Reported by: thom4fun ........ 2009-10-26 15:56 +0000 [r225871] Kevin P. Fleming * apps/app_fax.c: Backport audio handling loop fixes from trunk version of app_fax. This backport resolves some issues handling audio frames during FAX processing, and ensures that the FAX application doesn't accidentally get notified of a T.38 switchover at the end of a successful FAX. (closes issue #16127) 2009-10-23 14:46 +0000 [r225651] David Vossel * /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 | dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines Fixes an iterator memory leak and uninitialized memory ........ 2009-10-23 14:08 +0000 [r225585] Kevin P. Fleming * Makefile, /: Merged revisions 225582 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct 2009) | 17 lines Merged revisions 225581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on every build. For some reason the menuselect.makeopts file was listed as PHONY in the Makefile, resulting in 'make' needing to rebuild it for every build. This then resulted in the embedded module rules being rebuilt on every build, which can be slow and is unnecessary. This patch fixes the problem by properly allowing 'make' to know when the menuselect.makeopts file needs to be rebuilt (defining the proper dependencies). ........ ................ 2009-10-22 22:24 +0000 [r225516] Leif Madsen * README, /: Merged revisions 225515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225515 | lmadsen | 2009-10-22 17:24:03 -0500 (Thu, 22 Oct 2009) | 8 lines Update README documentation. Update the README documentation to correctly describe which CLI command you should use when attempting to get help from the CLI. (closes issue #16064) Reported by: thedavidfactor Patches: readme.patch uploaded by thedavidfactor (license 903) ........ 2009-10-22 21:55 +0000 [r225489] David Vossel * apps/app_externalivr.c, include/asterisk/tcptls.h, main/tcptls.c, /, channels/chan_sip.c: Merged revisions 225445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ ........ 2009-10-22 21:54 +0000 [r225488] Leif Madsen * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions 225485 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009) | 19 lines Merged revisions 225484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines Clean valgrind output by suppressing false errors. Update valgrind.txt documentation and add valgrind.supp file in order to allow those who are creating valgrind output to have less false errors in the logfile. (closes issue #16007) Reported by: atis Patches: valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp uploaded by atis (license 242) Tested by: atis, amorsen ........ ................ 2009-10-22 17:14 +0000 [r225363] Tilghman Lesher * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h: Merged revisions 225360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ 2009-10-21 22:00 +0000 [r225035-225308] David Vossel * channels/chan_iax2.c, /: Merged revisions 225307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ ................ * configs/sip.conf.sample, channels/chan_iax2.c, configs/iax.conf.sample, /, channels/chan_sip.c: Merged revisions 225033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines Merged revisions 225032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ ................ 2009-10-20 22:11 +0000 [r224859] Tilghman Lesher * main/audiohook.c, funcs/func_speex.c, /: Merged revisions 224856 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ ................ 2009-10-20 17:50 +0000 [r224777] Joshua Colp * /, main/features.c: Merged revisions 224774 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ ................ 2009-10-20 00:00 +0000 [r224674] Kevin P. Fleming * main/rtp.c, /: Merged revisions 224671 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct 2009) | 14 lines Merged revisions 224670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ ................ 2009-10-19 19:54 +0000 [r224571] Joshua Colp * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines Merged revisions 224565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ ................ 2009-10-19 19:41 +0000 [r224563] Kevin P. Fleming * formats/format_siren14.c, /: Merged revisions 224562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224562 | kpfleming | 2009-10-19 14:40:26 -0500 (Mon, 19 Oct 2009) | 1 line Remove useless debugging message. ........ 2009-10-19 00:13 +0000 [r224447-224451] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) | 3 lines Allow ODBC storage to be queried with multiple mailboxes, and remove multiple goto's. This corrects an issue reported on the -users list. ........ * configs/res_odbc.conf.sample, /: Merged revisions 224446 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18 Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias for "autocommit", but instead controls the default disposition of uncommitted transactions. ........ 2009-10-17 01:58 +0000 [r224334] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ ................ 2009-10-16 20:58 +0000 [r224264] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500 (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ ................ 2009-10-15 15:58 +0000 [r224181] Jeff Peeler * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 | jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines Readd removed ability to allow listening to one side of the call in app_chanspy (Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html ........ 2009-10-12 23:55 +0000 [r223835] Jeff Peeler * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines Merged revisions 223804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ ................ 2009-10-12 21:01 +0000 [r223757] David Vossel * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options SWP-151 ........ 2009-10-12 14:37 +0000 [r223655] Kevin P. Fleming * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque ........ 2009-10-11 17:32 +0000 [r223490] Russell Bryant * main/autoservice.c, /: Merged revisions 223487 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009) | 17 lines Merged revisions 223485-223486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines Don't use data outside of its scope. The purpose of this code was to have a hangup frame put on the list of deferred frames. However, the code that read the hangup frame was outside of the scope of where the hangup frame was declared. ........ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines Remove some unnecessary code. ........ ................ 2009-10-09 23:12 +0000 [r223406] Jeff Peeler * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation of PRIREDIRECTIONREASON set by chan_sip. This commit is the simplest way to solve a problem that has already been solved in trunk with the "COLP/CONP and Redirecting party information into Asterisk" commit. In trunk the redirection reason is translated into a generic redirect reason. I would have had to do the same fix except chan_sip never reads PRIREDIRECTREASON. So both chan_dahdi and chan_h323 have been modified to interpret the one different redirect reason of "no-answer" properly and set the ISDN reason code 2 of "no reply". (closes issue #15033) Reported by: steinwej 2009-10-09 21:01 +0000 [r223333] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines Initiate T.38 switchover when acting as called party, regardless of FAX direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque ........ 2009-10-09 18:53 +0000 [r223286] Matthew Nicholson * main/channel.c, /: Merged revisions 223273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct 2009) | 14 lines Merged revisions 223225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING when originating calls. (closes issue #15104) Reported by: nblasgen Patches: manager-timeout1.diff uploaded by mnicholson (license 96) Tested by: nblasgen, mnicholson ........ ................ 2009-10-09 18:29 +0000 [r223257] Mark Michelson * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c ........ ................ 2009-10-09 17:55 +0000 [r223208] David Vossel * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines Merged revisions 223205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines fixes sip registration using authuser in user.conf (closes issue #14954) Reported by: tornblad Tested by: mmichelson, tornblad, dvossel ........ ................ 2009-10-09 17:27 +0000 [r223173] Matthew Nicholson * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct 2009) | 8 lines Don't close the sqlite database when reloading. Only close the database when unloading. (closes issue #15953) Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by frawd (license 610) Tested by: frawd ........ 2009-10-09 17:09 +0000 [r223089-223133] David Vossel * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 | dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines 'auth=' did not parse md5 secret correctly (closes issue #15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad ........ * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 | dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines p->peerauth is always empty in transmit_register() When using callbackextension or specifing the peer name in a registration string, the peer's specific auth settings set by the "auth=" strings within the peer definition are not used by the registration. Thanks to ebroad for reporting the issue and providing the patch. (closes issue #15955) Reported by: ebroad Patches: regauthfix.patch uploaded by ebroad (license 878) ........ 2009-10-08 20:00 +0000 [r222883] Russell Bryant * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c, /, main/file.c: Merged revisions 222880 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ ................ 2009-10-08 19:41 +0000 [r222874] David Vossel * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions 222873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 | dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines fixes an ast_netsock_list memory leak. ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ ........ 2009-10-08 16:51 +0000 [r222695-222802] Richard Mudgett * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500 (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) | 12 lines Fix memory leak if chan_misdn config parameter is repeated. Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 ........ ................ * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500 (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 ........ ................ 2009-10-07 18:06 +0000 [r222549] Jason Parker * /, configs/queues.conf.sample: Merged revisions 222548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct 2009) | 5 lines Remove 'keepstats' queue option from sample config, as it's no longer used. https://reviewboard.asterisk.org/r/115/ (closes issue #15820) Reported by: kshumard ........ 2009-10-07 18:00 +0000 [r222547] Sean Bright * funcs/func_strings.c: Fix merge error. 2009-10-07 17:45 +0000 [r222544] David Vossel * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009) | 14 lines Merged revisions 222542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) | 8 lines crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) ........ ................ 2009-10-06 23:59 +0000 [r222354-222466] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500 (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) | 8 lines Add missing unlock(s) in dahdi_read (two cases in trunk, and 1.6.2) (closes issue #15683) Reported by: alecdavis ........ ................ * channels/chan_dahdi.c: Fix potential crash when entire span request is received. The variable index used in this scenario for accessing the dahdi_pvts was wrong and was most likely copied from the several other places it is used correctly. (closes issue #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch uploaded by tsearle (license 373) * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009) | 9 lines Fix 222298 (crash during destruction of second channel when variable set with setvar). I mistakenly reasoned that setvar would be used on all channels. Since it can be set per channel, give each dahdi channel a copy of the variable. (related to #15899) ........ 2009-10-06 19:41 +0000 [r222311] Tilghman Lesher * cdr/cdr_pgsql.c, res/res_config_pgsql.c, /: Merged revisions 222309 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 | tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10 lines Change schema query to involve the use of an optional schema parameter. This change is done in such a way as to allow the driver to continue to function with older databases which don't have these features. (closes issue #16000) Reported by: jamicque Patches: 20091002__issue16000.diff.txt uploaded by tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ 2009-10-06 19:27 +0000 [r222304] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009) | 9 lines Fix crash during destruction of second channel when variable set with setvar. The setvar line in chan_dahdi.conf is shared among all the channels, so make sure to only free the resources only when the last channel is destroyed. (closes issue #15899) Reported by: tzafrir ........ 2009-10-06 19:22 +0000 [r222289] Tilghman Lesher * res/ael/pval.c, /: Merged revisions 222273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 | tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines When we call a gosub routine, the variables should be scoped to avoid contaminating the caller. This affected the ~~EXTEN~~ hack, where a subroutine might have changed the value before it was used in the caller. Patch by myself, tested by ebroad on #asterisk ........ 2009-10-06 Leif Madsen * Released Asterisk 1.6.2.0-rc3 2009-10-06 01:39 +0000 [r222113-222187] Kevin P. Fleming * channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c, channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c, res/res_clialiases.c, /, channels/chan_sip.c, funcs/func_dialgroup.c, include/asterisk/astobj2.h, res/res_phoneprov.c: Merged revisions 222176 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ ................ * configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c, configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 222110 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ 2009-10-02 17:35 +0000 [r222032] David Vossel * channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a memcpy. ........ ................ 2009-10-02 17:01 +0000 [r221923-221974] Tilghman Lesher * main/astobj2.c, /: Merged revisions 221971 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 221970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines Ensure the result of the hash function is positive. Negative array offsets suck. ........ ................ * /, main/logger.c: Merged revisions 221920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 | tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines Initialize a variable that we check immediately upon startup. (closes issue #15973) Reported by: atis ........ 2009-10-02 01:35 +0000 [r221879] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: Merged revisions 221844 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines Merged revisions 221769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines Occasionally losing use of B channels in chan_misdn. I have not been able to reproduce the problem of losing channels. However, I have seen in the code a reentrancy problem that might give these symptoms. The reentrancy patch does several things: 1) Guards B channel and B channel structure allocation. 2) Makes the B channel structure find routines more precise in locating records. 3) Never leave a B channel allocated if we received cause 44. The last item may cause temporary outgoing call problems, but they should clear when the line becomes idle. (closes issue #15490) Reported by: slutec18 Patches: issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) Reported by: FabienToune Patches: issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: FabienToune, rmudgett, slutec18 ........ ................ 2009-10-02 00:07 +0000 [r221744-221780] Tilghman Lesher * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions 221777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines Fix a bunch of off-by-one errors ........ ................ * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 | tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers. ........ 2009-10-01 19:35 +0000 [r221698] David Vossel * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 | dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines outbound tls connections were not defaulting to port 5061 (closes issue #15854) Reported by: dvossel Patches: sip_port_config_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel ........ 2009-10-01 16:57 +0000 [r221660] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 221554,221589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu, 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE. ................ r221589 | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines Use unsigned ints for portinuri flags. ........ ................ 2009-10-01 16:25 +0000 [r221622] Kevin P. Fleming * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 221592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines Remove ability to control T.38 FAX error correction from udptl.conf. chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. ........ 2009-09-30 23:07 +0000 [r221477-221485] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 | mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2 lines Cleaned up merge from r221432 ........ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 221432 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines Merged revisions 221360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ ................ 2009-09-30 21:46 +0000 [r221371-221472] Matthias Nick * apps/app_queue.c, /: Merged revisions 221436 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 | mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines Prevents from division by zero ........ * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged revisions 221368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) | 23 lines Merged revisions 221153,221157,221303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines check bounds - prevents for buffer overflow ........ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines changed the prototype definition of csv_quote ........ ................ 2009-09-30 19:15 +0000 [r221304] Terry Wilson * configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: Merged revisions 221266 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ 2009-09-30 16:57 +0000 [r221204] Tilghman Lesher * main/channel.c, /: Merged revisions 221201 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009) | 14 lines Merged revisions 221200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines Avoid a potential NULL dereference. (closes issue #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt uploaded by tilghman (license 14) Tested by: kobaz ........ ................ 2009-09-30 14:57 +0000 [r221089] Sean Bright * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a() option. We require box numbers, not names as the documentation implies. (issue #14740) Reported by: pj Patches: __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10) Tested by: seanbright, lmadsen ........ 2009-09-30 04:41 +0000 [r221027-221047] Tilghman Lesher * /, funcs/func_lock.c: Recorded merge of revisions 221044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29 Sep 2009) | 8 lines Allow locks to be inherited through a masquerade without causing starvation. (closes issue #14859) Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: atis, tilghman ........ * include/asterisk/smdi.h, include/asterisk/optional_api.h (removed), apps/app_voicemail.c, include/asterisk/agi.h, include/asterisk/monitor.h: Remove optional_api from 1.6.2 branch, since it is not currently working. This is a blocking issue for the 1.6.2 release. (closes issue #15914) Reported by: mbeckwell Branch: http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162 Tested by: mbeckwell * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines Merged revisions 220873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines Reduce CPU usage related to building a peer merely for devicestates. This fixes a 100% CPU problem in the SIP driver, found by profiling the driver while the problem was occurring. (closes issue #14309) Reported by: pkempgen Patches: 20090924__issue14309.diff.txt uploaded by tilghman (license 14) Tested by: pkempgen, vrban ........ ................ 2009-09-29 20:24 +0000 [r220905-220934] Matthew Nicholson * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked. (closes issue #15965) Reported by: atis Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96) Tested by: atis * /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep 2009) | 5 lines Fix options 'm' and 's'. They were swapped in the code. Also document the fact that app_confbridge does not automatically answer the channel. (closes issue #15964) Reported by: shrift ........ 2009-09-29 17:06 +0000 [r220836] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines Make deletion of temporary greetings work properly with IMAP_STORAGE When imapgreetings was set to yes, the message was being deleted but wasn't actually being expunged. When imapgreetings was set to no, the file based message was not being deleted at all. All good now! (closes issue #14949) Reported by: noahisaac Patches: vm_tempgreeting_removal.patch uploaded by noahisaac (license 748), modified by me ........ 2009-09-28 19:13 +0000 [r220725] Sean Bright * /, Makefile.rules: Merged revisions 220721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep 2009) | 10 lines Merged revisions 220717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler so we override any default optimization levels for a particular install. ........ ................ 2009-09-28 19:11 +0000 [r220722] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 | jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines Fix building of registration entry in build_peer when using callbackextension Check for remotesecret option was unintentionally always true, which therefore caused the secret option to never be used. Thanks to dvossel for pointing out the exact fix. (closes issue #15943) Reported by: tpsast ........ 2009-09-27 20:45 +0000 [r220632] Michiel van Baak * funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009) | 3 lines add name argument for the CALLERID dialplan function to the xml documentation. Pointed out to me on IRC by snuff-home. Thanks ........ 2009-09-26 15:12 +0000 [r220589] Tilghman Lesher * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009) | 2 lines Allow AES to compile, when OpenSSL is not present. ........ 2009-09-24 20:38 +0000 [r220369] David Vossel * main/tcptls.c, /: Merged revisions 220365 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 | dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines fixes tcptls_session memory leak caused by ref count error (closes issue #15939) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/375/ ........ 2009-09-24 19:42 +0000 [r220292] Tilghman Lesher * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged revisions 220289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13 lines Merged revisions 220288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines Implicitly sending a progress signal breaks some applications. Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html ........ ................ 2009-09-24 18:22 +0000 [r220103-220221] Sean Bright * Makefile, /: Merged revisions 220217 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep 2009) | 9 lines Merged revisions 220213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep 2009) | 1 line Resolve parallel build warnings. Reported by Klaus Darilion on the asterisk-dev mailing list. ........ ................ * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400 (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep 2009) | 2 lines Remove the remaining bashisms in the Makefile/mkpkgconfig ........ ................ 2009-09-24 08:43 +0000 [r220031] Michiel van Baak * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200 (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009) | 7 lines mkpkgconfig does not need bash so make it use /bin/sh This fixes building on all systems that don't have bash at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on #asterisk-dev ........ ................ 2009-09-24 07:45 +0000 [r219989] Tilghman Lesher * apps/app_directory.c, /: Merged revisions 219987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009) | 8 lines Fix two possible crashes, one only in 1.6.1 and one in 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg Patches: 20090914__issue15739.diff.txt uploaded by tilghman (license 14) 20090922__issue15739.diff.txt uploaded by tilghman (license 14) Tested by: DLNoah, jeffg ........ 2009-09-22 21:48 +0000 [r219821] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines When IMAP variables were changed during a reload, Voicemail did not use the new values. This change introduces a configuration version variable, which ensures that connections with the old values are not reused but are allowed to expire normally. (closes issue #15934) Reported by: viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by tilghman (license 14) Tested by: viniciusfontes ........ ................ 2009-09-21 17:01 +0000 [r219722] David Vossel * channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500 (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 Sep 2009) | 3 lines Reverting merge 219520. This change was not necessary. ........ ................ 2009-09-20 18:21 +0000 [r219669] Tilghman Lesher * /, main/file.c: Merged revisions 219654 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009) | 15 lines Merged revisions 219653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines Really stop the stream, when ast_closestream() is called. (closes issue #15129) Reported by: bmh Patches: 20090918__issue15129.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/372/ ........ ................ 2009-09-19 03:14 +0000 [r219590] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219587 | russell | 2009-09-18 21:59:52 -0500 (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009) | 6 lines Make sure the iax_pvt exists before dereferencing it. This fixes the latest crash posted on issue 15609. (issue #15609) ........ ................ 2009-09-18 23:21 +0000 [r219452-219521] David Vossel * channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500 (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines iax2 frame double free The iax frame's retrans sched id was written over right before iax2_frame_free was called. In iax2_frame_free that retrans id is used to delete the sched item. By writing over the retrans field before the sched item could be deleted, it was possible for a retransmit to occur on a freed frame. ........ ................ * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) | 20 lines Merged revisions 219450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines via-header branches not updated correctly on INVITE INVITE requests must always contain a new unique branch id. When a new branch id is created for an INVITE, the dialog's invite_branch variable must be updated so CANCEL requests use the correct branch id. (closes issue #15262) Reported by: maniax Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608) invite_new_branch_trunk.diff uploaded by dvossel (license 671) Tested by: maniax, dvossel ........ ................ 2009-09-18 13:57 +0000 [r219415] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009) | 6 lines Missing value setting line for maxsecs/maxmessage (closes issue #15696) Reported by: fhackenberger Patches: maxsecs.patch uploaded by fhackenberger (license 592) ........ 2009-09-17 22:38 +0000 [r219376] David Vossel * /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 | dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines fixes deadlock when performing directed pickup w Invite/replaces (closes issue #15340) Reported by: lmsteffan Patches: deadlock.patch uploaded by lmsteffan (license 779) Tested by: lmsteffan ........ 2009-09-17 22:37 +0000 [r219370] Joshua Colp * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep 2009) | 12 lines Merged revisions 219320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep 2009) | 6 lines Send a 100 Trying response when we detect a spiral. This was problematic during spiral tests at SIPit... along with some other things as well. ........ ................ 2009-09-17 22:06 +0000 [r219307] David Vossel * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009) | 27 lines Merged revisions 219303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines INVITE w/Replaces deadlock fix This patch cleans up the locking logic in chan_sip.c's handle_invite_replaces() function as well as making use of ast_do_masquerade() rather than forcing the masquerade on an ast_read(). The code had several redundant unlocks that would result in 'freed more times than we've locked!' errors. I cleaned these up as well as moving all the unlock logic to the end of the function. This patch should also resolve the issue people were having with the replacecall channel never being unlocked with one legged calls. (closes issue #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff uploaded by dvossel (license 671) Tested by: irroot, dvossel Review: https://reviewboard.asterisk.org/r/371/ ........ ................ 2009-09-17 19:58 +0000 [r219267] Joshua Colp * /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 | file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines Ensure no spaces exist before "refresher=" when doing the comparison. ........ 2009-09-17 Leif Madsen * Released Asterisk 1.6.2.0-rc2 2009-09-17 15:38 +0000 [r219194] Matthew Nicholson * main/channel.c, /, include/asterisk/cdr.h, include/asterisk/channel.h: Merged revisions 219139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ ................ 2009-09-16 23:52 +0000 [r219063] Tilghman Lesher * main/config.c, configs/extensions.conf.sample, /: Merged revisions 219061 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009) | 15 lines Merged revisions 219023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ ................ 2009-09-16 19:40 +0000 [r218938] David Brooks * main/pbx.c, /: Merged revisions 218868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009) | 20 lines Merged revisions 218867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines Fixes CID pattern matching behavior to mirror that of extension pattern matching. Pattern matching for extensions uses a type of scoring system, giving values for specificity to each character in the pattern. Unfortunately, this is done character by character, in order. This does lead to some less specific patterns being first in line for matching, but it will usually get the job done. This patch merely brings CID matching to the same level as extension matching. This patch does not attempt to tackle the problem shared by extension matching. (closes issue #14708) Reported by: klaus3000 ........ ................ 2009-09-16 19:29 +0000 [r218937] Mark Michelson * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 | mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12 lines Reverse order of args to fread. This way, we don't always write a null byte into byte 1 of the buffer (closes issue #15905) Reported by: ebroad Patches: freadfix.patch uploaded by ebroad (license 878) Tested by: ebroad ........ 2009-09-16 19:25 +0000 [r218934] Joshua Colp * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 | file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On TCP and TLS connections do not attempt to stop retransmission of the packet internally. This was preventing responses from being properly processed because the packet was not being found causing handle_response to return prematurely. ........ 2009-09-16 13:38 +0000 [r218802] Russell Bryant * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged revisions 218799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009) | 16 lines Merged revisions 218798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines Remove the IAXy firmware from Asterisk. The firmware can now be found on downloads.digium.com, where the rest of our binary downloads live. This was the last part of our Asterisk tarballs that was considered non-free by Debian. :-) (closes issue #15838) Reported by: paravoid ........ ................ 2009-09-15 22:46 +0000 [r218733] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500 (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines If the user enters the same password as before, don't signal an error when the change does nothing. (closes issue #15492) Reported by: cbbs70a Patches: 20090713__issue15492.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-09-15 19:24 +0000 [r218688] David Vossel * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 | dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines upward bound checking for port string to int conversion ........ 2009-09-15 16:18 +0000 [r218590] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep 2009) | 15 lines Merged revisions 218578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep 2009) | 8 lines Send request contact header field with response to registrer queries instead of the address of record. (closes issue #14438) Reported by: ravindrad Patches: regquerypatch uploaded by ravindrad (license 684) Tested by: ravindrad ........ ................ 2009-09-15 16:06 +0000 [r218582] Tilghman Lesher * apps/app_followme.c, /: Merged revisions 218579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009) | 16 lines Merged revisions 218577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines Ensure FollowMe sets language in channels it creates. Also, not in the original bug report, but related fields are accountcode and musicclass, and the inheritance of datastores. (closes issue #15372) Reported by: Romik Patches: 20090828__issue15372.diff.txt uploaded by tilghman (license 14) Tested by: cervajs ........ ................ 2009-09-15 15:59 +0000 [r218576] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 218430 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor. After talking to rmudgett about some of his recent iflist locking changes, it was determined that the only place that would destroy a channel without being explicitly to do so was in handle_init_event. The loop to walk the interface list has been modified to wait to destroy the channel until the dahdi_pvt of the channel to be destroyed is no longer needed. (closes issue #15378) Reported by: samy ........ ................ 2009-09-15 15:42 +0000 [r218507-218575] Mark Michelson * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 | mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4 lines Use a better method of ensuring null-termination of the buffer while reading the SDP when using TCP. ........ * /, channels/chan_sip.c: Merged revisions 218499,218504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue, 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53 -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP socket is null-terminated. ........ 2009-09-15 15:05 +0000 [r218503] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 218500 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep 2009) | 9 lines Merged revisions 218497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep 2009) | 1 line Use proper hostname for downloading sound files. ........ ................ 2009-09-14 19:49 +0000 [r218364] Tilghman Lesher * sounds/Makefile, apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 218361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines Don't say "Please try again" if we don't give the user another chance to try again. (issue #15055, SWP-129) Reported by: jthurman ........ ................ 2009-09-14 18:18 +0000 [r218300] Joshua Colp * /, main/features.c: Merged revisions 218295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 | file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do not attempt to add a parking extension if an error occurred while reading the configuration. ........ 2009-09-14 15:20 +0000 [r218238] Matthew Nicholson * /, apps/app_directed_pickup.c: Merged revisions 218224 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500 (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines Ensure we don't pickup ourselves when doing pickup by exten. (closes issue #15100) Reported by: lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan (license 779) ........ ................ 2009-09-13 22:12 +0000 [r218219] Tzafrir Cohen * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0 that annoys gcc This memset doesn't write beyond the end of the buffer. (tmpbuf has size of 4). Merged revisions 218184 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-09-13 05:59 +0000 [r218151] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 218150 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1 line get rid of mfcr2 monitor thread condition, is problematic ........ 2009-09-11 06:00 +0000 [r217926-218055] Tilghman Lesher * main/pbx.c, /: Merged revisions 218050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 | tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines Check the origination priority for more matches, not the current priority. Found by Pavel Troller on the -dev list. ........ * apps/app_queue.c, /: Merged revisions 217990 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009) | 10 lines Merged revisions 217989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines Don't ring another channel, if there's not enough time for a queue member to answer. (Fixes AST-228) ........ ................ * channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /, channels/chan_sip.c: Merged revisions 217916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 | tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines Make calltoken support work with realtime users and peers. ........ 2009-09-10 21:21 +0000 [r217821] David Vossel * channels/chan_iax2.c, /: Merged revisions 217807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500 (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call Token security patch inadvertently broke the use of encryption due to the reorganization of code in the socket_process() function. When encryption is used, an incoming full frame must first be decrypted before the information elements can be parsed. The security release mistakenly moved IE parsing before decryption in order to process the new Call Token IE. To resolve this, decryption of full frames is once again done before looking into the frame. This involves searching for an existing callno, checking the pvt to see if encryption is turned on, and decrypting the packet before the internal fields of the full frame are accessed. (closes issue #15834) Reported by: karesmakro Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, karesmakro Review: https://reviewboard.asterisk.org/r/355/ ........ ................ 2009-09-10 19:56 +0000 [r217739] mnick : * res/res_musiconhold.c, /: Merged revisions 217730 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) | 17 lines Sets the correct musicclass after an announcement (closes issue #15279) Reported by: mbeckwell Patches: patch.txt uploaded by mnick (license ) Tested by: mnick (closes issue #15832) Reported by: mbeckwell Patches: patch.txt uploaded by mnick (license 874) Tested by: mnick ........ 2009-09-10 18:40 +0000 [r217665] Olle Johansson * /, channels/chan_sip.c: Merged revisions 216805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 | oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Since it's possible to have more than 999 calls, I'm changing the call counter roof to something higher. ........ 2009-09-10 18:19 +0000 [r217647] Tilghman Lesher * res/res_config_odbc.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 217638 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 | tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines Verify support for wide ODBC character types before using them. (closes issue #15870) Reported by: nic_bellamy ........ 2009-09-10 15:14 +0000 [r217632] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 217524 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1 line ast_log replaced for ast_verbose in MFCR2 event notifications ........ 2009-09-10 12:09 +0000 [r217594] Olle Johansson * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 | oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines Include ActionID in all events that are responsed to AMI Action SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy Patches: manager_SIPshowregistry_actionid.patch uploaded by nic bellamy (license 299) ........ 2009-09-09 20:37 +0000 [r217519] Tzafrir Cohen * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc 4.4 has more strict rules for aliasing. It doesn't like a struct sockaddr_in pointer pointing to a struct sockaddr. So we make it a union. Merged revisions 217445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk 2009-09-09 10:58 +0000 [r217369] Olle Johansson * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 | oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not having any TLS session to write to is a serious XMIT_ERROR. ........ 2009-09-08 22:20 +0000 [r217299] Sean Bright * /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 | seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4 lines Fix compilation of app_meetme. Reported by ebroad in #asterisk-bugs ........ 2009-09-08 20:33 +0000 [r217217] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009) | 14 lines Merged revisions 217156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines When MOH is playing on the channel, announcements sent through the conference are not heard. (closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman ........ ................ 2009-09-08 16:39 +0000 [r217077] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 217074 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 | kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9 lines Ensure that the default autoconf CFLAGS are not used. A recent change to the configure script that allows the user to specify CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That problem is now corrected. ........ 2009-09-08 15:36 +0000 [r217036] Tilghman Lesher * /, res/res_limit.c: Merged revisions 217033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 | tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines Remove what appears to be an unnecessary define. (closes issue #15851) Reported by: tzafrir ........ 2009-09-08 14:27 +0000 [r216994] David Vossel * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines caller id number empty parse_uri was not being given the correct scheme's, as a result, uri parsing did not parse the username correctly. One of the side effects of this is an empty caller id. (closes issue #15839) Reported by: ebroad Patches: blank_cidv2.patch uploaded by ebroad (license 878) parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: ebroad, dvossel ........ 2009-09-07 16:43 +0000 [r216647-216845] Olle Johansson * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 | oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Make sure we reset global_exclude_static at channel reload ........ * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 | oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If there is no session timer in the INVITE, set it to default value (not unset minimum = -1) Patch by oej closes issue #15621 Reported by: fnordian Tested by: atis ........ * CHANGES, UPGRADE.txt: Add docs * configs/sip.conf.sample, apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c, apps/app_disa.c: Merged revisions 216438 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ 2009-09-04 19:42 +0000 [r216598] David Vossel * /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 | dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines sip peer matching by address only with TCP/TLS This patch removes the contact header matching logic and adds logic to match all tcp/tls connections by ip only Review: https://reviewboard.asterisk.org/r/354/ ........ 2009-09-04 19:32 +0000 [r216597] Sean Bright * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep 2009) | 1 line Use ast_free() instead of free(). ........ 2009-09-04 17:53 +0000 [r216550-216553] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009) | 2 lines Fix trunk breakage. ........ * UPGRADE-1.6.txt, main/pbx.c, /: Merged revisions 216547 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04 Sep 2009) | 3 lines Enable turning off the application delimiter warning with the 'dontwarn' option. Suggested on the -dev list, and implemented in an alternate way by me. ........ 2009-09-04 15:11 +0000 [r216469-216509] Michiel van Baak * /, main/utils.c: Merged revisions 216506 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009) | 9 lines Merged revisions 216435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD ........ ................ * /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009) | 2 lines make sure canlog is set so we can compile with DEBUG_THREADS enabled on OpenBSD ........ 2009-09-04 13:57 +0000 [r216267-216436] Russell Bryant * /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 | russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines Do not treat every SIP peer as if they were configured with insecure=port. There was a problem in the function responsible for doing peer matching by IP address and port number such that during the second pass for checking for a peer configured with insecure=port, it would end up treating every peer as if it had been configured that way. These changes fix the logic in the peer IP and port comparison callback to handle insecure=port checking properly. This problem was introduced when SIP peers were converted to astobj2. Many thanks to dvossel for noticing this while working on another peer matching issue. ........ * doc/IAX2-security.txt (added), /: Merged revisions 216264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216264 | russell | 2009-09-04 05:48:44 -0500 (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216263 | russell | 2009-09-04 05:48:00 -0500 (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines Add a plain text version of the IAX2 security document. ........ ................ ................ 2009-09-04 06:14 +0000 [r216225] Michiel van Baak * main/astobj2.c, /: Merged revisions 216222 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 | mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines make sure 'start' is always initialized. Makes asterisk compile with --enable-dev-mode ........ 2009-09-03 19:44 +0000 [r216014-216099] Russell Bryant * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009) | 16 lines Merged revisions 216085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216085 | russell | 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt. ........ ................ ................ * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216009 | russell | 2009-09-03 13:45:54 -0500 (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216008 | russell | 2009-09-03 13:44:58 -0500 (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines Add IAX2 security document related to AST-2009-006. ........ ................ ................ 2009-09-03 18:42 +0000 [r216007] David Vossel * channels/chan_iax2.c, channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample, include/asterisk/acl.h, channels/iax2-parser.h, /, include/asterisk/astobj2.h, channels/iax2.h, main/acl.c: Merged revisions 215955 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ 2009-09-03 14:21 +0000 [r215887-215929] Olle Johansson * /, channels/chan_sip.c: Merged revisions 215891 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 | oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add known internal IP address when autodomain=yes (closes issue #14573) Reported by: pj Patches: sip-internip-autodomain1.diff uploaded by mnicholson (license 96) modified by oej Tested by: pj ........ * main/rtp.c, channels/chan_sip.c: Fix bad reports in "sip show channelstats". Not directly mergeable in svn trunk, needs more tests, therefore committed directly to 1.6.2. (closes issue #15819) Reported by: klaus3000 Patches: asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded by klaus3000 (license 65) Tested by: klaus3000, oej 2009-09-03 06:02 +0000 [r215841] Michiel van Baak * doc/manager_1_1.txt, /: Merged revisions 215838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215838 | mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines Document that SIPshowpeer and SKINNYshowline now include the configured parkinglot in their response. Prodded by snuff-work on #asterisk-dev IRC channel ........ 2009-09-03 03:44 +0000 [r215802] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 215801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 | tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines Default the callback extension to "s". This is a regression. (closes issue #15764) Reported by: elguero Change-type: bugfix ........ 2009-09-03 00:34 +0000 [r215795] Terry Wilson * /, channels/chan_sip.c: Merged revisions 215758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009) | 25 lines Merged revisions 215682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines Re-send non-100 provisional responses to prevent cancellation From section 13.3.1.1 of RFC 3261: If the UAS desires an extended period of time to answer the INVITE, it will need to ask for an "extension" in order to prevent proxies from canceling the transaction. A proxy has the option of canceling a transaction when there is a gap of 3 minutes between responses in a transaction. To prevent cancellation, the UAS MUST send a non-100 provisional response at every minute, to handle the possibility of lost provisional responses. (closes issue #11157) Reported by: rjain Tested by: twilson Review: https://reviewboard.asterisk.org/r/315/ ........ ................ 2009-09-02 21:46 +0000 [r215683] David Vossel * /, channels/chan_sip.c: Merged revisions 215681 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 | dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines port string to int conversion using sscanf There are several instances where a port is parsed from a uri or some other source and converted to an int value using atoi(), if for some reason the port string is empty, then a standard port is used. This logic is used over and over, so I created a function to handle it in a safer way using sscanf(). ........ 2009-09-02 21:37 +0000 [r215647-215680] Michiel van Baak * /, channels/chan_sip.c, channels/chan_skinny.c: Merged revisions 215665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 | mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines add Parkinglot info to sip show peer and skinny show line If we had this from the start, debugging the 'parking not using configured parkinglot' bug would have been easier. ........ * /, main/features.c: Merged revisions 215622 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215622 | mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines - lock channel before looking for a channel variable - Init the parkings list member of struct parkinglot. Thanks Sean for the explanation why this should be here. ........ 2009-09-02 18:52 +0000 [r215569-215570] Tilghman Lesher * /, main/Makefile, main/app.c: Merged revisions 215567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215567 | tilghman | 2009-09-02 13:37:25 -0500 (Wed, 02 Sep 2009) | 9 lines Close up to the soft open file limit (same on Linux, but varies drastically on OS X). Also, a Makefile fix for Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches: 20090901__issue14542.diff.txt uploaded by tilghman (license 14) Tested by: jtodd, tilghman Change-type: bugfix ........ * /, channels/chan_sip.c: Merged revisions 215222 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 | tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines Fix register such that lines with a transport string, but without an authuser, parse correctly. (AST-228) ........ 2009-09-02 17:35 +0000 [r215523] David Vossel * /, channels/chan_sip.c: Merged revisions 215522 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 | dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines SIP uri parsing cleanup Now, the scheme passed to parse_uri can either be a single scheme, or a list of schemes ',' delimited. This gets rid of the whole problem of having to create two buffers and calling parse_uri twice to check for separate schemes. Review: https://reviewboard.asterisk.org/r/343/ ........ 2009-09-02 16:35 +0000 [r215512] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 215479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009) | 3 lines like in chan_sip's sip_new skinny should copy the configured parkinglot from a line to the newly created channel. This makes callparking honor the configured parkinglot for skinny lines as well. ........ 2009-09-02 16:09 +0000 [r215467] David Vossel * /, channels/chan_sip.c: Merged revisions 215466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 | dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines SIP support for keep-alive event keep-alive events are used by Sipura/Linksys for NAT keepalive. There currently don't appear to be any problems with NAT, but everytime a keep-alive event is received, Asterisk responds with a "489 Bad event". This error may indicate to a user that NAT problems exist just because this even is not supported. Now, rather than respond with an error, the packet is consumed and a "200 ok" is sent just to indicate we received the packet. (issue #15084) Patches: chan_sip.keepalive.v1.diff uploaded by IgorG (license 20) ........ 2009-09-02 16:07 +0000 [r215465] Michiel van Baak * /, channels/chan_sip.c: Merged revisions 215462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 | mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12 lines Honor configured parkinglot when parking and retrieving parked calls Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer into the newly created channel. (closes issue #15538) Reported by: gracedman Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7) With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call) Tested by: gracedman, mvanbaak ........ 2009-09-02 01:49 +0000 [r215376] Dwayne M. Hubbard * /, apps/app_softhangup.c: Merged revisions 215338 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500 (Tue, 01 Sep 2009) | 18 lines Merged revisions 215270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) | 12 lines Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names In general channel names are in the form Foo/Bar-Z, but the channel name could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the channel name at the last hyphen. (closes issue #15810) Reported by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard (license 733) ........ ................ 2009-09-01 20:00 +0000 [r215165] Kevin P. Fleming * main/frame.c, /: Merged revisions 215161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215161 | kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3 lines Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS frames are properly decoded. ........ 2009-08-31 16:22 +0000 [r214822-214960] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 214945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500 (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009) | 7 lines Also unlock the "other" channel, when returning, due to glare. (closes issue #15787) Reported by: tim_ringenbach Patches: chan_local.diff uploaded by tim ringenbach (license 540) Tested by: tim_ringenbach ........ ................ * Makefile, /: Merged revisions 214898 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214898 | tilghman | 2009-08-30 17:10:35 -0500 (Sun, 30 Aug 2009) | 2 lines Force Darwin on ppc platforms to compile with a target level that supports aliasing. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c: Merged revisions 214819 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 | tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines If lua is detected with the lua5.1 prefix (or not), adjust the include path accordingly. Based upon feedback to a release announcement on the -users list. See http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html ........ 2009-08-29 Leif Madsen * Asterisk 1.6.2.0-rc1 released. 2009-08-28 20:17 +0000 [r214707] Tilghman Lesher * main/channel.c, /: Merged revisions 214702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009) | 15 lines Merged revisions 214701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) | 8 lines Modify comment to be a bit more accurate. We have kept this comment around long enough, that it's pretty clear that we're keeping the code, because changing the code would require a pretty fundamental architectural shift. We've also taken criticism in some quarters, because it was believed that it was referring to the code being nasty. No, the code isn't nasty, just the operation itself is rather odd. Fixed for eternity (probably not). ........ ................ 2009-08-28 20:05 +0000 [r214700] Kevin P. Fleming * makeopts.in, Makefile, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 214696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 | kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9 lines Ensure that CFLAGS and/or LDFLAGS provided to configure script are preserved. Cross-compilation environments want to provide 'defaults' for compiler and linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the environment or as command-line arguments to the configure script. This patch modifies the configure script and Makefile to preserve these settings and ensure they are used in the build process. ........ 2009-08-28 18:43 +0000 [r214653] Mark Michelson * /, include/asterisk/sched.h: Merged revisions 214650 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug 2009) | 3 lines Fix some incorrect documentation of sched_thread functions. ........ 2009-08-27 21:49 +0000 [r214202-214521] Tilghman Lesher * autoconf/libcurl.m4 (added), /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 214518 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009) | 14 lines Merged revisions 214517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009) | 7 lines Use autoconf to detect libcurl, as this enables cross-compilation checks, something we didn't allow before. (closes issue #15714) Reported by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded by tilghman (license 14) Tested by: pprindeville ........ ................ * main/manager.c, /: Merged revisions 214514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214514 | tilghman | 2009-08-27 16:26:37 -0500 (Thu, 27 Aug 2009) | 7 lines Ensure that we check for the special value CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by: a_villacis Patches: asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch uploaded by a villacis (license 660) (Plus a few of my own, to catch the remaining places within manager.c where it could have been a problem) ........ * autoconf/ast_ext_lib.m4, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 214466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009) | 9 lines Merged revisions 214436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009) | 2 lines One more build system change, to make the descriptions look better, if we have better information. ........ ................ * autoconf/ast_ext_lib.m4, /, configure, include/asterisk/autoconfig.h.in: Merged revisions 214360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500 (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009) | 3 lines Make autoheader descriptions render correctly in our autoconfig.h file. (Figured out while working with issue #14906) ........ ................ * /, channels/chan_sip.c: Merged revisions 214199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 | tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue #15362) Reported by: klaus3000 Patches: chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license 65) ........ 2009-08-26 16:39 +0000 [r214196] David Vossel * main/channel.c, /: Merged revisions 214195 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009) | 25 lines Merged revisions 214194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) | 19 lines ast_write() ignores ast_audiohook_write() results In ast_write(), if a channel has a list of audiohooks, those lists are written to and the resulting frame is what ast_write() should continue with. The problem was the returned audiohook frame was not being handled at all, and the original frame passed into it did not contain the mixed audio, so essentially audio was being lost. One result of this was chan_spy's whisper mode no longer worked. To complicate the issue, frames passed into ast_write may either be a single frame, or a list of frames. So, as the list of frames is processed in the audiohook_write, the returned frames had to be added to a new list. (closes issue #15660) Reported by: corruptor Tested by: dvossel ........ ................ 2009-08-25 22:43 +0000 [r213903-214155] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 214152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 | tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines Not all versions of gnu-linux use glibc, which contains iconv. Some (especially embedded systems) don't have iconv at all. (closes issue #15169) Reported by: pprindeville ........ * /, main/say.c: Merged revisions 214071 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009) | 17 lines Merged revisions 214068-214069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009) | 6 lines Fix pronunciation of German dates. (closes issue #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded by Benjamin Kluck (license 803) ........ r214069 | tilghman | 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should always compile before committing... ........ ................ * /, pbx/pbx_dundi.c: Merged revisions 213975 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213975 | tilghman | 2009-08-25 01:51:12 -0500 (Tue, 25 Aug 2009) | 6 lines DUNDILOOKUP function in 1.6 should use comma delimiters. (closes issue #15322) Reported by: chappell Patches: dundilookup-0015322.patch uploaded by chappell (license 8) ........ * main/pbx.c, /: Merged revisions 213971 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009) | 14 lines Merged revisions 213970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009) | 7 lines Improve error message by informing user exactly which function is missing a parethesis. (closes issue #15242) Reported by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by loloski (license 68) ........ ................ * Makefile, /: Merged revisions 213904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213904 | tilghman | 2009-08-24 21:54:07 -0500 (Mon, 24 Aug 2009) | 6 lines The DTD should be installed in the same path as the rest of the XML documentation. (closes issue #15344) Reported by: tzafrir Patches: makefile_appdocs_dtd.diff uploaded by tzafrir (license 46) ........ * Makefile, /: Merged revisions 213900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213900 | tilghman | 2009-08-24 21:41:17 -0500 (Mon, 24 Aug 2009) | 11 lines Merged revisions 213899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009) | 4 lines Use the default runlevels for Debian derivatives, instead of making up our own. (closes issue #14730) Reported by: pkempgen ........ ................ 2009-08-24 16:49 +0000 [r213836] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 213833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009) | 14 lines Fix storage of greetings when using IMAP_STORAGE The store macro was not getting called preventing storage of IMAP greetings at all. This has been corrected along with fixing checking if the imapgreetings option is turned on to store the greeting in IMAP. Lastly, the attachment filename was incorrectly using the full path instead of just the basename, which was causing problems with retrieval of the greeting. (closes issue #14950) Reported by: noahisaac (closes issue #15729) Reported by: lmadsen ........ 2009-08-24 04:54 +0000 [r213791] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 213790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213790 | moy | 2009-08-24 00:46:28 -0400 (Mon, 24 Aug 2009) | 1 line improve handling of openr2_chan_disconnect_call API failure, unlikely, but happened on openr2 library bug ........ 2009-08-21 22:54 +0000 [r213739] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 213738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 | tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines Clarifying comments in sip_register, and removing a dead section ........ 2009-08-21 22:23 +0000 [r213721] David Vossel * /, channels/chan_sip.c: Merged revisions 213716 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 | dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines Register request line contains wrong address when user domain and register host differ (closes issue #15539) Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ 2009-08-21 21:44 +0000 [r213698] Kevin P. Fleming * apps/app_voicemail.c, /: Merged revisions 213697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug 2009) | 12 lines Ensure that realtime mailboxes properly report status on subscription. This patch modifies app_voicemail's response to mailbox status subscriptions (via the internal event system) to ensure that a subscription triggers an explicit poll of the mailbox, so the subscriber can get an immediate cached event with that status. Previously, the cache was only populated with the status of non-realtime mailboxes. (closes issue #15717) Reported by: natmlt ........ 2009-08-21 21:12 +0000 [r213636] David Vossel * /, channels/chan_sip.c: Merged revisions 213635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 | dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines fixes sip register parsing when user@domain is used (issue #15008) (issue #15672) ........ 2009-08-21 16:55 +0000 [r213563] Tilghman Lesher * include/asterisk.h, /: Merged revisions 213560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009) | 14 lines Merged revisions 213559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009) | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files. (closes issue #15698) Reported by: slavon Patches: 20090817__issue15698.diff.txt uploaded by tilghman (license 14) Tested by: slavon, tilghman ........ ................ 2009-08-21 16:06 +0000 [r213497] Jason Parker * /, configs/queues.conf.sample: Merged revisions 213494 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213494 | qwell | 2009-08-21 11:04:21 -0500 (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) | 5 lines Clarify queues.conf comments to specify that variables should be set in the dialplan. (closes issue #15755) Reported by: trendboy ........ ................ 2009-08-21 04:25 +0000 [r213475-213481] Moises Silva * channels/chan_dahdi.c, /: Merged revisions 213454 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213454 | moy | 2009-08-21 00:09:26 -0400 (Fri, 21 Aug 2009) | 1 line increment the mfcr2 monitor count when clearing the call request ........ * channels/chan_dahdi.c, /: Merged revisions 213216 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213216 | moy | 2009-08-19 23:26:59 -0400 (Wed, 19 Aug 2009) | 1 line fixed bug caused by calling ast_request without calling ast_call on an R2 channel, ie, CHANISAVAIL ........ 2009-08-21 03:53 +0000 [r213453] Terry Wilson * main/loader.c, /: Merged revisions 213450 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 | twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines Make LOAD_ORDER actually work ........ 2009-08-20 21:50 +0000 [r213413] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 213404 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009) | 12 lines Fix greeting retrieval from IMAP Properly check for the current voicemail state and if it doesn't exist, create it. (closes issue #14597) Reported by: wtca Patches: 14597_v2.patch uploaded by mmichelson (license 60) Tested by: jpeeler ........ 2009-08-20 20:37 +0000 [r213350] Matthew Nicholson * /, main/features.c: Merged revisions 213327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 | mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7 lines Fix a crash by checking the proper pointer for validity before deferencing it. (closes issue #15751) Reported by: atis Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license 242) ........ 2009-08-19 22:41 +0000 [r213182] Jason Parker * main/alaw.c, main/ulaw.c, /: Merged revisions 213179 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) | 5 lines Fix compile when certain G711 menuselect options are enabled. (closes issue #15697) Reported by: slavon ........ 2009-08-19 21:25 +0000 [r213128] David Vossel * apps/app_mixmonitor.c, /: Merged revisions 213113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500 (Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009) | 8 lines Fixes memory leak caused by incorrectly freeing mixmonitor (closes issue #15699) Reported by: edantie Patches: mixmonitor.patch uploaded by edantie (license 862) ........ ................ 2009-08-19 21:22 +0000 [r213095-213117] Tilghman Lesher * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions 213098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 | tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines Better parsing for the "register" line Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman ........ * /, channels/chan_sip.c: Merged revisions 213093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 | tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines If we have realtime caching enabled, 'sip reload' must purge users/peers, even if the config files haven't changed. (closes issue #12869) Reported by: bcnit Patches: 20090819__issue12869__2.diff.txt uploaded by tilghman (license 14) Tested by: lasko ........ 2009-08-19 15:35 +0000 [r213047] Russell Bryant * /, main/features.c: Merged revisions 213046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r213046 | russell | 2009-08-19 10:32:18 -0500 (Wed, 19 Aug 2009) | 4 lines Don't blow up on a NULL cdr. Reported in #asterisk-dev. ........ 2009-08-18 20:34 +0000 [r212931-212944] Kevin P. Fleming * /: Merged revisions 212939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212939 | kpfleming | 2009-08-18 15:33:34 -0500 (Tue, 18 Aug 2009) | 1 line Remove some accidentally-committed properties. ........ * sounds/Makefile, doc/tex/asterisk.tex, CREDITS, /, UPGRADE-1.4.txt, sounds/sounds.xml, build_tools/prep_tarball: Merged revisions 212922 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212922 | kpfleming | 2009-08-18 15:29:37 -0500 (Tue, 18 Aug 2009) | 6 lines Convert this branch to Opsound music-on-hold. For more details: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ ........ 2009-08-18 19:28 +0000 [r212866] Tilghman Lesher * /, configs/extconfig.conf.sample: Merged revisions 212857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18 Aug 2009) | 4 lines Make the default extconfig.conf match entries with the sample res_mysql.conf. This eliminates a future source of possible confusion with the configuration of 1.6.1 and higher. ........ 2009-08-18 16:56 +0000 [r212769] Richard Mudgett * channels/misdn/isdn_lib.c, /: Merged revisions 212758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500 (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009) | 1 line Removed some deadwood and added some doxygen comments. ........ ................ 2009-08-18 16:41 +0000 [r212767] Sean Bright * main/manager.c, /: Merged revisions 212764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r212764 | seanbright | 2009-08-18 12:38:36 -0400 (Tue, 18 Aug 2009) | 18 lines Merged revisions 212763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug 2009) | 11 lines Delay the creation of temporary files until we have a valid manager command to handle. Without this patch, asterisk creates a temporary file before determining if the specified command is valid. If invalid, we weren't properly cleaning up the file. (closes issue #15730) Reported by: zmehmood Patches: M15730.diff uploaded by junky (license 177) Tested by: zmehmood ........ ................ 2009-08-17 20:01 +0000 [r212631] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 212627 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009) | 4 lines Check the return value of opendir(3), or we may crash. (closes issue #15720) Reported by: tobias_e ........ 2009-08-17 18:56 +0000 [r212580-212584] Sean Bright * /, channels/chan_agent.c: Merged revisions 212581 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug 2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in chan_agent. (closes issue #15668) Reported by: davidw ........ * main/logger.c: Merged revisions 212574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212574 | seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8 lines Correct the return value check for ast_safe_system. The logic here was reversed as ast_safe_system returns -1 on error and not on success. Fix suggested by reporter. (closes issue #15667) Reported by: loic ........ 2009-08-17 16:52 +0000 [r212509] Jeff Peeler * channels/misdn_config.c, /: Merged revisions 212506 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500 (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If more ports were specified than configured in misdn.conf a reload would crash asterisk. The problem was the unconfigured port was using data from the previously configured port. When the data for an unconfigured port was freed a crash would result from the double free. (closes issue #12113) Reported by: agupta Patches: bug12113.patch uploaded by jpeeler (license 325) ........ ................ 2009-08-17 15:51 +0000 [r212434] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 212431 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500 (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix uninitialized variable causing random MWI indications. (closes issue #15727) Reported by: doda Patches: dahdi_changes.patch uploaded by doda (license 853) ........ r212430 | rmudgett | 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix uninitialized variable. ........ ................ 2009-08-14 17:37 +0000 [r212250] Tilghman Lesher * funcs/func_curl.c, /: Merged revisions 212249 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212249 | tilghman | 2009-08-14 12:36:40 -0500 (Fri, 14 Aug 2009) | 2 lines Add SSL_VERIFYPEER, as requested on the -users list ........ 2009-08-13 15:47 +0000 [r212116] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 212113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 | kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3 lines Ensure that T38FaxVersion is put into outgoing SDP in the proper case. ........ 2009-08-13 13:56 +0000 [r212070] Joshua Colp * /, channels/chan_sip.c: Merged revisions 212067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 | file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines Check an actual populated variable when seeing if we need to do video or not. ........ 2009-08-13 11:47 +0000 [r212030] Gavin Henry * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 212027 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r212027 | ghenry | 2009-08-13 12:37:12 +0100 (Thu, 13 Aug 2009) | 6 lines Fixed typo (closes issue #15710) Reported by: suretec ........ 2009-08-12 23:16 +0000 [r211951-211959] Matthew Nicholson * apps/app_queue.c, /: Merged revisions 211957 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug 2009) | 17 lines Merged revisions 211953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug 2009) | 10 lines This patch adds additional checking when generating queue log TRANSFER events. The additional checks prevent generation of false TRANSFER events in certain situations. (closes issue #14536) Reported by: aragon Patches: queue-log-xfer-fix1.diff uploaded by mnicholson (license 96) Tested by: aragon, mnicholson ........ ................ * /, channels/chan_sip.c: Merged revisions 211876 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 | mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11 lines Make asterisk handle 423 Interval Too Short messages better. This change uses separate values for the acceptable minimum expiry provided by the 423 error and the expiry value stored in the configuration file. Previously, the value pulled from the configuration file would be overwritten. (closes issue #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch uploaded by Nick (license 657) Tested by: mnicholson ........ 2009-08-12 16:21 +0000 [r211785] Gavin Henry * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 211767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211767 | ghenry | 2009-08-12 17:00:46 +0100 (Wed, 12 Aug 2009) | 33 lines Added three new attributes and applied a patch to res_config_ldap.c attributetype ( AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent' DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) and patch fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725) Reported by: macogeek Patches: fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license 863) Tested by: suretec ........ 2009-08-10 19:51 +0000 [r211580-211585] Tilghman Lesher * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500 (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10 Aug 2009) | 1 line Conversion specifiers, not format specifiers ........ ................ * apps/app_queue.c, apps/app_talkdetect.c, agi/eagi-sphinx-test.c, res/res_config_curl.c, channels/chan_usbradio.c, channels/chan_misdn.c, res/snmp/agent.c, apps/app_sms.c, apps/app_verbose.c, apps/app_stack.c, apps/app_mixmonitor.c, main/asterisk.c, main/dsp.c, main/timing.c, doc/CODING-GUIDELINES, funcs/func_speex.c, main/frame.c, utils/muted.c, apps/app_meetme.c, apps/app_alarmreceiver.c, cdr/cdr_pgsql.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c, main/enum.c, main/indications.c, res/res_config_sqlite.c, channels/misdn_config.c, utils/frame.c, main/cli.c, pbx/pbx_loopback.c, channels/chan_phone.c, funcs/func_enum.c, res/res_smdi.c, channels/chan_skinny.c, funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c, res/res_config_ldap.c, apps/app_adsiprog.c, funcs/func_dialplan.c, main/pbx.c, main/dnsmgr.c, funcs/func_sprintf.c, funcs/func_timeout.c, channels/chan_sip.c, apps/app_privacy.c, res/res_limit.c, apps/app_waitforsilence.c, codecs/codec_speex.c, agi/eagi-test.c, apps/app_morsecode.c, funcs/func_cut.c, channels/chan_oss.c, main/netsock.c, apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c, pbx/pbx_dundi.c, utils/extconf.c, pbx/pbx_config.c, apps/app_chanspy.c, res/res_odbc.c, apps/app_voicemail.c, apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, /, apps/app_record.c, main/utils.c, cdr/cdr_adaptive_odbc.c, res/res_http_post.c, main/config.c, res/ael/pval.c, main/cdr.c, main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c, main/manager.c, apps/app_setcallerid.c, apps/app_osplookup.c, main/features.c, main/http.c, channels/xpmr/xpmr.c, apps/app_rpt.c, channels/chan_mgcp.c, res/res_config_pgsql.c, channels/chan_agent.c, funcs/func_math.c, apps/app_waituntil.c, apps/app_disa.c, main/acl.c, apps/app_originate.c, channels/iax2-provision.c: AST-2009-005 2009-08-10 14:15 +0000 [r211350] Joshua Colp * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 | file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix retrieval of the port used for the video stream when adding SDP to a SIP message. (closes issue #15121) Reported by: jsmith ........ 2009-08-09 15:43 +0000 [r211235-211278] Tilghman Lesher * /, main/astfd.c: Merged revisions 211275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009) | 9 lines Merged revisions 211274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009) | 2 lines Small oops. Clear the flags which have been checked. ........ ................ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 | tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines Check for NULL frame, before dereferencing pointer. (closes issue #15617) Reported by: rain ........ 2009-08-07 20:18 +0000 [r211122] Russell Bryant * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009) | 11 lines Recorded merge of revisions 211112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009) | 4 lines Resolve a deadlock involving app_chanspy and masquerades. (ABE-1936) ........ ................ 2009-08-07 18:20 +0000 [r211051] Tilghman Lesher * apps/app_queue.c, /: Merged revisions 211040 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009) | 21 lines Merged revisions 211038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername. This is a partial revert of revision 82590, which was an attempted cleanup, but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended as a method by which component interfaces could be queried from the queue. Membername isn't useful here, because that field cannot be used to obtain further information about the member. See the documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various AMI commands which take a member argument for further justification. (closes issue #15664) Reported by: rain Patches: app_queue-queue_member_list.diff uploaded by rain (license 327) ........ ................ 2009-08-07 13:10 +0000 [r210995] Kevin P. Fleming * main/udptl.c, /: Merged revisions 210992 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 | kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13 lines Workaround broken T.38 endpoints that offer tiny MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as the maximum IFP size that should be sent to them, rather than the maximum packet payload size. If such an endpoint also requests UDPRedundancy as the error correction mode, we'll end up calculating a tiny maximum IFP size, so small as to be unusable. This patch sets a lower bound on what we'll consider the remote's maximum IFP size to be, assuming that endpoints that do this really can accept larger packets than they've offered to accept. (closes issue #15649) Reported by: dazza76 ........ 2009-08-06 21:47 +0000 [r210911-210917] Tilghman Lesher * main/channel.c, /: Merged revisions 210914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009) | 14 lines Merged revisions 210913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009) | 7 lines Because channel information can be accessed outside of the channel thread, we must lock the channel prior to modifying it. (closes issue #15397) Reported by: caspy Patches: 20090714__issue15397.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged revisions 210908 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines Allow Gosub to recognize quote delimiters without consuming them. (closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ ........ 2009-08-06 17:49 +0000 [r210820] Joshua Colp * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 | file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines Accept additional T.38 reinvites after an initial one has been handled. Discussion of this subject has yielded that it is not actually acceptable to change T.38 parameters after the initial reinvite but declining is harsh and can cause the fax to fail when it may be possible to allow it to continue. This patch changes things so that additional T.38 reinvites are accepted but parameter changes ignored. This gives the fax a fighting chance. (closes issue #15610) Reported by: huangtx2009 ........ 2009-08-05 20:43 +0000 [r210686] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500 (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) | 14 lines Dialplan starts execution before the channel setup is complete. * Issue 15655: For the case where dialing is complete for an incoming call, dahdi_new() was asked to start the PBX and then the code set more channel variables. If the dialplan hungup before these channel variables got set, asterisk would likely crash. * Fixed potential for overlap incoming call to erroneously set channel variables as global dialplan variables if the ast_channel structure failed to get allocated. * Added missing set of CALLINGSUBADDR in the dialing is complete case. (closes issue #15655) Reported by: alecdavis ........ ................ 2009-08-05 18:56 +0000 [r210565-210566] Leif Madsen * /: Merged revisions 210564 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) | 11 lines Update imapstorage.txt documentation. Updated the imapstorage.txt documentation to reflect that issues with c-client versions older than 2007 seem to cause crashing issues that are not seen with more recent versions. Documentation has been updated to reflect this. (closes issue #14496) Reported by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, dbrooks ........ ................ * doc/tex/imapstorage.tex: Merged revisions 210564 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) | 11 lines Update imapstorage.txt documentation. Updated the imapstorage.txt documentation to reflect that issues with c-client versions older than 2007 seem to cause crashing issues that are not seen with more recent versions. Documentation has been updated to reflect this. (closes issue #14496) Reported by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, dbrooks ........ ................ 2009-08-04 14:55 +0000 [r210191-210241] Kevin P. Fleming * Makefile, /: Merged revisions 210238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug 2009) | 16 lines Merged revisions 210237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug 2009) | 10 lines Eliminate spurious compiler warnings from system headers on *BSD platforms. Ensure that system headers located in /usr/local/include are actually treated as system headers by the compiler, and not as local headers which are subject to warnings from the -Wundef compiler option and others. (closes issue #15606) Reported by: mvanbaak ........ ................ * configs/sip.conf.sample, configs/skinny.conf.sample, main/rtp.c, channels/chan_mgcp.c, doc/chan_sip-perf-testing.txt, contrib/scripts/realtime_pgsql.sql, /, channels/chan_sip.c, channels/chan_skinny.c, configs/mgcp.conf.sample, doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt, configs/res_ldap.conf.sample: Merged revisions 210190 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines Rename 'canreinvite' option to 'directmedia', with backwards compatibility. It is clear from multiple mailing list, forum, wiki and other sorts of posts that users don't really understand the effects that the 'canreinvite' config option actually has, and that in some cases they think that setting it to 'no' will actually cause various other features (T.38, MOH, etc.) to not work properly, when in fact this is not the case. This patch changes the proper name of the option to what it should have been from the beginning ('directmedia'), but preserves backwards compatibility for existing configurations. ........ 2009-08-01 11:33 +0000 [r209837-209906] Russell Bryant * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209887 | russell | 2009-08-01 06:29:25 -0500 (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009) | 5 lines Resolve a valgrind warning about a read from uninitialized memory. (issue #15396) Reported by: aragon ........ ................ * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209839 | russell | 2009-08-01 06:02:07 -0500 (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) | 13 lines Modify how Playtones() is used in Milliwatt() to resolve gain issue. When Milliwatt() was changed internally to use Playtones() so that the proper tone was used, it introduced a drop in gain in the output signal. So, use the playtones API directly and specify a volume argument such that the output matches the gain of the original Milliwatt() code. (closes issue #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff uploaded by russell (license 2) Tested by: rue_mohr ........ ................ * /, main/event.c: Merged revisions 209835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 | russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines Fix ast_event_queue_and_cache() to actually do the cache() part. (closes issue #15624) Reported by: ffossard Tested by: russell ........ 2009-08-01 01:34 +0000 [r209816] Kevin P. Fleming * pbx/pbx_config.c, channels/misdn/isdn_lib.c, utils/frame.c, main/pbx.c, /, main/Makefile, channels/misdn/ie.c: Merged revisions 209760-209761 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines Minor changes inspired by testing with latest GCC. The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. ........ ................ r209761 | kpfleming | 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert accidental Makefile change. ................ 2009-07-31 22:01 +0000 [r209715] Russell Bryant * /, main/event.c: Merged revisions 209711 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 | russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines Fix some places where ast_event_type was used instead of ast_event_ie_type. ........ 2009-07-30 18:51 +0000 [r209594] David Brooks * channels/chan_console.c, include/asterisk/abstract_jb.h, apps/app_forkcdr.c, channels/chan_dahdi.c, contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, codecs/lpc10/pitsyn.c: Merged revisions 209554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ 2009-07-30 14:40 +0000 [r209518] Mark Michelson * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 | mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8 lines Fix a crash that can result if text codecs are allowed but textsupport is disabled. (closes issue #15596) Reported by: fabled Patches: sip-red.patch uploaded by fabled (license 448) ........ 2009-07-28 Leif Madsen * Release Asterisk 1.6.2.0-beta4 2009-07-28 00:19 +0000 [r209328] Tilghman Lesher * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009) | 9 lines Merged revisions 209315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) | 2 lines Publish French extra sounds ........ ................ 2009-07-27 21:44 +0000 [r209265-209282] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 | kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7 lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages, clean up some looping logic, and correct an improper use of ast_free() for freeing an ast_frame. ........ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 | kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10 lines Make T.38 switchover in ReceiveFAX synchronous. In receive mode, if the channel that ReceiveFAX is running on supports T.38, we should *always* attempt to switch T.38, rather than listening for an incoming CNG tone and only triggering on that. The channel may be using a low-bitrate codec that distorts the CNG tone, the sending FAX endpoint may not send CNG at all, or there could be a variety of other reasons that we don't detect it, but in all those cases if T.38 is available we certainly want to use it. ........ 2009-07-27 20:58 +0000 [r209238] Mark Michelson * main/rtp.c, /: Merged revisions 209235 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 | mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5 lines Gracefully handle malformed RTP text packets. AST-2009-004 ........ 2009-07-27 20:33 +0000 [r209234] David Brooks * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /, include/asterisk/module.h, main/features.c, res/res_agi.c: Merged revisions 209098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ 2009-07-27 20:23 +0000 [r209135-209222] Mark Michelson * res/res_musiconhold.c, /: Merged revisions 209197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul 2009) | 9 lines Honor channel's music class when using realtime music on hold. (closes issue #15051) Reported by: alexh Patches: 15051.patch uploaded by mmichelson (license 60) Tested by: alexh ........ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions 209132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) | 24 lines Merged revisions 209131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ ................ 2009-07-27 16:07 +0000 [r209063] David Brooks * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing typos "recieved" with "received". From issue #15360, forgot to apply to trunk and other branches. 2009-07-27 15:40 +0000 [r209059] Kevin P. Fleming * Makefile, /: Merged revisions 209056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 | kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10 lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes. During the recent Makefile improvements I made, it seemed the 'make' was automatically carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so I removed the explict export of them. However, there are some circumstances where make does this, and some where it does not, so I've brought them back to ensure they are always exported. I also removed an extraneous double setting of _ASTLDFLAGS on *BSD platforms. ........ 2009-07-27 01:23 +0000 [r208927] Jeff Peeler * channels/chan_iax2.c, /, main/translate.c: Merged revisions 208924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) | 9 lines Merged revisions 208923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines Fix logic errors from 208746 ........ ................ 2009-07-26 14:07 +0000 [r208889] Michiel van Baak * contrib/scripts/install_prereq, /: Merged revisions 208886 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26 Jul 2009) | 2 lines add OpenBSD to the install_prereq script ........ 2009-07-25 12:31 +0000 [r208816-208853] Michiel van Baak * contrib/scripts/install_prereq, /: Merged revisions 208848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25 Jul 2009) | 2 lines libxml2-dev is needed as well by default. ........ * main/cli.c, /, configs/cli_aliases.conf.sample: Merged revisions 208813 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 | mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10 lines add default alias reload to run module reload. Requiring 'module reload' to reload everything, including core etc makes russell very unhappy. The default configuration already loads the 'friendly' aliases template. Added 'reload=module reload' to that template. Also removed the comment in main/cli.c that reload should come back. ........ 2009-07-25 06:26 +0000 [r208755] Jeff Peeler * channels/chan_iax2.c, /, channels/chan_skinny.c, main/translate.c: Merged revisions 208749 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) | 13 lines Merged revisions 208746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. ........ ................ 2009-07-24 21:13 +0000 [r208695-208710] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 208709 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208709 | russell | 2009-07-24 16:12:43 -0500 (Fri, 24 Jul 2009) | 2 lines Remove trailing whitespace. ........ * main/cli.c, /: Merged revisions 208706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208706 | russell | 2009-07-24 15:54:37 -0500 (Fri, 24 Jul 2009) | 6 lines Note that "reload" needs to be added back. I keep getting annoyed at having to type "module reload" to reload everything, so I'm adding a note that we need to add "reload" back. "module reload" doesn't really make sense as the command to reload everything, including the core. ........ * main/cli.c, /: Merged revisions 208693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208693 | russell | 2009-07-24 15:25:23 -0500 (Fri, 24 Jul 2009) | 2 lines Don't log a warning for something that does not affect operation. ........ 2009-07-24 19:42 +0000 [r208664] Mark Michelson * /: Fixing trunk-blocked property. 2009-07-24 18:56 +0000 [r208596] Russell Bryant * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) | 14 lines Merged revisions 208592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines Do not log an ERROR if autoservice_stop() returns -1. This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) ........ ................ 2009-07-24 18:32 +0000 [r208591] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16 lines Merged revisions 208587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines Only send a BYE when hanging up a channel that is up. For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac ........ ................ 2009-07-24 15:06 +0000 [r208551] Kevin P. Fleming * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 208548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ 2009-07-24 14:39 +0000 [r208545] Michiel van Baak * contrib/scripts/install_prereq, /: Merged revisions 208542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24 Jul 2009) | 13 lines use aptitude for debian based systems The function to check wether we need to install packages was using dpkg-query which was gives wrong output on Debian 5 Also, the apt-get has been replaced with aptitude because aptitude is now the preferred way to handle packages on Debian (closes issue #15570) Reported by: mvanbaak Patches: 2009072400_installprereq-aptitude.diff uploaded by mvanbaak (license 7) ........ 2009-07-23 22:31 +0000 [r208501] Kevin P. Fleming * include/asterisk/frame.h, main/rtp.c, main/channel.c, main/udptl.c, main/frame.c, /, channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h: Merged revisions 208464 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ 2009-07-23 19:36 +0000 [r208391] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul 2009) | 24 lines Merged revisions 208386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) | 17 lines Fix a problem where a 491 response could be sent out of dialog. This generalizes the fix for issue 13849. The initial fix corrected the problem that Asterisk would reply with a 491 if a reinvite were received from an endpoint and we had not yet received an ACK from that endpoint for the initial INVITE it had sent us. This expansion also allows Asterisk to appropriately handle an INVITE with authorization credentials if Asterisk had not received an ACK from the previous transaction in which Asterisk had responded to an unauthorized INVITE with a 407. (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ ................ 2009-07-23 19:25 +0000 [r208387] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500 (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) | 6 lines Only set the priindication setting when not performing a reload (closes issue #14696) Reported by: fdecher ........ ................ 2009-07-23 16:30 +0000 [r208266-208320] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul 2009) | 9 lines Merged revisions 208312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) | 3 lines Remove inaccurate XXX comment. ........ ................ * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul 2009) | 15 lines Merged revisions 208262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) | 8 lines Properly handle 183 responses which do not contain an SDP. (closes issue #15442) Reported by: ffloimair Patches: 15442.patch uploaded by mmichelson (license 60) Tested by: tkarl, ffloimair ........ ................ 2009-07-22 21:46 +0000 [r208116] Jason Parker * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 | qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines Restore an int declaration on PPC platforms. This x is one crafty little bugger... It was used for 2 different things (one of which was only done on PPC) in 1.4. One of the uses were removed in trunk, and with it went the declaration. (closes issue #14038) Reported by: ffloimair ........ 2009-07-22 16:52 +0000 [r207949-208053] Tilghman Lesher * /, res/res_realtime.c: Merged revisions 208052 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208052 | tilghman | 2009-07-22 11:49:42 -0500 (Wed, 22 Jul 2009) | 7 lines Clarify documentation on 'realtime update2' to show more than one condition. (closes issue #15357) Reported by: snuffy Patches: bug_fix_doc_update2.diff uploaded by snuffy (license 35) (slightly modified by me) ........ * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500 (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) | 8 lines Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional). This change makes URIENCODE and QUOTE behave similarly, since the documentation states that the argument is not optional, for both. (closes issue #15439) Reported by: pkempgen Patches: 20090706__issue15439.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-07-21 22:23 +0000 [r207930] Russell Bryant * doc/CODING-GUIDELINES, /: Merged revisions 207925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207925 | russell | 2009-07-21 17:22:18 -0500 (Tue, 21 Jul 2009) | 4 lines Note that we use tabs instead of spaces for indentation. I'm surprised this was never actually in here... ........ 2009-07-21 20:30 +0000 [r207785-207862] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500 (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines Wait for wink before dialing when using E&M wink signaling There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simpl