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2010-01-07Merged revisions 238361 via svnmerge from dvossel1-1/+1
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@238363 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-06Merged revisions 238181 via svnmerge from jpeeler1-1/+1
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@238183 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-06Merged revisions 238010 via svnmerge from russell1-2/+4
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) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@238012 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-05Merged revisions 237920 via svnmerge from dvossel1-5/+2
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@237922 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-04Merged revisions 237327 via svnmerge from dvossel1-0/+5
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@237329 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-28Merged revisions 236667 via svnmerge from tilghman1-1/+1
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@236669 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-28Merged revisions 236510 via svnmerge from seanbright1-30/+34
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@236512 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-23Merged revisions 236300 via svnmerge from tilghman1-2/+2
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@236302 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-15Merged revisions 235010 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r235010 | kpfleming | 2009-12-15 08:35:46 -0600 (Tue, 15 Dec 2009) | 5 lines spandsp does in fact support V.17 modulation at 14.4 kilobits per second, so we should generate T38MaxBitRate of 14400 (even though that doesn't really affect the FAX transmission much at all) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@235012 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-15Merged revisions 234893 via svnmerge from alecdavis1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r234893 | alecdavis | 2009-12-15 15:29:50 +1300 (Tue, 15 Dec 2009) | 9 lines fixes escape to extensions 'o' and 'a', for digits '0' and '*' (closes issue #16437) Reported by: alecdavis Tested by: alecdavis Patch extension_o_a_fix.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@234895 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-15Merged revisions 234855 via svnmerge from alecdavis1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r234855 | alecdavis | 2009-12-15 13:54:44 +1300 (Tue, 15 Dec 2009) | 9 lines ast_stream_and_wait(chan,dir-usingkeypad) didn't capture the dialled DTMF. (closes issue #16409) Reported by: alecdavis Tested by: alecdavis Patch bug_16409.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@234862 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-11Merged revisions 234380 via svnmerge from jpeeler1-20/+38
https://origsvn.digium.com/svn/asterisk/trunk ................ r234380 | jpeeler | 2009-12-11 17:17:09 -0600 (Fri, 11 Dec 2009) | 18 lines Merged revisions 234379 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) | 11 lines Fix talking detection status after conference user is muted. This patch ensures that when a conference user is muted that the accompanying AMI Meetme talking off event is sent. Also, the meetme list output is updated to show the muted user as unmonitored. (closes issue #16247) Reported by: dimas Patches: v3-16247.patch uploaded by dimas (license 88) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@234426 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-08Fixed compile error with OSP Toolkit 3.6.transnexus1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@233688 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-04Merged revisions 233121 via svnmerge from dvossel1-3/+9
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@233166 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Merged revisions 232854 via svnmerge from tilghman1-66/+79
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) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@232865 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Merged revisions 232587 via svnmerge from diruggles1-46/+45
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/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@232812 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Merged revisions 232356 via svnmerge from file1-0/+1
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@232358 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Merged revisions 231688 via svnmerge from mnicholson1-2/+10
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/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@231690 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Turn off debug mode in 1.6.1; fix such that debug mode and non-debug mode ↵tilghman1-5/+5
functions return the same types. (Fixes an issue brought up in chat by twilson) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@231608 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Merged revisions 231556 via svnmerge from dvossel1-2/+13
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@231559 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-24Merged revisions 231134 via svnmerge from tilghman1-73/+88
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@231135 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-19Merged revisions 230509 via svnmerge from dvossel1-23/+70
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/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230511 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-16Merged revisions 230381 via svnmerge from kpfleming1-2/+3
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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230383 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-16Merged revisions 230343 via svnmerge from kpfleming1-1/+5
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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230345 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Merged revisions 229966 via svnmerge from file1-0/+5
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@229968 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-11Merged revisions 229460 via svnmerge from dbrooks1-1/+1
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@229491 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-10Merged revisions 229351 via svnmerge from tilghman1-1/+11
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@229353 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05Merged revisions 228196 via svnmerge from tilghman1-1/+2
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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@228197 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05Merged revisions 228191 via svnmerge from tilghman1-1/+2
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@228193 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05Merged revisions 228189 via svnmerge from jpeeler1-1/+2
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@228192 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05Merged revisions 228015 via svnmerge from tilghman1-0/+5
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@228016 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Merged revisions 227829 via svnmerge from mnicholson1-13/+62
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/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@227832 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Merged revisions 227368 via svnmerge from lmadsen1-1/+1
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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@227370 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02Merged revisions 226890 via svnmerge from file1-4/+27
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) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@226892 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-26Backport audio handling loop fixes from trunk version of app_fax.kpfleming1-18/+23
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. (issue #16127) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@225870 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-22Merged revisions 225445 via svnmerge from dvossel1-3/+1
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/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@225490 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-22Merged revisions 225360 via svnmerge from tilghman1-1/+1
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@225362 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Merged revisions 224567 via svnmerge from file1-4/+9
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@224570 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@224450 ↵tilghman1-84/+92
f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-15Merged revisions 224178 via svnmerge from jpeeler1-1/+6
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@224180 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223832 via svnmerge from jpeeler1-1/+3
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@223834 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223652 via svnmerge from kpfleming1-1/+54
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@223654 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223330 via svnmerge from kpfleming1-2/+2
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@223332 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223215 via svnmerge from mmichelson1-0/+1
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@223241 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Merged revisions 222176 via svnmerge from kpfleming1-5/+44
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/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@222186 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221436 via svnmerge from mnick1-1/+1
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@221470 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221085 via svnmerge from seanbright1-1/+12
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@221088 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-29Avoid a deadlock in chanspy, just in case the spyee is masqueraded and ↵mnicholson1-3/+4
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@220938 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-29Merged revisions 220833 via svnmerge from jpeeler1-6/+8
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@220835 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-24Merged revisions 220289 via svnmerge from tilghman2-9/+1
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@220291 f38db490-d61c-443f-a65b-d21fe96a405b
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