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2010-02-01Merged revisions 244243 via svnmerge from tilghman1-6/+29
https://origsvn.digium.com/svn/asterisk/trunk ................ r244243 | tilghman | 2010-02-01 17:16:12 -0600 (Mon, 01 Feb 2010) | 18 lines Merged revisions 244242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244242 | tilghman | 2010-02-01 17:13:44 -0600 (Mon, 01 Feb 2010) | 11 lines Backup and restore original textfile, for prosthesis (gerund of prepend). Also, fix menuselect such that changing voicemail build options correctly causes rebuild. (closes issue #16415) Reported by: tomo1657 Patches: prepention.patch uploaded by tomo1657 (license 484) (with modifications by me to backport to 1.4) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@244303 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Merged revisions 243693 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r243693 | jpeeler | 2010-01-27 14:37:33 -0600 (Wed, 27 Jan 2010) | 12 lines Merged revisions 243691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010) | 5 lines Revert 243570, I should have looked at this closer. Will reopen the issue, but am leaving the review closed as the change was pointless. (issue #16488) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@243694 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Merged revisions 243571 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r243571 | jpeeler | 2010-01-27 12:49:52 -0600 (Wed, 27 Jan 2010) | 16 lines Merged revisions 243570 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010) | 9 lines Extend announcement URL used with Queue from 80 chars to PATH_MAX. (closes issue #16488) Reported by: syspert Patches: soundfilelen.pacth-2 uploaded by syspert (license 938) Review: https://reviewboard.asterisk.org/r/475/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@243572 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-18Merged revisions 240842 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r240842 | dvossel | 2010-01-18 09:52:55 -0600 (Mon, 18 Jan 2010) | 2 lines fixes spelling error. s/memeber/member ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@240845 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-15Merged revisions 240415 via svnmerge from tilghman1-3/+73
https://origsvn.digium.com/svn/asterisk/trunk ................ r240415 | tilghman | 2010-01-15 14:54:24 -0600 (Fri, 15 Jan 2010) | 22 lines Merged revisions 240414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15 Jan 2010) | 15 lines Disallow leaving more than maxmsg voicemails. This is a possibility because our previous method assumed that no messages are left in parallel, which is not a safe assumption. Due to the vmu structure duplication, it was necessary to track in-process messages via a separate structure. If at some point, we switch vmu to an ao2-reference-counted structure, which would eliminate the prior noted duplication of structures, then we could incorporate this new in-process structure directly into vmu. (closes issue #16271) Reported by: sohosys Patches: 20100108__issue16271.diff.txt uploaded by tilghman (license 14) 20100108__issue16271__trunk.diff.txt uploaded by tilghman (license 14) 20100108__issue16271__1.6.0.diff.txt uploaded by tilghman (license 14) Tested by: jsutton ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@240416 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-13Merged revisions 239712 via svnmerge from dvossel2-0/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r239712 | dvossel | 2010-01-13 10:31:14 -0600 (Wed, 13 Jan 2010) | 24 lines add silence gen to wait apps asterisk.conf's 'transmit_silence' option existed before this patch, but was limited to only generating silence while recording and sending DTMF. Now enabling the transmit_silence option generates silence during wait times as well. To achieve this, ast_safe_sleep has been modified to generate silence anytime no other generators are present and transmit_silence is enabled. Wait apps not using ast_safe_sleep now generate silence when transmit_silence is enabled as well. (closes issue #16524) Reported by: kobaz (closes issue #16523) Reported by: kobaz Tested by: dvossel Review: https://reviewboard.asterisk.org/r/456/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@239716 f38db490-d61c-443f-a65b-d21fe96a405b
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/1.6.0@238364 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/1.6.0@238182 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/1.6.0@238011 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/1.6.0@237923 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/1.6.0@236668 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/1.6.0@236511 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/1.6.0@236301 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/1.6.0@235011 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/1.6.0@234894 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/1.6.0@234404 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/1.6.0@233167 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Merged revisions 232854 via svnmerge from tilghman1-153/+279
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/1.6.0@232864 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Merged revisions 232587 via svnmerge from diruggles1-58/+57
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/1.6.0@232811 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/1.6.0@232357 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/1.6.0@231691 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/1.6.0@231560 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-19Merged revisions 230509 via svnmerge from dvossel1-24/+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/1.6.0@230512 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/1.6.0@230382 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/1.6.0@230344 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/1.6.0@229967 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/1.6.0@229475 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/1.6.0@229352 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/1.6.0@228190 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/1.6.0@227833 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/1.6.0@227369 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/1.6.0@226891 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/1.6.0@225869 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/1.6.0@225361 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/1.6.0@224568 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Merged revisions 224448 via svnmerge from tilghman1-2/+18
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. This corrects an issue reported on the -users list. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224449 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/1.6.0@224179 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/1.6.0@223833 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/1.6.0@223653 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/1.6.0@223331 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/1.6.0@223226 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Merged revisions 222176 via svnmerge from kpfleming1-5/+42
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/1.6.0@222185 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/1.6.0@221087 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/1.6.0@220940 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-29Merged revisions 220833 via svnmerge from jpeeler1-5/+7
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/1.6.0@220834 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/1.6.0@220290 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-22Merged revisions 219818 via svnmerge from tilghman1-0/+24
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219819 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-18Merged revisions 219412 via svnmerge from tilghman1-0/+1
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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219413 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218731 via svnmerge from tilghman1-8/+7
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) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218732 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218579 via svnmerge from tilghman1-0/+4
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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@218580 f38db490-d61c-443f-a65b-d21fe96a405b
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