2009-11-18 Leif Madsen * Release Asterisk 1.6.1.10 2009-11-13 Leif Madsen * Release Asterisk 1.6.1.10-rc3 2009-11-13 15:57 +0000 [r229914] 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 [r229751] 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 16:48 +0000 [r229672] 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 20:48 +0000 [r229569] 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 19:54 +0000 [r229491-229501] 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 [r229364] 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 [r229358] 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 21:31 +0000 [r229353] 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 [r229284] 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 17:53 +0000 [r229233] 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 17:38 +0000 [r229230] 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 15:38 +0000 [r229099] Matthew Nicholson * channels/chan_sip.c: Reverted revision 202008. (closes issue #16175) Reported by: paul-tg 2009-11-10 15:36 +0000 [r229095-229098] David Vossel * res/res_config_pgsql.c: reverting changes made by r229095 as they are not applicable to 1.6.1 * 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 11:22 +0000 [r229067] 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 22:52 +0000 [r229016] 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 22:18 +0000 [r229014] 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 0015848) 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 1.6.1.10-rc2 2009-11-09 15:39 +0000 [r228899] 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 14:54 +0000 [r228860] 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 [r228695] 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 20:37 +0000 [r228650] 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 18:41 +0000 [r228550] 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 1.6.1.10-rc1 2009-11-06 17:53 +0000 [r228502] 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 [r228422-228451] 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 [r228412] 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 15:44 +0000 [r228267-228341] 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 ........ * /, 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 ........ 2009-11-05 22:13 +0000 [r228193-228197] 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) ........ * /, 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 21:24 +0000 [r228192] 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 19:41 +0000 [r228147] 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 19:19 +0000 [r228090] 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 17:10 +0000 [r228016] 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 23:56 +0000 [r227948] 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 21:15 +0000 [r227761-227832] 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 19:27 +0000 [r227723-227745] 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 [r227374] 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 [r227363-227370] 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 18:11 +0000 [r227279] 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 15:38 +0000 [r227169] 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 15:24 +0000 [r227164] 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 13:32 +0000 [r227155] Olle Johansson * /, 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 21:05 +0000 [r226977] 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 18:11 +0000 [r226892] 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 [r226814] 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 18:15 +0000 [r226534] Joshua Colp * channels/chan_local.c, /, doc/tex/localchannel.tex: 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 20:15 +0000 [r226380-226386] 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 18:05 +0000 [r226169-226307] 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 17:04 +0000 [r226100] 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 ........ 2009-10-26 23:48 +0000 [r226053] 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 19:41 +0000 [r225913] 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 15:51 +0000 [r225870] 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. (issue #16127) 2009-10-23 14:50 +0000 [r225652] 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 14:07 +0000 [r225584] 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 22:07 +0000 [r225490] David Vossel * main/tcptls.c, /, channels/chan_sip.c, apps/app_externalivr.c, include/asterisk/tcptls.h: 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 [r225487] 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 [r225362] 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 22:02 +0000 [r225062-225309] 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/ ........ ................ * channels/chan_iax2.c, configs/iax.conf.sample, /, channels/chan_sip.c, configs/sip.conf.sample: 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/ ........ ................ 2009-10-21 03:17 +0000 [r224935] Russell Bryant * include/asterisk/frame.h, include/asterisk/translate.h, main/dsp.c, main/frame.c, /, main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c: Merged revisions 224932 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224932 | russell | 2009-10-20 22:09:04 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ ................ 2009-10-20 22:11 +0000 [r224858] Tilghman Lesher * funcs/func_speex.c, /, main/audiohook.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 17:49 +0000 [r224776] 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 ........ ................ 2009-10-19 23:56 +0000 [r224673] 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 19:51 +0000 [r224570] 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 ........ ................ 2009-10-17 02:02 +0000 [r224333-224336] Jeff Peeler * channels/chan_dahdi.c: fix typo, sorry * 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 20:53 +0000 [r224263] 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 [r224180] 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 [r223834] 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 21:03 +0000 [r223758] 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 14:32 +0000 [r223654] 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 17:31 +0000 [r223489] 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 23:11 +0000 [r223405] 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 21:00 +0000 [r223332] 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 18:36 +0000 [r223277] 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 18:25 +0000 [r223241] 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 17:56 +0000 [r223209] 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 [r223171] 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 17:10 +0000 [r223090-223134] 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 19:57 +0000 [r222882] 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 19:42 +0000 [r222875] 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 16:49 +0000 [r222694-222801] 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 17:46 +0000 [r222545] 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 23:58 +0000 [r222353-222465] 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) (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) Modified: branches/1.4/channels/chan_dahdi.c * 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 19:34 +0000 [r222310] Tilghman Lesher * res/res_config_pgsql.c, /, cdr/cdr_pgsql.c: Recorded merge of 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 19:26 +0000 [r222303] 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 19:20 +0000 [r222282] 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 ........ 2009-11-04 Leif Madsen * Release Asterisk 1.6.1.9 * AST-2009-008 and AST-2009-009 2009-10-26 Leif Madsen * Release Asterisk 1.6.1.8 * AST-2009-007 2009-10-06 Leif Madsen * Release Asterisk 1.6.1.7-rc2 2009-10-06 01:36 +0000 [r222112-222186] Kevin P. Fleming * apps/app_queue.c, channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c, /, channels/chan_sip.c, funcs/func_dialgroup.c, include/asterisk/astobj2.h, res/res_phoneprov.c, channels/chan_console.c, res/res_musiconhold.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/ ........ ................ * main/udptl.c, /, channels/chan_sip.c, configs/udptl.conf.sample, UPGRADE.txt, configs/sip.conf.sample: 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 17:36 +0000 [r222035] 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 [r221969-221973] Tilghman Lesher * main/astobj2.c, /, funcs/func_lock.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. ........ ................ * funcs/func_lock.c: Hash needs to return a positive integer 2009-10-02 13:04 +0000 [r221964] Sean Bright * funcs/func_strings.c: Revert XML docs that ended up in the 1.6.0 and 1.6.1 branches during a merge. 2009-10-02 03:06 +0000 [r221922] Tilghman Lesher * /, 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 01:26 +0000 [r221871] 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 00:06 +0000 [r221743-221779] 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 19:52 +0000 [r221702] 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 17:01 +0000 [r221661] 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 16:19 +0000 [r221602] 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 23:10 +0000 [r221478-221487] 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 ........ * /, channels/chan_sip.c, configs/sip.conf.sample: 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 21:41 +0000 [r221370-221470] 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 18:58 +0000 [r221302] Terry Wilson * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h, configs/sip.conf.sample: 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 [r221203] 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 14:55 +0000 [r221088] 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 [r220998-221046] 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 ........ * /, 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 20:25 +0000 [r220938] 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 2009-09-29 17:05 +0000 [r220835] 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 [r220724] 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. ........ ................ 2009-09-26 15:12 +0000 [r220588] 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 [r220371] 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 [r220291] 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 [r220102-220220] 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 08:40 +0000 [r220030] 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 07:44 +0000 [r219988] 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 21:47 +0000 [r219820] 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 17:02 +0000 [r219723] 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 [r219667] 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 03:10 +0000 [r219589] 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 23:22 +0000 [r219453-219522] 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 [r219414] 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 22:36 +0000 [r219367] 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 22:04 +0000 [r219306] 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 [r219266] 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 1.6.1.7-rc1 2009-09-17 15:44 +0000 [r219199] 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 [r219062] 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 19:27 +0000 [r218936] 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 19:24 +0000 [r218932] 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 18:23 +0000 [r218890] 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 13:37 +0000 [r218801] 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 [r218727-218734] 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) ........ ................ * /, channels/chan_gtalk.c: Merged revisions 139281,175058,175089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk (closes issue #13985) ................ r139281 | phsultan | 2008-08-21 04:55:31 -0500 (Thu, 21 Aug 2008) | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel! (closes issue #13310) Reported by: eliel Patches: chan_gtalk.c.patch uploaded by eliel (license 64) ................ r175058 | phsultan | 2009-02-12 04:31:36 -0600 (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ ................ r175089 | phsultan | 2009-02-12 08:25:03 -0600 (Thu, 12 Feb 2009) | 6 lines Issue a warning message if our candidate's IP is the loopback address. (closes issue #13985) Reported by: jcovert Tested by: phsultan ................ 2009-09-15 19:27 +0000 [r218689] 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 [r218592] 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 16:05 +0000 [r218581] 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 15:42 +0000 [r218574] 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. ........ 2009-09-15 15:41 +0000 [r218569] 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 15:12 +0000 [r218506] Mark Michelson * /, 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 15:04 +0000 [r218502] 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 [r218363] 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 18:17 +0000 [r218297] 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 15:17 +0000 [r218227] 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 21:48 +0000 [r218218] 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 2009-09-12 13:15 +0000 [r218112] Michiel van Baak * main/rtp.c: Use the ip for the new 'rtp set debug ip '. Since 1.6.X still has the deprecated 'rtp debug ip ' this patch is different from the fix that went into trunk (closes issue 0015711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) Tested by: davidw 2009-09-11 05:59 +0000 [r217924-218054] 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 21:23 +0000 [r217826] 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 19:55 +0000 [r217738] 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 18:18 +0000 [r217642] 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 12:11 +0000 [r217595] 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 20:30 +0000 [r217518] 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 http://svn.digium.com/svn/asterisk/trunk 2009-09-09 11:02 +0000 [r217370] 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 [r217295] 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 20:32 +0000 [r217213] 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 [r217076] 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 [r217035] 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 [r216995] 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 16:41 +0000 [r216646-216844] 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 setting in the INVITE, set it to default value (not unset minimum = -1) Patch by oej closes issue #15621 Reported by: fnordian Tested by: atis ........ * configs/sip.conf.sample: Make code and documentation agree with each other * CHANGES, channels/chan_sip.c: Turning off premature media by default * apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c, apps/app_disa.c, configs/sip.conf.sample: 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 19:51 +0000 [r216599] 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 [r216596] 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 [r216549-216552] 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. ........ * main/pbx.c, /, UPGRADE-1.6.txt: 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 15:09 +0000 [r216440-216508] 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 13:56 +0000 [r216266-216434] 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 06:13 +0000 [r216224] 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 19:42 +0000 [r216013-216098] Russell Bryant * UPGRADE.txt: tweak * /, 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 18:41 +0000 [r216004] 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 Leif Madsen * Asterisk 1.6.1.6 released * AST-2009-006 2009-08-28 Leif Madsen * Asterisk 1.6.1.5 released 2009-08-11 Tilghman Lesher * Asterisk 1.6.1.5-rc1 released 2009-08-10 19:51 +0000 [r211569-211586] 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 ........ ................ * channels/chan_iax2.c, res/ael/pval.c, main/cdr.c, main/channel.c, main/manager.c, apps/app_setcallerid.c, apps/app_rpt.c, main/asterisk.c, res/res_config_pgsql.c, apps/app_dahdibarge.c, funcs/func_rand.c, funcs/func_timeout.c, apps/app_record.c, codecs/codec_speex.c, apps/app_morsecode.c, main/acl.c, funcs/func_cut.c, cdr/cdr_pgsql.c, apps/app_followme.c, main/enum.c, res/res_config_sqlite.c, main/config.c, agi/eagi-sphinx-test.c, channels/misdn_config.c, channels/chan_dahdi.c, funcs/func_channel.c, apps/app_macro.c, apps/app_sms.c, pbx/pbx_config.c, apps/app_verbose.c, main/dsp.c, apps/app_voicemail.c, apps/app_adsiprog.c, funcs/func_speex.c, channels/chan_sip.c, res/res_limit.c, channels/chan_agent.c, agi/eagi-test.c, funcs/func_math.c, main/utils.c, channels/iax2-provision.c, apps/app_talkdetect.c, main/indications.c, channels/chan_oss.c, main/cli.c, res/res_config_curl.c, pbx/pbx_loopback.c, res/res_smdi.c, apps/app_osplookup.c, channels/chan_misdn.c, channels/chan_skinny.c, pbx/pbx_dundi.c, utils/extconf.c, apps/app_mixmonitor.c, channels/chan_mgcp.c, main/timing.c, main/pbx.c, doc/CODING-GUIDELINES, utils/muted.c, apps/app_readfile.c, /, apps/app_meetme.c, apps/app_privacy.c, apps/app_waituntil.c, cdr/cdr_adaptive_odbc.c, pbx/dundi-parser.c, res/res_http_post.c, res/res_musiconhold.c, apps/app_queue.c, main/netsock.c, utils/frame.c, channels/chan_usbradio.c, funcs/func_enum.c, channels/chan_phone.c, apps/app_waitforring.c, pbx/pbx_spool.c, funcs/func_odbc.c, apps/app_minivm.c, main/features.c, res/res_agi.c, main/http.c, res/snmp/agent.c, res/res_config_ldap.c, apps/app_chanspy.c, apps/app_stack.c, res/res_odbc.c, funcs/func_dialplan.c, main/dnsmgr.c, main/frame.c, apps/app_waitforsilence.c, funcs/func_strings.c, apps/app_disa.c, apps/app_alarmreceiver.c: AST-2009-005 2009-08-10 14:12 +0000 [r211349] 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 [r211234-211277] 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 20:17 +0000 [r211115] 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 18:19 +0000 [r211047] 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 13:09 +0000 [r210994] 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 [r210910-210916] 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 17:48 +0000 [r210819] 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 20:28 +0000 [r210681] 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 18:57 +0000 [r210567] Leif Madsen * 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 14:54 +0000 [r210240] 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 ........ ................ 2009-08-01 11:32 +0000 [r209836-209900] 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 01:25 +0000 [r209781] Kevin P. Fleming * channels/misdn/isdn_lib.c, utils/frame.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 21:58 +0000 [r209714] 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 18:46 +0000 [r209593] David Brooks * include/asterisk/abstract_jb.h, channels/chan_dahdi.c, contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, codecs/lpc10/pitsyn.c, channels/chan_console.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 [r209517] 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 00:19 +0000 [r209327] 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 [r209262-209281] 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 20:57 +0000 [r209237] 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 20:28 +0000 [r209233] 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 20:17 +0000 [r209134-209199] 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 15:40 +0000 [r209058] 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 01:22 +0000 [r208926] 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 14:04 +0000 [r208888] 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 06:25 +0000 [r208754] 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 18:52 +0000 [r208595] 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 [r208590] 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 15:05 +0000 [r208550] 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 14:38 +0000 [r208544] 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 22:32 +0000 [r208484-208503] Kevin P. Fleming * UPGRADE.txt: Use correct formatting for T.38 change note in UPGRADE.txt * 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 20:45 +0000 [r208459] David Brooks * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing typos "recieved" with "received". (closes issue #15360) Reported by: okrief 2009-07-23 19:35 +0000 [r208390] 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 19:24 +0000 [r208385] 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 [r208265-208318] 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 21:45 +0000 [r208115] 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 ........ 2009-07-21 22:48 +0000 [r207948] Tilghman Lesher * /, 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 20:29 +0000 [r207784-207861] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500 (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines Wait for wink before dialing when using E&M wink signaling There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simply add SIG_EMWINK to the list. (closes issue #14434) Reported by: araasch ........ ................ * channels/chan_dahdi.c: Revert r207637, this approach could potentially block for an unacceptable amount of time. 2009-07-21 14:31 +0000 [r207726] Mark Michelson * main/manager.c, /: Merged revisions 207723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul 2009) | 11 lines Merged revisions 207714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) | 5 lines Document default timeout for AMI originations. AST-224 ........ ................ 2009-07-21 13:48 +0000 [r207684] Kevin P. Fleming * channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, funcs/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules, pbx/Makefile, res/Makefile: Merged revisions 207680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either* in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. ........ ................ 2009-07-21 04:45 +0000 [r207637] Jeff Peeler * channels/chan_dahdi.c: Wait for wink before dialing when using E&M wink signaling This patch adds a new dahdi_wait function to specifically wait for the wink event. If the wink is not eventually received the channel is hung up. (closes issue #14434) Reported by: araasch Patches: emwinkmod uploaded by araasch (license 693) 2009-07-20 20:02 +0000 [r207426] Mark Michelson * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) | 39 lines Merged revisions 207423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines Answer video SDP offers properly when videosupport is not enabled. Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch *only* helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 ........ ................ 2009-07-20 16:40 +0000 [r207363] Russell Bryant * main/channel.c, /: Merged revisions 207361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009) | 16 lines Merged revisions 207360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines Only do the chan->fdno check in ast_read() in a developer build. I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller ........ ................ 2009-07-18 04:17 +0000 [r207321] Tilghman Lesher * apps/app_voicemail.c, /: Recorded merge of revisions 207317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17 Jul 2009) | 3 lines Flag field in wrong position. Reported by "Hoggins!" on asterisk-dev list. ........ 2009-07-18 02:09 +0000 [r207287] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, doc/tex/misdn.tex, channels/chan_misdn.c, main/callerid.c, configs/misdn.conf.sample: Merged revisions 145293,158010 from https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. These changes are already on trunk. ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. ................ r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines Merged revision 157977 from https://origsvn.digium.com/svn/asterisk/team/group/issue8824 ........ Fixes JIRA ABE-1726 The dial extension could be empty if you are using MISDN_KEYPAD to control ISDN provider features. ................ 2009-07-17 22:30 +0000 [r207227-207256] Tilghman Lesher * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17 Jul 2009) | 2 lines Add flag here, too (as requested by jsmith) ........ * /, doc/tex/odbcstorage.tex, UPGRADE.txt: Recorded merge of revisions 207224 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17 Jul 2009) | 2 lines Document the "flag" field in the voicemessages table. ........ 2009-07-17 19:39 +0000 [r207101-207158] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500 (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines Fix format specifier to print out an unsigned long long. Yep, it's even ifdefed out code. But it made it to the RR list... (closes issue #14726) Reported by: lmadsen ........ ................ * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) | 2 lines Update some missing allowed options for overlapdial ........ 2009-07-17 17:53 +0000 [r206870-207031] David Vossel * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 | dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines sip option flags handled incorrectly (closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima ........ * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines Merged revisions 206938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ ................ * /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009) | 6 lines TIMEOUT(absolute) returned negative value. (closes issue #15513) Reported by: ys ........ * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ ................ * /, main/callerid.c: Merged revisions 206868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) | 14 lines Merged revisions 206867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines avoid segfault caused by user error If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. ........ ................ 2009-07-16 16:53 +0000 [r206810] Tilghman Lesher * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500 (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517) Reported by: adomjan Patches: func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487) ........ ................ 2009-07-15 22:06 +0000 [r206774] David Vossel * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 | dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines Session timer were not activated if Supported header field in INVITE had both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license ........ 2009-07-15 21:40 +0000 [r206764] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: Merged revisions 206707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) | 33 lines Merged revisions 206706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... ................ ................ 2009-07-15 20:21 +0000 [r206704] David Vossel * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 | dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines callerid(num) is wrong when username is missing A domain only sip uri would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor ........ 2009-07-15 16:03 +0000 [r206638] Sean Bright * /, codecs/codec_dahdi.c: Merged revisions 206636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400 (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we are asking for it. ........ ................ 2009-07-14 20:25 +0000 [r206596] Tilghman Lesher * /, contrib/scripts/meetme.sql: Recorded merge of revisions 206567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines Document all meetme realtime fields, and in the process, make some field lengths more consistent. (closes issue #15493) Reported by: lasko Patches: meetme.diff uploaded by lasko (license 833) ........ 2009-07-14 18:32 +0000 [r206558] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500 (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ ................ 2009-07-14 14:56 +0000 [r206388] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 206386 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206386 | russell | 2009-07-14 09:51:44 -0500 (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ ................ ................ 2009-07-14 01:35 +0000 [r206372] Richard Mudgett * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 206341 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009) | 11 lines Merged revisions 206284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ ................ 2009-07-13 23:33 +0000 [r206282] David Vossel * /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 | dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines dns lookup of peername rather than peer's host in transmit_register() (closes issue #15052) Reported by: fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by fsantulli (license 818) Tested by: fsantulli ........ 2009-07-13 16:24 +0000 [r206186] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009) | 2 lines Remove reference to non-existent help file ........ 2009-07-10 21:52 +0000 [r205987] David Vossel * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 | dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines SIP register not using peer's outbound proxy If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ ........ 2009-07-10 18:45 +0000 [r205941] Kevin P. Fleming * main/udptl.c, /: Merged revisions 205939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 | kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line Update comments about the level of T.38 support in Asterisk. ........ 2009-07-10 17:50 +0000 [r205881] Mark Michelson * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) | 30 lines Merged revisions 205877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ ................ 2009-07-10 16:48 +0000 [r205842] David Vossel * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines Merged revisions 205804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ ................ 2009-07-10 15:57 +0000 [r205778] Mark Michelson * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ 2009-07-10 15:36 +0000 [r205772] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines Fix some remaining T.38 negotiation problems in app_fax. Revision 205696 did not quite fix all the issues with the T.38 negotiation changes and app_fax; this patch corrects them, along with a couple of other minor issues. (closes issue #15480) Reported by: dimas Patches: test2-15480.patch uploaded by dimas (license 88) ........ 2009-07-09 23:51 +0000 [r205730] Richard Mudgett * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller. Add missing clearing of the dialing flag when the ISDN call is CONNECTED. (i.e. When libpri generates the event PRI_EVENT_ANSWER.) (closes issue #15420) Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585) Tested by: scottbmilne, alecdavis (closes issue #15416) Reported by: avinoash (closes issue #15389) Reported by: alecdavis This patch should also fix the following issue: (issue #15205) Reported by: vinsik ........ 2009-07-09 21:27 +0000 [r205698] Kevin P. Fleming * include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 205696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ 2009-07-09 16:20 +0000 [r205596-205605] David Vossel * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ ................ * channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /: Merged revisions 205479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ 2009-07-09 08:33 +0000 [r205534] Michiel van Baak * /, main/ssl.c: Merged revisions 205532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 | mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines pthread_self returns a pthread_t which is not an unsigned int on all pthread implementations. Casting it to an unsigned int fixes compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit ........ 2009-07-08 22:16 +0000 [r205414] David Vossel * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c, include/asterisk/pbx.h: Merged revisions 205412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ ................ 2009-07-08 19:27 +0000 [r205352] Mark Michelson * apps/app_queue.c, /: Merged revisions 205350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines Merged revisions 205349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines Prevent phantom calls to queue members. If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) ........ ................ 2009-07-08 18:21 +0000 [r205299] Jason Parker * config.guess, config.sub, /: Merged revisions 205291 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205291 | qwell | 2009-07-08 13:19:46 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line Update config.guess and config.sub from the savannah.gnu.org git repo. ........ ................ 2009-07-08 18:07 +0000 [r205279] David Brooks * /, main/features.c: Merged revisions 205254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 | dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines Fixes Park() argument handling Park() was not respecting the arguments passed to it. Any extension/context/priority given to it was being ignored. This patch remedies this. (closes issue #15380) Reported by: DLNoah ........ 2009-07-08 16:59 +0000 [r205222] Tilghman Lesher * main/say.c: oops, fixing build 2009-07-08 16:56 +0000 [r205218] David Vossel * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ ................ 2009-07-08 16:29 +0000 [r205203] Tilghman Lesher * /, main/say.c: Merged revisions 205196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines Add redirection warnings for the invalid language codes previously removed. ........ ................ 2009-07-08 15:57 +0000 [r205147-205153] Russell Bryant * /, main/ssl.c: Merged revisions 205151 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 | russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines Use tabs instead of spaces for indentation. ........ * res/res_jabber.c, main/asterisk.c, /, main/Makefile, res/res_crypto.c, main/ssl.c (added), include/asterisk/_private.h: Merged revisions 205120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ 2009-07-06 14:24 +0000 [r204976] Ryan Brindley * main/config.c, /: Merged revisions 202753 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 | rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9 lines If we delete the info, lets also delete the lines (closes issue 0014509) Reported by: timeshell Patches: 20090504__bug14509.diff.txt uploaded by tilghman (license 14) Tested by: awk, timeshell ........ 2009-07-06 13:40 +0000 [r204950] Kevin P. Fleming * main/channel.c, /: Merged revisions 204948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 | kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 lines Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This change allows applications that request T.38 negotiation on a channel that does not support it to get the proper indication that it is not supported, rather than thinking that negotiation was started when it was not. ........ 2009-07-02 22:05 +0000 [r204837] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ ................ 2009-07-02 16:28 +0000 [r204736] David Vossel * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c: Merged revisions 204710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ ................ 2009-06-30 21:30 +0000 [r204612] Tilghman Lesher * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar. (closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-06-30 18:52 +0000 [r204477] Jason Parker * /, main/say.c: Merged revisions 204475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | 9 lines Merged revisions 204474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing. ........ ................ 2009-06-30 18:44 +0000 [r204472] Tilghman Lesher * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge of revisions 204470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines Recorded merge of revisions 204469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines "tw" is the language specification for Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier ........ ................ 2009-06-29 22:53 +0000 [r204249-204303] Mark Michelson * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines Merged revisions 204300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ ................ * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines Fix build oops. ........ ................ 2009-06-27 01:18 +0000 [r203918] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines The ISDN CPE side should not exclusively pick B channels normally. Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni ........ ................ 2009-06-26 22:13 +0000 [r203856] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500 (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo channel after dahdi restart (closes issue #14477) Reported by: timking ........ ................ 2009-06-26 21:26 +0000 [r203781-203823] Russell Bryant * /, main/file.c: Merged revisions 203802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines Merged revisions 203785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines Don't fast forward past the end of a message. This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj ........ ................ * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 | russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines Ensure the TCP read buffer is fully initialized before handling each packet. (closes issue #14452) Reported by: umberto71 ........ 2009-06-26 20:18 +0000 [r203727] David Brooks * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines Fixing voicemail's error in checking max silence vs min message length Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented as seconds. Also, the inequality was reversed. The warning, if triggered, was "Max silence should be less than minmessage or you may get empty messages", which should have been logged if max silence was greater than minmessage, but the check was for less than. Also, conforming if statement to coding guidelines. closes issue #15331) Reported by: markd Review: https://reviewboard.asterisk.org/r/293/ ........ 2009-06-26 19:56 +0000 [r203718] Jeff Peeler * channels/chan_dahdi.c: reverse whitespace change 203713 that was based on looking at sig_analog (which has about a 1000 line indentation change that is not worth doing here) 2009-06-26 19:48 +0000 [r203714] David Vossel * channels/chan_iax2.c, /: Merged revisions 203710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009) | 7 lines moving debug message from level 0 to 1. (closes issue #15404) Reported by: leobrown Patches: iax_codec_debug.patch uploaded by leobrown (license 541) ........ 2009-06-26 19:48 +0000 [r203713] Jeff Peeler * channels/chan_dahdi.c: whitespace fix 2009-06-26 19:37 +0000 [r203704] Russell Bryant * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c: Merged revisions 203702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 | russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines Make invalid hints report Unavailable instead of Idle. (closes issue #14413) Reported by: pj ........ 2009-06-26 19:31 +0000 [r203703] Joshua Colp * include/asterisk/frame.h, main/rtp.c, main/channel.c, main/frame.c, /, channels/chan_sip.c, apps/app_fax.c, configs/sip.conf.sample: Merged revisions 203699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ 2009-06-26 19:28 +0000 [r203700] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) | 16 lines Check if polarityonanswerdelay has elapsed before setting a channel as answered after a polarity reversal. Previously on a polarity switch event chan_dahdi would set the channel immediately as answered. This would cause problems if a polarity reversal occurred when the line was picked up as the dial would not have yet occurred. Now if the polarity reversal occurs before delay has elapsed after coming off hook or an answer, it is ignored. Also, some refactoring was done in _handle_event. (closes issue #13917) Reported by: alecdavis Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ 2009-06-25 21:46 +0000 [r203446] David Vossel * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25 Jun 2009) | 4 lines fixes a few redundant conditions (issue #15269) ........ 2009-06-25 21:19 +0000 [r203393] Terry Wilson * main/cli.c, /: Merged revisions 203381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009) | 11 lines Merged revisions 203380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines I didn't see that Mark already fixed the underlying issue! Yay for removing useless code. ........ ................ 2009-06-25 21:07 +0000 [r203378] Russell Bryant * /, main/features.c: Merged revisions 203376 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines Merged revisions 203375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines Fix a case where CDR answer time could be before the start time involving parking. (closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks ........ ................ 2009-06-25 19:27 +0000 [r203274] Jason Parker * channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) | 10 lines Unmute when we get a dtmfup (we muted on dtmfdown) event. This would occasionally cause one-way audio when using hardware DTMF detection. (closes issue #14761) Reported by: tzafrir Patches: v1-14761.patch uploaded by dimas (license 88) Tested by: tzafrir, dimas ........ 2009-06-25 16:07 +0000 [r203118] Russell Bryant * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) | 18 lines Merged revisions 203115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines Resolve a crash related to a T.38 reinvite race condition. This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) ........ ................ 2009-06-24 21:22 +0000 [r203057] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500 (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. ........ ................ 2009-06-24 18:30 +0000 [r202969] Mark Michelson * /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun 2009) | 9 lines Merged revisions 202966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding the same thing in-line. ........ ................ 2009-06-24 18:10 +0000 [r202927] Joshua Colp * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 | file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines Ensure the default settings are applied for T.38 when we set it up for a peer. ........ 2009-06-23 22:11 +0000 [r202764] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) | 1 line I could have sworn I committed this patch ages ago, but... bug fix with setting NAI properly on linksets in certain situations. ........ 2009-06-23 16:34 +0000 [r202674] David Vossel * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009) | 18 lines Merged revisions 202671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport (closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ ........ ................ 2009-06-22 20:18 +0000 [r202503] Russell Bryant * main/channel.c, /: Merged revisions 202497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009) | 11 lines Merged revisions 202496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines Report CallerID change during a masquerade. Reported by: markster ........ ................ 2009-06-22 16:31 +0000 [r202472] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid potential crashes during reload. Pointed out by Russell while working on the CEL branch. ........ 2009-06-22 16:14 +0000 [r202418] Russell Bryant * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) | 9 lines Merged revisions 202414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) | 2 lines Make Polycom subscription type override check more explicit. ........ ................ 2009-06-22 15:41 +0000 [r202412] David Vossel * main/loader.c, /, include/asterisk/module.h: Merged revisions 202410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 | dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines attempting to load running modules Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice. ........ 2009-06-22 15:10 +0000 [r202339-202345] Mark Michelson * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines Fix a situation in which Asterisk would not stop retransmitting 487s. If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line left from previous commit. ........ ................ * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) | 31 lines Merged revisions 202336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines Fix a possible infinite loop in SDP parsing during glare situation. There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon ........ ................ 2009-06-21 16:15 +0000 [r202260-202264] Russell Bryant * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 | russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines Fix possibility of crashiness during reload in custom fields handling. ........ * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 | russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines Standardize return values of load_config() so reload() doesn't report an error on success. ........ 2009-06-20 19:14 +0000 [r202185] Sean Bright * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 | seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5 lines Fix version detection for API changes in spandsp. (closes issue #15355) Reported by: deuffy ........ 2009-06-19 21:08 +0000 [r202008] Matthew Nicholson * channels/chan_sip.c: Added deadlock protection to try_suggested_sip_codec in chan_sip.c. Review: https://reviewboard.asterisk.org/r/287/ 2009-06-19 20:26 +0000 [r201996] David Vossel * channels/chan_iax2.c, /: Merged revisions 201994 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500 (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) | 8 lines timestamp was being converted to host order as a short rather than a long (closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) ........ ................ 2009-06-19 15:48 +0000 [r201784-201905] Tilghman Lesher * res/res_config_odbc.c, /: Merged revisions 201904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009) | 4 lines Fix 2 typos and add support for wide character types. Reported by Benny Amorsen via the asterisk-users mailing list. http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html ........ * main/features.c: If the "h" extension fails, give it another chance in main/pbx.c. If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. * /, apps/Makefile: Merged revisions 201783 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 | tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines One of the changes in 1.6.1 was to allow app_directory to use functionality within app_voicemail for directory functions. It is therefore no longer necessary for app_directory to be linked against the ODBC libraries (and it never was necessary for app_directory to be linked against IMAP, though it was). ........ 2009-06-18 16:51 +0000 [r201680] David Vossel * channels/misdn/isdn_lib.c, utils/conf2ael.c, main/ast_expr2.c, utils/stereorize.c, main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c, utils/extconf.c, channels/xpmr/xpmr.c, pbx/pbx_config.c, res/res_config_ldap.c, apps/app_rpt.c, main/asterisk.c, codecs/gsm/src/gsm_destroy.c, /, channels/h323/ast_h323.cxx: Merged revisions 201678 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel ........ 2009-06-18 15:36 +0000 [r201613] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 201610 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201610 | russell | 2009-06-18 10:27:10 -0500 (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines Fix memory corruption and leakage related reloads of non files mode MoH classes. For Music on Hold classes that are not files mode, meaning that we are executing an application that will feed us audio data, we use a thread to monitor the external application and read audio from it. This thread also makes use of the MoH class object. In the MoH class destructor, we used pthread_cancel() to ask the thread to exit. Unfortunately, the code did not wait to ensure that the thread actually went away. What needed to be done is a pthread_join() to ensure that the thread fully cleans up before we proceed. By adding this one line, we resolve two significant problems: 1) Since the thread was never joined, it never fully goes away. So, on every reload of non-files mode MoH, an unused thread was sticking around. 2) There was a race condition here where the application monitoring thread could still try to access the MoH class, even though the thread executing the MoH reload has already destroyed it. (issue #15109) Reported by: jvandal (issue #15123) Reported by: axisinternet (issue #15195) Reported by: amorsen (issue AST-208) ........ ................ 2009-06-18 15:24 +0000 [r201601] David Vossel * /, channels/chan_sip.c: Merged revisions 201570 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 | dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines parsing extension correctly from sip register lines If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'. (closes issue #15111) Reported by: ffs Patches: chan_sip.c_register-parser.patch uploaded by ffs (license 730) Tested by: ffs, dvossel ........ 2009-06-17 21:32 +0000 [r201532] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 201531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009) | 7 lines Initialize additional variables, to prevent a possible crash. (closes issue #15186) Reported by: ajohnson Patches: 20090528__issue15186.diff.txt uploaded by tilghman (license 14) Tested by: ajohnson ........ 2009-06-17 20:11 +0000 [r201460-201464] Mark Michelson * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines Fix problem with no audio due to ignoring the SDP. A recent change to our SDP version comparison made audio not function on some calls. This was because of a test wherein we were trying to see if an unsigned value was less than 0. This is a dumb comparison and arguably the compiler should have warned about it. Alas, though, it slipped past. Now it's fixed by changing the variable to be a signed type. Found by several developers. Tested by mnicholson and dbrooks. ........ * main/channel.c, /: Merged revisions 201458 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. ........ ................ 2009-06-17 20:01 +0000 [r201448-201456] David Vossel * doc/datastores.txt, /: Merged revisions 201453 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 | dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines ast_channel_datastore_alloc is no longer used. updating datastores.txt to reflect that. ........ * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines StopMixMonitor race condition (not giving up file immediately) StopMixMonitor only indicates to the MixMonitor thread to stop writing to the file. It does not guarantee that the recording's file handle is available to the dialplan immediately after execution. This results in a race condition. To resolve this, the filestream pointer is placed in a datastore on the channel. When StopMixMonitor is called, the datastore is retrieved from the channel and the filestream is closed immediately before returning to the dialplan. Documentation indicating the use of StopMixMonitor to free files has been updated as well. (closes issue #15259) Reported by: travisghansen Tested by: dvossel Review: https://reviewboard.asterisk.org/r/283/ ........ ................ 2009-06-17 19:39 +0000 [r201444] David Brooks * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) | 16 lines Merged revisions 201380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks ........ ................ 2009-06-17 15:32 +0000 [r201365] David Vossel * /, channels/chan_sip.c: Merged revisions 201344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 | dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines SIP registry ref count error During a sip reload, the list of sip_registry objects are supposed to be traversed, unlinked, and destroyed, but destruction never takes place due to a ref counting error. This causes a memory leak when registry items are removed from sip.conf and reloaded. While the registries are removed from the global list, they are not removed from the scheduler. Because of this, SIP register attempts continue to be sent out for the item even though it may no longer be in the .conf. (closes issue #15295) Reported by: amorsen Review: https://reviewboard.asterisk.org/r/282/ ........ 2009-06-17 12:05 +0000 [r201264] Kevin P. Fleming * /, include/asterisk/linkedlists.h: Merged revisions 201262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is *not*, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ ................ 2009-06-16 22:31 +0000 [r201225] David Vossel * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 | dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines fix issue with build_contact introduced by the "SIP trasnport type issues" commit ........ 2009-06-16 19:42 +0000 [r200989-201096] Kevin P. Fleming * include/asterisk/frame.h, apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c, main/autoservice.c, main/frame.c, /, apps/app_meetme.c, main/slinfactory.c, include/asterisk/linkedlists.h, main/file.c, include/asterisk/channel.h: Merged revisions 201056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ ................ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged revisions 201090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler attribute checking. Defaulting to 'static' for the function scope was bad... so remove it. ........ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged revisions 200985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 | kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7 lines Fix problems with new compiler attribute checking in configure script. The last changes to ast_gcc_attribute.m4 caused some problems checking for various attributes, because the scope of the symbol the attribute is applied to can be important; this patch allows the scope to be specified for the check. ........ 2009-06-16 16:34 +0000 [r200987] David Vossel * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines SIP transport type issues What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ ........ 2009-06-16 16:04 +0000 [r200947] Michiel van Baak * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009) | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail can only use one storage module at the moment. Because it's unclear that selecting one of the storage modules in menuselect will disable filesystem storage we now have a FILE_STORAGE option that conflicts with the other modules. (closes issue #15333) ........ 2009-06-16 01:32 +0000 [r200707-200766] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines Ensure that configure-script testing for compiler attributes actually works. The configure script tests for compiler attributes didn't actually enable enough warnings or provide a proper test harness to determine whether the compiler supports the attribute in question or not; this caused gcc 4.1 to report that it supports 'weakref', but it doesn't actually support it in the way that is needed for our optional API mechanism. The new configure script test will properly distinguish between full support and partial support for this attribute, among others. ........ * CHANGES, /: Merged revisions 200726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 | kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 lines Document the new automatic 'ignoresdpversion' behavior. Asterisk will now automatically ignore incorrect incoming SDP version numbers when necessary to complete a T.38 re-INVITE operation. ........ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 165180,200689 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ 2009-06-15 15:23 +0000 [r200516] Mark Michelson * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) | 11 lines Merged revisions 200513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ ................ 2009-06-12 19:08 +0000 [r200363] Mark Michelson * main/channel.c, /: Merged revisions 200361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines Merged revisions 200360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins ........ ................ 2009-06-11 22:44 +0000 [r200229] Sean Bright * Makefile, /: Merged revisions 199781 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 | seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 lines Fix all of the parallel build warnings issued when running make -j#. ........ 2009-06-11 21:25 +0000 [r200171] Terry Wilson * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null 2009-06-11 21:18 +0000 [r200152] Mark Michelson * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 | mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 lines Fix a crash due to a potentially NULL p->options. Thanks to mnicholson for pointing it out. ........ 2009-06-11 12:16 +0000 [r200041] Leif Madsen * build_tools/make_version_h, /, build_tools/make_version_c: Merged revisions 200039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 | lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines Fix path for .flavor and .version (issue #14737) Reported by: davidw Patches: flavor.patch uploaded by davidw (license 780) Tested by: davidw ........ 2009-06-10 20:35 +0000 [r199996] David Brooks * main/pbx.c, /: Fixes the argument order in definition of new_find_extension(). In the definition of new_find_extension(), the arguments 'callerid' and 'label' were swapped. The prototype declaration and all calls to the function are ordered 'callerid' then 'label', but the function itself was ordered 'label' then 'callerid'. (closes issue #15303) Reported by: JimDickenson 2009-06-10 20:18 +0000 [r199963] Mark Michelson * /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 | mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6 lines Only try to use the invite_branch on outgoing INVITEs with auth credentials. I have added a comment to the code to help ease understanding of the logic here as well. ........ 2009-06-10 16:13 +0000 [r199859] Sean Bright * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ ................ 2009-06-09 20:50 +0000 [r199745-199820] David Vossel * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer. (closes issue #15283) Reported by: jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614) Tested by: jthurman, dvossel ........ * main/loader.c, /, res/res_timing_pthread.c, include/asterisk/module.h, res/res_timing_dahdi.c: Merged revisions 199743 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines module load priority This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ ........ 2009-06-08 19:39 +0000 [r199633] Sean Bright * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ ................ 2009-06-08 17:35 +0000 [r199590] Mark Michelson * /, channels/chan_sip.c: Recorded merge of revisions 199588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon, 08 Jun 2009) | 9 lines Fix a deadlock that could occur when setting rtp stats on SIP calls. (closes issue #15143) Reported by: cristiandimache Patches: 15143.patch uploaded by mmichelson (license 60) Tested by: cristiandimache ........ 2009-06-05 21:32 +0000 [r199300] David Vossel * include/asterisk/devicestate.h, /, main/devicestate.c: Merged revisions 199298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ ................ 2009-06-05 13:51 +0000 [r199229] Mark Michelson * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines Correct "dahdi show channels" output when specifying a group. Since a DAHDI channel may belong to multiple groups, we need to use a bitwise and instead of equivalence to determine whether to display the channel information. (closes issue #15248) Reported by: gentian Patches: 15248.patch uploaded by mmichelson (license 60) Tested by: gentian ........ 2009-06-04 19:16 +0000 [r199141] David Vossel * channels/chan_iax2.c, /: Merged revisions 199139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ ................ 2009-06-04 14:53 +0000 [r199053] Sean Bright * main/asterisk.c, main/loader.c, /, include/asterisk/_private.h: Merged revisions 199051 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ ................ 2009-06-03 15:26 +0000 [r198826-198887] David Vossel * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 198856 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ ........ * channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines fixes issue with channels not going down after transfer Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop. (closes issue #15216) Reported by: oxymoron Tested by: dvossel ........ 2009-06-02 13:50 +0000 [r198793] Joshua Colp * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 198791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ 2009-06-01 18:44 +0000 [r198628] Tilghman Lesher * /, contrib/scripts/meetme.sql: Merged revisions 198626 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 Jun 2009) | 2 lines Add information for new meetme realtime fields ........ 2009-05-31 01:58 +0000 [r198441] Eliel C. Sardanons * /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) | 11 lines Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded. if dahdi_test_timer() fails, timing_funcs_handle remains NULL causing a crash when calling ast_unregister_timing_interface() with a NULL pointer. (closes issue #15234) Reported by: eliel Patches: timing_dahdi1.diff uploaded by eliel (license 64) ........ 2009-05-30 20:21 +0000 [r198373-198390] Sean Bright * res/res_jabber.c, /: Merged revisions 198375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 | seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 lines Properly terminate the receive buffer before sending to iksemel. aji_io_recv takes the maximum number of bytes to read (instead of the total buffer size), so we have to subtract 1 from our buffer size. Without this, when we receive packets that are larger than our buffer, iksemel will choke and things get wonky. (closes issue #15232) Reported by: lp0 Patches: 05302009_res_jabber.c.patch uploaded by seanbright (license 71) Tested by: seanbright, lp0 ........ * res/res_jabber.c, /: Merged revisions 198371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May 2009) | 19 lines Merged revisions 198370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines Properly terminate AMI JabberSend response messages. The response message (either Error or Success) needs an extra trailing \r\n after the fields to inform the client that the message is complete. (closes issue #14876) Reported by: srt Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71) asterisk_14876.patch uploaded by srt (license 378) trunk-14876-2.diff uploaded by phsultan (license 73) ........ ................ 2009-05-30 03:49 +0000 [r198314] Russell Bryant * res/res_smdi.c, /: Merged revisions 198312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) | 12 lines Merged revisions 198311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines Fix a crash that occurred when MWI SMDI messages expired. (closes issue #14561) Reported by: cmoss28 ........ ................ 2009-05-30 03:28 +0000 [r198295] Sean Bright * apps/app_dial.c, /: Merged revisions 198285 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 198251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we treat a missing one. (closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer ........ ................ 2009-05-30 02:34 +0000 [r198249] Joshua Colp * /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 | file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines When removing all packets from a dialog we also need to free the data if present. ........ 2009-05-29 23:05 +0000 [r198147-198187] Russell Bryant * /, configs/modules.conf.sample: Merged revisions 198186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29 May 2009) | 2 lines Suggesting that only a single timing module be loaded is no longer necessary. ........ * /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009) | 2 lines Improve handling of trying to ACK too many timer expirations. ........ * /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009) | 38 lines Resolve issues with choppy sound when using res_timing_pthread. The situation that caused this problem was when continuous mode was being turned on and off while a rate was set for a timing interface. A very easy way to replicate this bug was to do a Playback() from behind a Local channel. In this scenario, a rate gets set on the channel for doing file playback. At the same time, continuous mode gets turned on and off about every 20 ms as frames get queued on to the PBX side channel from the other side of the Local channel. Essentially, this module treated continuous mode and a set rate as mutually exclusive states for the timer to be in. When I dug deep enough, I observed the following pattern: 1) Set timer to tick every 20 ms. 2) Wait almost 20 ms ... 3) Continuous mode gets turned on for a queued up frame 4) Continuous mode gets turned off 5) The timer goes back to its tick per 20 ms. state but starts counting at 0 ms. 6) Goto step 2. Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick, but not most of the time. This is what produced the choppy sound (or sometimes no sound at all). Now, the module treats continuous mode and a set rate as completely independent timer modes. They can be enabled and disabled independently of each other and things work as expected. (closes issue #14412) Reported by: dome Patches: issue14412.diff.txt uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt uploaded by russell (license 2) Tested by: DennisD, russell ........ 2009-05-29 19:13 +0000 [r198074] Matthew Nicholson * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged revisions 198072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ ................ 2009-05-29 18:39 +0000 [r198065] Joshua Colp * /, main/file.c: Merged revisions 198064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 | file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix a memory leak of the write buffer when writing a file. ........ 2009-05-29 18:17 +0000 [r198005] Sean Bright * Makefile, /: Merged revisions 198000 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 197998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May 2009) | 8 lines Fix 'make config' target for Slackware. There was a missing semi-colon after the echo statement in the Makefile that was causing problems for some users. Fix suggested by reporter. (closes issue #15225) Reported by: pdavis ........ ................ 2009-05-29 16:19 +0000 [r197969] Russell Bryant * /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009) | 2 lines Trim trailing whitespace so that I can work on this bug without it bothering me. :-) ........ 2009-05-28 23:59 +0000 [r197897] Leif Madsen * apps/app_mixmonitor.c: Update MixMonitor documentation. Updated the MixMonitor documentation for the 'b' option so that it is more obvious that you must not optimize awat the Local channel when using this option. (issue #14829) 2009-05-28 18:47 +0000 [r197700] Joshua Colp * channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2 lines Fix a bug where the trunkmtu setting was not set to the default value of 1240 on load but was on reload. ........ 2009-05-28 18:26 +0000 [r197696] Eliel C. Sardanons * /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines Merged revisions 197562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines Use the address we already know when reloading a peer with nat=yes. If we already have an address for a peer, and we are reloading the sip configuration, try to use that address to contact the peer, instead of getting it from the Contact. (closes issue #15194) Reported by: ibc Patches: sip.patch uploaded by eliel (license 64) Tested by: manwe ........ ................ 2009-05-28 16:08 +0000 [r197623] David Vossel * channels/chan_iax2.c: 'iax show peer blah' now outputs whether or not peer 'blah' is in trunk mode or not. 2009-05-28 15:39 +0000 [r197545-197618] Mark Michelson * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: Merged revisions 197606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 ........ ................ * apps/app_chanspy.c, /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 197543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ ................ 2009-05-28 14:54 +0000 [r197470-197540] Joshua Colp * /, main/utils.c: Merged revisions 197538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 | file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix a bug in stringfields where it did not actually free the pools of memory. (closes issue #15074) Reported by: pj ........ * /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines Merged revisions 197466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting. The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated (or it passes through unauthenticated) the proper nat flag is set. (closes issue #13823) Reported by: dimas ........ ................ 2009-05-28 11:40 +0000 [r197440] Gavin Henry * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, doc/ldap.txt, configs/res_ldap.conf.sample: issue #15155 and issue #15156 from trunk 2009-05-27 20:11 +0000 [r197262] Sean Bright * Makefile, /: Merged revisions 197260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 | seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6 lines Use bash explicitly when calling build_tools/mkpkgconfig from the Makefile. Since we use bashisms in build_tools/mkpkgconfig, we should call on bash explicitly when running from the Makefile, otherwise we get errors during a 'make install.' (closes issue #15209) Reported by: seandarcy ........ 2009-05-27 19:29 +0000 [r197245] Tilghman Lesher * /, funcs/func_cut.c: Recorded merge of revisions 197209 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500 (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines Use a different determinator on whether to print the delimiter, since leading fields may be blank. (closes issue #15208) Reported by: ramonpeek Patch by me, though inspired in part by a patch from ramonpeek ........ ................ 2009-05-27 17:21 +0000 [r197145] Jeff Peeler * main/channel.c, include/asterisk/channel.h: Fix broken attended transfers The bridge was terminating immediately after the attended transfer was completed. The problem was because upon reentering ast_channel_bridge nexteventts was checked to see if it was set and if so could possibly return AST_BRIDGE_COMPLETE. (closes issue #15183) Reported by: andrebarbosa Tested by: andrebarbosa, tootai, loloski 2009-05-27 16:12 +0000 [r197091] Sean Bright * configs/smdi.conf.sample, configs/extensions.conf.sample, configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /, configs/vpb.conf.sample: Merged revisions 197089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May 2009) | 6 lines Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf in the sample configuration files. (closes issue #15207) Reported by: seandarcy ........ 2009-05-27 15:59 +0000 [r197087] David Vossel * channels/chan_sip.c: Fixes merge issue for r196453. 2009-05-27 13:05 +0000 [r196990] Sean Bright * /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines Display an error message when chan_alsa fails to load due to a missing or inaccessible configuration file. Before this change, when chan_alsa failed to load due to a missing or inaccessible configuration file, no message would be displayed. With this change, when chan_alsa fails to load due to a missing or inaccessible configuration file, a message will be displayed. (closes issue #14760) Reported by: Nick_Lewis Patches: chan_alsa.c-confload.patch uploaded by Nick (license 657) ........ 2009-05-26 22:42 +0000 [r196869-196947] Russell Bryant * /, autoconf/ast_check_osptk.m4 (added), configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 196946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 | russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines Update configure script to check for OSP toolkit 3.5.0. (closes issue #14988) Reported by: tzafrir Patches: configure.ac.diff uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick (license 91) ........ * /, res/res_convert.c: Merged revisions 196843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009) | 16 lines Merged revisions 196826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines Resolve a file handle leak. The frames here should have always been freed. However, out of luck, there was never any memory leaked. However, after file streams became reference counted, this code would leak the file stream for the file being read. (closes issue #15181) Reported by: jkroon ........ ................ 2009-05-26 13:46 +0000 [r196660-196723] Joshua Colp * /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 | file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix a bug where the sip unregister CLI command did not completely unregister the peer. (closes issue #15118) Reported by: alecdavis Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585) ........ * contrib/scripts/safe_asterisk, /: Merged revisions 196658 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue, 26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7 lines Remove some bash specific stuff from safe_asterisk. (closes issue #10812) Reported by: paravoid Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license 46) ........ ................ 2009-05-22 22:35 +0000 [r196453] David Vossel * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 196416 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ 2009-05-22 13:58 +0000 [r196119] Joshua Colp * channels/chan_misdn.c, /: Merged revisions 196117 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri, 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist. (closes issue #12286) Reported by: lmamane ........ ................ 2009-05-21 19:13 +0000 [r195998] David Vossel * channels/chan_iax2.c, /: Merged revisions 195995 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement. (closes issue #15032) Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380) Tested by: guillecabeza (closes issue #14216) Reported by: Andrey Sofronov ........ ................ 2009-05-21 16:19 +0000 [r195892] Matthew Nicholson * main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195882 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ ................ 2009-05-20 23:31 +0000 [r195841] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 195839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195839 | tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines If a variable had a blank value upon the initial setting, then it would do nothing. Identified by Dmitry Andrianov via private email, fixed by me. ........ 2009-05-20 17:34 +0000 [r195638-195705] Joshua Colp * /, main/features.c: Merged revisions 195698 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 lines Fix some code that wrongly assumed a pointer would always be non-NULL when dealing with CDRs after a bridge. (closes issue #15079) Reported by: barryf ........ ................ * /, apps/app_meetme.c: Merged revisions 195636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 lines Fix a bug where the MeetMe option 'D' did not actually prompt for the pin. (closes issue #15050) Reported by: pmhaddad ........ ................ 2009-05-19 20:18 +0000 [r195526] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 195521 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195521 | tilghman | 2009-05-19 15:16:01 -0500 (Tue, 19 May 2009) | 14 lines Merged revisions 195520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009) | 7 lines Ensure thread keys are initialized before attempting to access them. (closes issue #14889) Reported by: jaroth Patches: app_voicemail.c.patch uploaded by msirota (license 758) Tested by: msirota, BlargMaN ........ ................ 2009-05-19 14:47 +0000 [r195451] Joshua Colp * /, channels/chan_sip.c: Merged revisions 195449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) | 14 lines Merged revisions 195448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 lines Fix a bug where direct RTP setup would partially occur even when disabled if the calling channel was answered. (issue #13545) Reported by: davidw (issue #14244) Reported by: mbnwa ........ ................ 2009-05-18 21:31 +0000 [r195429] Eliel C. Sardanons * main/manager.c, /: Merged revisions 195369 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195369 | eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines Fix the CLI command 'manager show command' documentation and functionality. The CLI command 'manager show command' supports passing multiple action names in the same line, but it was not allowing that because of a incorrect check in the argumentes counter. Also the documentation was updated to show that this usage of the command is possible. ........ 2009-05-18 20:54 +0000 [r195358-195372] Tilghman Lesher * apps/app_queue.c, include/asterisk/smdi.h, apps/app_voicemail.c, res/res_smdi.c, /, include/asterisk/monitor.h: Recorded merge of revisions 195370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195370 | tilghman | 2009-05-18 15:52:33 -0500 (Mon, 18 May 2009) | 15 lines Recorded merge of revisions 195366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ ................ * main/asterisk.c, /: Merged revisions 195320 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195320 | tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines Move the spawn of astcanary down, until after the call to daemon(3). This avoids possible conflicts with the internal implementation of daemon(3). (closes issue #15093) Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir ........ 2009-05-18 19:00 +0000 [r195318] Mark Michelson * /, apps/app_externalivr.c: Merged revisions 195316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May 2009) | 18 lines Fix externalivr's setvariable command so that it properly sets multiple variables. The command had a for loop that was guaranteed to only execute once since the continuation operation of the loop would set the input buffer NULL. I rewrote the loop so that its operation was more obvious, and it would set multiple variables correctly. I also reduced stack space required for the function, constified the input string, and modified the function so that it would not modify the input string while I was at it. (closes issue #15114) Reported by: chris-mac Patches: 15114.patch uploaded by mmichelson (license 60) Tested by: chris-mac ........ 2009-05-18 15:55 +0000 [r195209] Joshua Colp * main/frame.c, /: Merged revisions 195207 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195207 | file | 2009-05-18 12:53:26 -0300 (Mon, 18 May 2009) | 14 lines Merged revisions 195206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 lines Fix a typo which caused loss of audio when using G729 in some scenarios with a smoother present. (closes issue #15105) Reported by: bamby Patches: process-vad-correctly.diff uploaded by bamby (license 430) ........ ................ 2009-05-18 15:13 +0000 [r195167] Eliel C. Sardanons * apps/app_dial.c, main/pbx.c, /, apps/app_macro.c: Merged revisions 195162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195162 | eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines Warn about the use of the application WaitExten() within a Macro(). Update applications documentation to warn the user about the use of the WaitExten() application within a Macro(). Recommend the use of Read() instead. (closes issue #14444) Reported by: ewieling ........ 2009-05-18 13:58 +0000 [r195091-195098] Joshua Colp * main/rtp.c, /: Merged revisions 195096 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) | 12 lines Merged revisions 195095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited. (closes issue #13569) Reported by: bkw918 ........ ................ * /, channels/chan_sip.c: Merged revisions 195089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 | file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where specifying an empty outboundproxy would cause packets to get sent to ourself. (closes issue #15106) Reported by: timeshell ........ 2009-05-18 13:07 +0000 [r195023] Russell Bryant * main/manager.c, /: Merged revisions 195021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195021 | russell | 2009-05-18 07:59:11 -0500 (Mon, 18 May 2009) | 12 lines Recorded merge of revisions 195020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009) | 5 lines Don't try to unlock a bogus channel. (closes issue #15144) Reported by: cristiandimache ........ ................ 2009-05-15 22:46 +0000 [r194835-194876] David Vossel * channels/chan_iax2.c, /: Merged revisions 194874 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194874 | dvossel | 2009-05-15 17:44:44 -0500 (Fri, 15 May 2009) | 23 lines Merged revisions 194873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001) (closes issue #14867) Reported by: aragon Tested by: dvossel (closes issue #14717) Reported by: mobeck Patches: regauth_loop_update_patch.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ * channels/chan_iax2.c, channels/iax2-parser.c, channels/iax2-parser.h, /, channels/iax2.h: Merged revisions 194833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009) | 24 lines Merged revisions 194557,194685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away. (closes issue #14207) Reported by: clive18 Review: https://reviewboard.asterisk.org/r/246/ ........ r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines Update to previous IAX2 "Ghost" Channels patch. Fixed some comments made on reviewboard for the previous patch. (issue #14207) ........ ................ 2009-05-15 18:44 +0000 [r194716-194767] Russell Bryant * configs/logger.conf.sample, /: Merged revisions 194765 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194765 | russell | 2009-05-15 13:43:42 -0500 (Fri, 15 May 2009) | 10 lines Merged revisions 194764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) | 2 lines Fix some spelling fail. ........ ................ * /, codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Merged revisions 194722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194722 | russell | 2009-05-15 12:59:08 -0500 (Fri, 15 May 2009) | 4 lines Shuttle some bits around to address some gain issues with G.722. (closes AST-209) ........ * codecs/Makefile, codecs/g722/Makefile (removed), /: Merged revisions 194718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194718 | russell | 2009-05-15 12:37:12 -0500 (Fri, 15 May 2009) | 2 lines Further simplify codec_g722 build. ........ * codecs/Makefile, /: Merged revisions 194714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194714 | russell | 2009-05-15 12:24:39 -0500 (Fri, 15 May 2009) | 2 lines Actually force running make for g722. ........ 2009-05-14 22:30 +0000 [r194542] Kevin P. Fleming * /: Merged revisions 194520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194520 | kpfleming | 2009-05-14 17:26:02 -0500 (Thu, 14 May 2009) | 9 lines Merged revisions 194509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May 2009) | 1 line Update URL to Reviewboard ........ ................ 2009-05-14 22:23 +0000 [r194507] Mark Michelson * /, channels/chan_sip.c: Merged revisions 194496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May 2009) | 30 lines Merged revisions 194484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines Fix a race condition where a reinvite could trigger a 482 response. The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ 2009-05-14 17:07 +0000 [r194436] Joshua Colp * /, apps/app_meetme.c: Merged revisions 194434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194434 | file | 2009-05-14 14:05:33 -0300 (Thu, 14 May 2009) | 7 lines Fix a bug where the 'T' option to Meetme did not work. (closes issue #15031) Reported by: Stochastic (closes issue #13801) Reported by: justdave ........ 2009-05-13 13:41 +0000 [r194212] Joshua Colp * main/rtp.c, /: Merged revisions 194209 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) | 18 lines Merged revisions 194208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. (closes issue #14815) Reported by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue #14460) Reported by: moliveras Tested by: moliveras ........ ................ 2009-05-13 00:54 +0000 [r194140] Tilghman Lesher * main/pbx.c, /: Merged revisions 194138 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194138 | tilghman | 2009-05-12 19:52:49 -0500 (Tue, 12 May 2009) | 14 lines Merged revisions 194137 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009) | 7 lines Fix logic for how to proceed with a single digit extension. (closes issue #15091) Reported by: andrew Patches: 20090512__issue15091.diff.txt uploaded by tilghman (license 14) Tested by: andrew ........ ................ 2009-05-12 23:01 +0000 [r194062] Matthew Nicholson * apps/app_queue.c, /: Merged revisions 194057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194057 | mnicholson | 2009-05-12 17:32:13 -0500 (Tue, 12 May 2009) | 22 lines Merged revisions 194028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May 2009) | 16 lines This change modifies app_queue to properly generate CDR records in failure situations. This involves setting a proper cdr disposition coresponding to the given failure condition and ensuring the proper information is stored in the cdr record. (closes issue #13691) Reported by: dferrer Tested by: mnicholson (closes issue #13637) Reported by: atis Tested by: atis ........ ................ 2009-05-12 20:51 +0000 [r193961] Mark Michelson * /, channels/chan_sip.c: Merged revisions 193954 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 | mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 lines Update spiral support in trunk and 1.6.X to match what is in 1.4. In 1.4, a SIP spiral is treated the same way as a call forward. This works much better than what is currently in trunk and 1.6.X. The code in trunk and 1.6.X did not create a new call to the recipient of the spiral, instead trying to continue the same call. In addition to just being plain wrong, this also had the side effect of only being able to spiral calls to other SIP channels. With this in place, as long as call forwards are honored, SIP spirals will work properly. This means that it will work for outbound calls made by the Queue, Dial, and Page applications. For originated calls and spool calls, however, the spiral will not work properly until a generic call forward mechanism is introduced into Asterisk. (relates to issue #13630) ........ 2009-05-12 20:42 +0000 [r193822-193958] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 193956 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193956 | tilghman | 2009-05-12 15:40:22 -0500 (Tue, 12 May 2009) | 13 lines Merged revisions 193955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009) | 6 lines Avoid initializing routines if the authentication fails. Fixes a crash (RR) issue. (closes issue #14508) Reported by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license 377) ........ ................ * apps/app_voicemail.c, /: Merged revisions 193870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193870 | tilghman | 2009-05-12 12:29:33 -0500 (Tue, 12 May 2009) | 2 lines Convert a THREADSTORAGE object into a simple malloc'd object (as suggested by Russell on -dev) ........ * apps/app_voicemail.c, /: Recorded merge of revisions 193756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193756 | tilghman | 2009-05-11 17:50:47 -0500 (Mon, 11 May 2009) | 25 lines Recorded merge of revisions 193755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009) | 18 lines Move 300 bytes around on the stack, to make more room for an extension buffer. This allows more concurrent extensions to be copied for a single voicemail, without creating a possibility of upsetting existing users, where a dialplan could run out of stack space where it had run fine before. Alternatively, we could have allocated off the heap, but that is a larger change and would have increased the chance for instability introduced by this change. This is really solved starting in 1.6.0.11, as the use of an ast_str buffer allows an unlimited number of extensions (up to available memory). We additionally create a new warning message when the buffer length is exceeded, permitting administrators to see an issue after the fact, whereas previously the list was silently truncated. (closes issue #14739) Reported by: p_lindheimer Patches: 20090417__bug14739.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer ........ ................ 2009-05-11 19:16 +0000 [r193616] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 193614 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500 (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ ................ 2009-05-11 18:07 +0000 [r193547] Leif Madsen * /, funcs/func_channel.c: Recorded merge of revisions 193545 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193545 | lmadsen | 2009-05-11 14:01:44 -0400 (Mon, 11 May 2009) | 14 lines Recorded merge of revisions 193544 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009) | 7 lines Document CHANNEL(transfercapability) in CLI documentation. (issue #15073) Reported by: pkempgen Patches: 20090511__issue15073.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-05-08 20:51 +0000 [r193389] David Vossel * /, channels/chan_sip.c: Merged revisions 193387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 | dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines TCP not matching valid peer. find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it. Review: http://reviewboard.digium.com/r/236/ ........ 2009-05-08 15:36 +0000 [r193335] Sean Bright * funcs/func_devstate.c, /: Merged revisions 193274 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193274 | seanbright | 2009-05-08 11:18:40 -0400 (Fri, 08 May 2009) | 2 lines Fix the spelling of UNAVAILABLE in func_devstate CLI completion. ........ 2009-05-08 14:54 +0000 [r193265] David Vossel * channels/misdn_config.c, /: Merged revisions 193263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500 (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines "misdn show config" segfaults asterisk, if no MSN lists (closes issue #14976) Reported by: alecdavis Patches: misdn_config.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, FabienToune ........ ................ 2009-05-08 14:10 +0000 [r193196] Kevin P. Fleming * configs/logger.conf.sample, /, main/logger.c: Merged revisions 193194 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May 2009) | 13 lines Merged revisions 193193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines Make absolute paths for logger channels work properly (Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. ........ ................ 2009-05-07 23:44 +0000 [r193122] Tilghman Lesher * main/pbx.c, /: Merged revisions 193120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009) | 26 lines Merged revisions 193119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) | 19 lines Fix Background within a Macro for FreePBX. If the single digit DTMF is an extension in the specified context, then go there and signal no DTMF. Otherwise, we should exit with that DTMF. If we're in Macro, we'll exit and seek that DTMF as the beginning of an extension in the Macro's calling context. If we're not in Macro, then we'll simply seek that extension in the calling context. Previously, someone complained about the behavior as it related to the interior of a Gosub routine, and the fix (#14011) inadvertently broke FreePBX (#14940). This change should fix both of these situations, but with the possible incompatibility that if a single digit extension does not exist (but a longer extension COULD have matched), it would have previously gone immediately to the "i" extension, but will now need to wait for a timeout. (closes issue #14940) Reported by: p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer ........ ................ 2009-05-07 22:42 +0000 [r193079] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 193077 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500 (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines Give a more helpful message when an incoming call's dialed extension does not match. Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. ........ ................ 2009-05-07 17:52 +0000 [r192935-193007] Tilghman Lesher * /, funcs/func_odbc.c: Merged revisions 193006 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193006 | tilghman | 2009-05-07 12:51:13 -0500 (Thu, 07 May 2009) | 7 lines Second result should not contain data from the first result. (closes issue #15039) Reported by: jims Patches: 20090506__issue15039.diff.txt uploaded by tilghman (license 14) Tested by: jims ........ * channels/chan_unistim.c, /: Merged revisions 192938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009) | 6 lines Send DTMF frame before playing back audio. (closes issue #14858) Reported by: barryf Patches: 20090507__bug14858.diff.txt uploaded by tilghman (license 14) ........ * /, channels/chan_sip.c: Merged revisions 192933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) | 17 lines Merged revisions 192932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines Eliminate repetition of fullcontact during reconstruction. If the fullcontact field appears in both the sippeers and the sipregs table, then during reconstruction of the field, it will otherwise be doubled. (closes issue #14754) Reported by: Alexei Gradinari Patches: 20090506__bug14754.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ ................ 2009-05-06 22:19 +0000 [r192869] Jeff Peeler * /, main/features.c: Merged revisions 192861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009) | 17 lines Merged revisions 192858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines Make ParkedCall application stop execution of the dialplan after hang up Just changed park_exec to always return non-zero. I really wasn't entirely sure at first if this was a bug. Decided it was since it would be surprising when not using ParkedCall in the dialplan to hang up and have dialplan execution continue. (closes issue #14555) Reported by: francesco_r ........ ................ 2009-05-06 17:53 +0000 [r192812] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 190946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) | 1 line Make sure that we do not clear the down flag on the BRI during PTMP link transients. Also refix SS7 audio that the early media patch broke. ........ 2009-05-06 17:39 +0000 [r192636-192809] Joshua Colp * channels/chan_iax2.c, /: Merged revisions 192808 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) | 10 lines Fix a bug where a timer would be created but not acknowledged. This scenario crept up if chan_iax2 was loaded with no configuration file present. It would create a timer and tell it to go at an interval but the thread that normally acknowledges it would not be created because no configuration file was present. The timer will now be closed if no configuration file is present. (closes issue #15014) Reported by: madkins ........ * /, channels/chan_sip.c: Merged revisions 192634 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) | 14 lines Merged revisions 192633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled. (closes issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded by dimas (license 88) ........ ................ 2009-05-05 20:02 +0000 [r192527] Sean Bright * /, static-http/astman.js: Merged revisions 192525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192525 | seanbright | 2009-05-05 15:57:49 -0400 (Tue, 05 May 2009) | 18 lines Merged revisions 192524 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May 2009) | 11 lines Fix Javascript error when using astman.js in Internet Explorer. Internet Explorer (tested with 7.0) does not like trailing commas on constructs like object initializers, so get rid of them to avoid some errors. (closes issue #15026) Reported by: rajnishgiri Patches: bug15026.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ 2009-05-05 18:26 +0000 [r192401-192473] Joshua Colp * /, main/features.c: Merged revisions 192462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) | 15 lines Merged revisions 192454 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines Fix an incorrect assumption that certain values on the channel will always exist when they may not. The CDR code involved with bridges wrongly assumed that the currently executing application and data values will always exist. It is possible for this to be false when call forwarding is involved. (closes issue #14984) Reported by: gincantalupo ........ ................ * apps/app_followme.c, /: Merged revisions 192430 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192430 | file | 2009-05-05 14:46:51 -0300 (Tue, 05 May 2009) | 12 lines Merged revisions 192429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5 lines Fix a bug where the followme application would continue trying numbers after the caller hung up. (closes issue #13624) Reported by: sgenyuk ........ ................ * /, channels/chan_sip.c: Merged revisions 192387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 | file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines Fix a bug with setting t38pt_udptl at the user or peer level. If an incoming call authenticated as a user or peer and t38pt_udptl was not set to yes in general then no UDPTL session would be present and any T38 related things would fail. This commit changes it so that if after authenticating T38 is enabled but no UDPTL session is present one will be created. (issue AST-215) ........ 2009-05-05 13:37 +0000 [r192281-192359] Kevin P. Fleming * main/astobj2.c, include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 192357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 | kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5 lines Correct some flaws in the memory accounting code for stringfields and ao2 objects Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that. ........ * main/astobj2.c, main/datastore.c, main/channel.c, /, include/asterisk/astobj2.h, include/asterisk/datastore.h, include/asterisk/channel.h: Merged revisions 192318 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May 2009) | 5 lines Properly account for memory allocated for channels and datastores As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself. ........ * include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 192279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 | kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5 lines Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function. ........ 2009-05-04 22:48 +0000 [r192216] David Vossel * channels/chan_iax2.c, /: Merged revisions 192214 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192214 | dvossel | 2009-05-04 17:44:51 -0500 (Mon, 04 May 2009) | 17 lines Merged revisions 192213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines global mohinterpret setting is ignored mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers. (closes issue #14728) Reported by: dimas Patches: v1-14728.patch uploaded by dimas (license 88) Tested by: dimas, dvossel ........ ................ 2009-05-04 19:30 +0000 [r192172] Tilghman Lesher * /, configure, res/res_agi.c: Recorded merge of revisions 192171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192171 | tilghman | 2009-05-04 14:29:13 -0500 (Mon, 04 May 2009) | 8 lines Restore 'asyncagi break' command to 1.6.1 and higher. (closes issue #14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt uploaded by tilghman (license 14) 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: nikkk ........ 2009-05-04 19:20 +0000 [r192154] Kevin P. Fleming * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 192059 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192059 | kpfleming | 2009-05-04 18:24:16 +0200 (Mon, 04 May 2009) | 5 lines Ensure that astobj2 memory allocations are properly accounted for when MALLOC_DEBUG is used This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function. ........ 2009-05-04 18:44 +0000 [r192134] Tilghman Lesher * autoconf/ast_ext_tool_check.m4, /: Merged revisions 192132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04 May 2009) | 6 lines Pass libraries in LIBS, not LDFLAGS. (closes issue #14671) Reported by: Chainsaw Patches: asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by Chainsaw (license 723) ........ 2009-05-04 17:30 +0000 [r192094] Leif Madsen * apps/app_forkcdr.c: Resolve grammatical mistakes in the application description in app_forkcdr. (closes issue #14801) Reported by: festr 2009-05-04 10:00 +0000 [r191957] Kevin P. Fleming * /, configs/modules.conf.sample: Merged revisions 191955 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04 May 2009) | 8 lines Ensure that by default only one console channel driver is loaded This configuration file was changed to ensure that only one console channel driver (chan_oss) is loaded by default, but the change would only work if chan_console was not built. Now it will work as expected; if chan_alsa or chan_console are built and installed, they will not be loaded unless explicity requested. ........ 2009-05-02 18:45 +0000 [r191777] Kevin P. Fleming * /, main/logger.c: Merged revisions 191775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191775 | kpfleming | 2009-05-02 20:39:48 +0200 (Sat, 02 May 2009) | 5 lines Fix an error in queue_log file rotation optimization code This code was copy-and-pasted without properly changing references to event_rotate into queue_rotate, so under some conditions the log rotation would rotate queue_log even though it was not necessary. ........ 2009-05-02 15:52 +0000 [r191702] Sean Bright * main/asterisk.c, /: Merged revisions 191700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191700 | seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1 line Update copyright year to 2009 ........ 2009-05-01 20:02 +0000 [r191553-191562] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 191560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009) | 13 lines Merged revisions 191559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1. (closes issue #14993) Reported by: BigJimmy Patches: causepatch uploaded by BigJimmy (license 371) ........ ................ * channels/chan_iax2.c, /: Merged revisions 191494 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009) | 4 lines Set debug message back to DEBUG level. (closes issue #15007) Reported by: hulber ........ 2009-05-01 18:20 +0000 [r191505] Jeff Peeler * main/channel.c, /: Merged revisions 191489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009) | 15 lines Merged revisions 191488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) | 9 lines Fix DTMF not being sent to other side after a partial feature match This fixes a regression from commit 176701. The issue was that ast_generic_bridge never exited after the feature digit timeout had elapsed, which prevented the queued DTMF from being sent to the other side. This issue was reported to me directly. ........ ................ 2009-05-01 16:26 +0000 [r191454] Sean Bright * apps/app_queue.c: Fix a crash in app_queue with very long member lists. A user reported via #asterisk that with very long lists of members, a crash occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead of stack allocating copys of each interface name. (Related to revision 191041 in branches/1.4) 2009-04-30 17:45 +0000 [r191223-191369] Tilghman Lesher * main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 191367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 | tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines Detect eaccess (or euidaccess) before using it. Reported by Andrew Lindh via the -dev list. ........ * main/asterisk.c, /: Merged revisions 191283 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191283 | tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11 lines Change working directory to / under certain conditions. If backgrounding and no core will be produced, then changing the directory won't break anything; likewise, if the CWD isn't accessible by the current user, then a core wasn't possible anyway. (closes issue #14831) Reported by: chris-mac Patches: 20090428__bug14831.diff.txt uploaded by tilghman (license 14) 20090430__bug14831.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac ........ * /, channels/h323/ast_h323.cxx, channels/chan_h323.c: Merged revisions 191219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 | tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines Make H.323 compile with FDLEAK detection code enabled ........ 2009-04-29 18:40 +0000 [r191138] David Brooks * pbx/pbx_config.c, /: Merged revisions 191136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191136 | dbrooks | 2009-04-29 13:32:58 -0500 (Wed, 29 Apr 2009) | 3 lines Removing crufty code that is no longer necessary. Code cleanup. ........ 2009-04-29 08:45 +0000 [r190988] TransNexus OSP Development * apps/app_osplookup.c, /: Merged revisions 190830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190830 | transnexus | 2009-04-28 17:10:42 +0800 (Tue, 28 Apr 2009) | 2 lines Updated for OSP Toolkit 3.5. ........ 2009-04-28 17:33 +0000 [r190906] Tilghman Lesher * doc/tex/cdrdriver.tex, /: Merged revisions 190904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190904 | tilghman | 2009-04-28 12:31:43 -0500 (Tue, 28 Apr 2009) | 2 lines UniqueID column has a maximum size of 150 ........ 2009-04-28 14:13 +0000 [r190731-190863] Kevin P. Fleming * /, Makefile.rules: Merged revisions 190861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190861 | kpfleming | 2009-04-28 09:12:09 -0500 (Tue, 28 Apr 2009) | 5 lines Remove Makefile rules for bison and flex sources We never, ever want these files to processed automatically, because we store the output files in Subversion and users should never need to rebuild them. ........ * /, configure, include/asterisk/autoconfig.h.in: Merged revisions 190725 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr 2009) | 13 lines Merged revisions 190721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines Fix 'inconsistent line endings' when autoconf 2.63 is used Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway. ........ ................ 2009-04-27 19:36 +0000 [r190728] Tilghman Lesher * main/pbx.c, /: Merged revisions 190726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190726 | tilghman | 2009-04-27 14:34:48 -0500 (Mon, 27 Apr 2009) | 4 lines Don't warn on pipe in the System call. (closes issue #14979) Reported by: pj ........ 2009-08-10 Tilghman Lesher * Asterisk 1.6.1.4 released * AST-2009-005 2009-07-27 Leif Madsen * Asterisk 1.6.1.2 released * AST-2009-004 2009-06-05 Leif Madsen * Asterisk 1.6.1.1 released 2009-06-04 David Vossel * channels/chan_iax2.c: Additional updates for AST-2009-001 2009-06-04 David Vossel * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001 2009-04-27 Leif Madsen * Create Asterisk 1.6.1.0 2009-04-20 Leif Madsen * Create Asterisk 1.6.1.0-rc5 2009-04-20 17:08 +0000 [r189352] Joshua Colp * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 | file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines Fix a bug with non-UDP connections that caused dialogs to not get freed. This issue crept up because of a reference count issue on non-UDP based dialogs. The dialog reference count was increased when transmitting a packet reliably but never decreased. This caused the dialog structure to hang around despite being unlinked from the dialogs container. (closes issue #14919) Reported by: vrban ........ 2009-04-20 14:06 +0000 [r189280] Mark Michelson * main/channel.c, /: Merged revisions 189278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr 2009) | 18 lines Merged revisions 189277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr 2009) | 12 lines Move the check for chan->fdno == -1 to after the zombie/hangup check. Many users were finding that their hung up channels were staying up and causing 100% CPU usage. (issue #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch uploaded by mmichelson (license 60) Tested by: falves11, bamby ........ ................ 2009-04-18 01:38 +0000 [r189206] David Vossel * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500 (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening. (closes issue #14091) Reported by: evandro Patches: autologoff.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/225/ ........ ................ 2009-04-17 21:55 +0000 [r189139] Richard Mudgett * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 189137 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009) | 17 lines Merged revisions 188833,189134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines Modifed/added some debug messages. JIRA ABE-1835 ........ ................ 2009-04-17 20:21 +0000 [r189103] Mark Michelson * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines Prevent a crash when SIP blonde transferring an unbridged call. If one attempts to use the attended transfer button on a SIP phone to transfer an unbridged call (such as a call to an IVR) but hangs up while the target of the transfer is still ringing, we need to not crash. The problem was that ast_hangup was called from outside the channel thread. AST-211 ........ 2009-04-17 19:46 +0000 [r189080] Sean Bright * main/asterisk.c, /: Merged revisions 189077 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 | seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1 line Fix copy/paste error with 'transmit silence' flag. ........ 2009-04-17 17:33 +0000 [r189069] Matthew Nicholson * main/pbx.c, /: Merged revisions 189010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr 2009) | 12 lines Merged revisions 189009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr 2009) | 5 lines Make Busy() application set the CDR disposition to BUSY. (closes issue #14306) Reported by: cristiandimache ........ ................ 2009-04-17 14:48 +0000 [r188940-188949] Joshua Colp * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines Merged revisions 188946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines Fix a bug where a value used to create the channel name was bogus. This commit fixes the scenario where an incoming call is authenticated using a peer entry. Previously the channel name was created using either the username setting from the sip.conf entry or the IP address that the call came from. Now the channel name will be created using the peer name itself. This commit will not change the way the channel name is generated for users or friends. (closes issue #14256) Reported by: Nick_Lewis Patches: chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, file ........ ................ * channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri, 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 lines Fix a situation where the DAHDI channel private structure lock was not unlocked when it should have been. (issue AST-210) ........ ................ 2009-04-16 22:05 +0000 [r188776-188838] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009) | 14 lines Merged revisions 188835 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) | 7 lines Only update realtime, if global option rtupdate != false (closes issue #14885) Reported by: deepesh Patches: 20090413__bug14885.diff.txt uploaded by tilghman (license 14) Tested by: deepesh ........ ................ * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500 (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009) | 4 lines Umask should not be exported into global namespace. (closes issue #14912) Reported by: jcapp ........ ................ 2009-04-15 22:12 +0000 [r188649] David Vossel * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500 (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines National prefix inserted even when caller ID not available When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank. (closes issue #13207) Reported by: shawkris Patches: national_prefix.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/220/ ........ ................ 2009-04-15 20:20 +0000 [r188473-188596] Mark Michelson * /, main/file.c: Merged revisions 188585 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr 2009) | 13 lines Merged revisions 188582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr 2009) | 7 lines Update ast_readvideo_callback to match ast_readaudio_callback. This fixes potential refcount errors that may occur on ast_filestreams. AST-208 ........ ................ * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 | mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3 lines Fix a couple of queue member reference leaks. ........ 2009-04-14 17:43 +0000 [r188254-188415] Joshua Colp * main/rtp.c, /: Merged revisions 188413 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 | file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix an incorrect clock rate when sending T140 text. (closes issue #14029) Reported by: epicac ........ * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 | file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix a bug with the change I made yesterday to outbound proxy support. Per discussion with oej on IRC we need the actual IP address, not the outbound proxy IP address, in the sa field. Upon further inspection this should make the behaviour of all other uses of the outbound proxy in the code. ........ 2009-04-14 05:46 +0000 [r188208-188212] Tilghman Lesher * main/pbx.c, /: Merged revisions 188210 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 | tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines As suggested by Russell, warn users when their dialplan arguments contain pipes, but not commas. ........ * /, utils/smsq.c: Merged revisions 188206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 | tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines Application delimiter is ',', not '|'. (closes issue #14881) Reported by: stegro Patches: smsq.patch uploaded by stegro (license 752) ........ 2009-04-13 19:33 +0000 [r188104] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 188102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr 2009) | 5 lines Fix another crash related to cached realtime music on hold. This was another off-by-one problem caused by moh_register. ........ 2009-04-13 16:32 +0000 [r188069] Joshua Colp * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 | file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1. Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will be sending to. This has to be done because the logic that determines what local IP address to use in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address we are sending to. (closes issue #12006) Reported by: mnicholson ........ 2009-04-13 14:20 +0000 [r188038] Mark Michelson * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 | mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6 lines Set all queue variables on both the caller and member channels. This allows for the variables to be accessed if a member macro is run. Thanks to Grigoriy Puzankin for bringing this up on the -dev list. ........ 2009-04-10 20:28 +0000 [r187914] Jeff Peeler * channels/Makefile, /: Merged revisions 187906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 | jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines Fix module embedding for chan_h323. Include libchanh323.a in the modules.link file so that all the symbols can be resolved at link time. (closes issue #11966) Reported by: dome Patches: issue_11966.patch uploaded by kpfleming (license 421) Tested by: jpeeler ........ 2009-04-10 17:30 +0000 [r187767] Tilghman Lesher * contrib/scripts/sip-friends.sql, contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500 (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009) | 2 lines Add lastms column to the contributed table designs ........ ................ 2009-04-10 16:54 +0000 [r187723] Kevin P. Fleming * /, build_tools/embed_modules.xml: Merged revisions 187721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10 Apr 2009) | 5 lines clean up some patterns for files to remove add embedding support for bridge and test modules ........ 2009-04-10 16:03 +0000 [r187678] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 | tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines Ensure pvt is not NULL before dereferencing it. (closes issue #14784) Reported by: pj ........ 2009-04-10 16:00 +0000 [r187676] Russell Bryant * tests/test_heap.c, /: Merged revisions 187675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 | russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines Disable test modules by default. ........ 2009-04-10 03:56 +0000 [r187600] Tilghman Lesher * main/channel.c, main/pbx.c, main/manager.c, /, include/asterisk/linkedlists.h, main/features.c, main/http.c, main/app.c, include/asterisk/lock.h, main/audiohook.c: Merged revisions 187599 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines Modify headers and macros, according to Russell's suggestions on the -dev list ........ 2009-04-09 19:14 +0000 [r187495] Mark Michelson * /, channels/chan_sip.c: Merged revisions 187488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187488 | mmichelson | 2009-04-09 13:58:41 -0500 (Thu, 09 Apr 2009) | 24 lines Merged revisions 187484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr 2009) | 18 lines Handle a SIP race condition (reinvite before an ACK) properly. RFC 5047 explains the proper course of action to take if a reINVITE is received before the ACK from a previous invite transaction. What we are to do is to treat the reINVITE as if it were both an ACK and a reINVITE and process it normally. Later, when we receive the ACK we had been expecting, we will ignore it since its CSeq is less than the current iseqno of the sip_pvt representing this dialog. (closes issue #13849) Reported by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson (license 60) Tested by: mmichelson, klaus3000 ........ ................ 2009-04-09 18:54 +0000 [r187486] Tilghman Lesher * main/manager.c, /, include/asterisk/linkedlists.h, include/asterisk/lock.h: Merged revisions 187483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines Race condition between ast_cli_command() and 'module unload' could cause a deadlock. Add lock timeouts to avoid this potential deadlock. (closes issue #14705) Reported by: jamessan Patches: 20090320__bug14705.diff.txt uploaded by tilghman (license 14) Tested by: jamessan ........ ................ 2009-04-09 17:43 +0000 [r187427] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 187421,187424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu, 09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using cached realtime moh. The moh_register function links an mohclass and then immediately unrefs the class since the container now has a reference. The problem with using realtime music on hold is that the class is allocated, registered, and started in one fell swoop. The refcounting logic resulted in the count being off by one. The same problem did not happen when using a static config because the allocation and registration of an mohclass is a separate operation from starting moh. This also did not affect non-cached realtime moh because the classes are not registered at all. I also have modified res_musiconhold to use the _t_ variants of the ao2_ functions so that more info can be gleaned when attempting to trace the refcounts. I found this to be incredibly helpful for debugging this issue and there's no good reason to remove it. (closes issue #14661) Reported by: sum ........ r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr 2009) | 3 lines Use safe macro practices even though they really aren't necessary. ........ 2009-04-09 17:22 +0000 [r187305-187388] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 | tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines Allow '/' in username portion of register; this is a regression. (closes issue #14668) Reported by: Netview ........ * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions 187363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) | 10 lines Merged revisions 187362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) | 3 lines Permit zero-length text messages in SIP. (Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal") ........ ................ * main/asterisk.c, agi/Makefile, build_tools/cflags.xml, utils/Makefile, include/asterisk.h, /, main/Makefile, main/file.c, main/astfd.c (added): Merged revisions 187302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines Add debugging mode for diagnosing file descriptor leaks. (Related to issue #14625) ........ r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, missed this file in the last commit. ........ ................ 2009-04-08 16:53 +0000 [r186987-187048] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 187046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500 (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines Fix a small logical error when loading moh classes. We were unconditionally incrementing the number of mohclasses registered. However, we should actually only increment if the call to moh_register was successful. While this probably has never caused problems, I noticed it and decided to fix it anyway. ........ ................ * main/channel.c, /: Merged revisions 186985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr 2009) | 30 lines Merged revisions 186984 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines Make a couple of changes with regards to a new message printed in ast_read(). "ast_read() called with no recorded file descriptor" is a new message added after a bug was discovered. Unfortunately, it seems there are a bunch of places that potentially make such calls to ast_read() and trigger this error message to be displayed. This commit does two things to help to make this message appear less. First, the message has been downgraded to a debug level message if dev mode is not enabled. The message means a lot more to developers than it does to end users, and so developers should take an effort to be sure to call ast_read only when a channel is ready to be read from. However, since this doesn't actually cause an error in operation and is not something a user can easily fix, we should not spam their console with these messages. Second, the message has been moved to after the check for any pending masquerades. ast_read() being called with no recorded file descriptor should not interfere with a masquerade taking place. This could be seen as a simple way of resolving issue #14723. However, I still want to try to clear out the existing ways of triggering this message, since I feel that would be a better resolution for the issue. ........ ................ 2009-04-08 05:07 +0000 [r186900] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 | tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines Add lastms to the require API call. ........ 2009-04-08 00:10 +0000 [r186835-186844] Mark Michelson * /, formats/format_wav.c, formats/format_wav_gsm.c: Merged revisions 186842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr 2009) | 14 lines Merged revisions 186841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines Fix a few typos of the word "frequency." (closes issue #14842) Reported by: jvandal Patches: frequency-typo.diff uploaded by jvandal (license 413) ........ ................ * /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 | mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 lines Fix bad merge from fix for issue 13867. (closes issue #14686) Reported by: davidw ........ * main/channel.c, /: Merged revisions 186833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr 2009) | 15 lines Merged revisions 186832 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, warning sounds will not be properly played to either party of the bridge. (closes issue #14845) Reported by: adomjan ........ ................ 2009-04-07 22:33 +0000 [r186806] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 186799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) | 10 lines Merged revisions 186775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines Fix Macro documentation to match current (and intended) behavior. (See -dev mailing list) ........ ................ 2009-04-07 20:53 +0000 [r186722] Mark Michelson * main/manager.c, /: Merged revisions 186720 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr 2009) | 12 lines Merged revisions 186719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines Ensure that \r\n is printed after the ActionID in an OriginateResponse. (closes issue #14847) Reported by: kobaz ........ ................ 2009-04-03 20:21 +0000 [r186466] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later). ........ ................ 2009-04-03 20:04 +0000 [r186448] Tilghman Lesher * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 186444,186447 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) | 14 lines Merged revisions 186415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ ................ r186447 | tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines Merged revisions 186445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) | 2 lines Found a conflict in the last commit, due to multiple targets ........ ................ 2009-04-03 16:38 +0000 [r186381] David Vossel * /, main/audiohook.c: Merged revisions 186379 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 | dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 4 lines audio_audiohook_write_list() did not correctly update sample size after ast_translate. audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) ........ 2009-04-03 Leif Madsen * Asterisk 1.6.1.0-rc4 released. 2009-04-03 15:54 +0000 [r186323] Joshua Colp * include/asterisk/crypto.h, /: Merged revisions 186321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines Fix a problem with the crypto variable definitions not actually being defined properly. (closes issue #14804) Reported by: jvandal ........ ................ 2009-04-03 14:33 +0000 [r186288] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr 2009) | 20 lines Fix the ability to retrieve voicemail messages from IMAP. A recent change made interactive vm_states no longer get added to the list of vm_states and instead get stored in thread-local storage. In trunk and all the 1.6.X branches, the problem is that when we search for messages in a voicemail box, we would attempt to update the appropriate vm_state struct by directly searching in the list of vm_states instead of using the get_vm_state_by_imap_user function. This meant we could not find the interactive vm_state that we wanted. (closes issue #14685) Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........ 2009-04-03 02:06 +0000 [r186232] Russell Bryant * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009) | 29 lines Merged revisions 186229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) | 21 lines Fix a memory leak in cdr_radius. I came across this while doing some testing of my ast_channel_ao2 branch. After running a test overnight that generated over 5 million calls, Asterisk had taken up about 1 GB of my system memory. So, I re-ran the test with MALLOC_DEBUG turned on. However, it showed no leaks in Asterisk during the test, even though Asterisk was still consuming it somehow. Instead, I turned to valgrind, which when run with --leak-check=full, told me exactly where the leak came from, which was from allocations inside the radiusclient-ng library. This explains why MALLOC_DEBUG did not report it. After a bit of analysis, I found that we were leaking a little bit of memory every time a CDR record was passed to cdr_radius. I don't actually have a radius server set up to receive CDR records. However, I always have my development systems compile and install all modules. In addition to making sure there are not build errors across modules, always loading modules helps find bugs like this, too, so it is strongly recommend for all developers. ........ ................ 2009-04-02 21:59 +0000 [r186177] Mark Michelson * configs/features.conf.sample, /: Merged revisions 186175 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines Fix instructions in one-step parking comment to make more sense. Changed a capital K to a lowercase k. ........ ................ 2009-04-02 17:27 +0000 [r186108] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized ........ ................ 2009-04-02 17:14 +0000 [r186062] Tilghman Lesher * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 186060 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ 2009-04-02 13:53 +0000 [r185956] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior. this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized. ........ ................ 2009-04-01 19:06 +0000 [r185848] David Vossel * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) | 16 lines Merged revisions 185845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491 Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. (closes issue #12013) Reported by: alx Review: http://reviewboard.digium.com/r/213/ ........ ................ 2009-04-01 13:50 +0000 [r185774] Russell Bryant * main/channel.c, /: Merged revisions 185772 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009) | 14 lines Merged revisions 185771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines Fix a case where DTMF could bypass audiohooks. This change fixes a situation where an audiohook that wants DTMF would not actually get it. This is in the code path where we end DTMF digit length emulation while handling a NULL frame. ........ ................ 2009-03-31 22:38 +0000 [r185666] Kevin P. Fleming * utils, /: Merged revisions 185664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 | kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line ignore copied (generated) file ........ 2009-03-31 22:05 +0000 [r185471-185602] Mark Michelson * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar 2009) | 12 lines Merged revisions 185599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar 2009) | 6 lines Fix crash that would occur if an empty member was specified in queues.conf. (closes issue #14796) Reported by: pida ........ ................ * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500 (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar 2009) | 8 lines Fix Russian voicemail intro to say the word "messages" properly. (closes issue #14736) Reported by: chappell Patches: voicemail_no_messages.diff uploaded by chappell (license 8) ........ ................ 2009-03-31 17:48 +0000 [r185427] David Brooks * /, channels/chan_gtalk.c: Merged revisions 185363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: Review: http://reviewboard.digium.com/r/181/ ........ ................ 2009-03-31 14:57 +0000 [r185263] Russell Bryant * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 | russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines Don't free() an astobj2 object. (closes issue #14672) Reported by: makoto ........ 2009-03-31 14:10 +0000 [r185199] Joshua Colp * /, main/audiohook.c: Merged revisions 185197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) | 15 lines Merged revisions 185196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines Fix crash when moving audiohooks between channels. Handle the scenario where we are called to move audiohooks between channels and the source channel does not actually have any on it. (closes issue #14734) Reported by: corruptor ........ ................ 2009-03-30 20:50 +0000 [r185126-185127] Richard Mudgett * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged revisions 185123 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines Make chan_misdn BRI TE side normally defer channel selection to the NT side. Channel allocation collisions are not handled by chan_misdn very well. This patch simply avoids the problem for BRI only. For PRI, allocation collisions are still possible but less likely since there are simply more channels available and each end could use a different allocation strategy. misdn.conf options available: te_choose_channel - Use to force the TE side to allocate channels. method - Specify the channel allocation strategy. (closes issue #13488) Reported by: Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, festr ........ ................ 2009-03-30 16:47 +0000 [r185088] Mark Michelson * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar 2009) | 45 lines Merged revisions 185031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked. (This is copied and pasted from the review request I made for this patch) Asterisk has some odd behavior when queue weights are used. The current logic used when potentially calling a queue member is: If the member we are going to call is part of another queue and _that other queue has any callers in it_ and has a higher weight than the queue we are calling from, then don't try to contact that member. The issue here is what I have marked with underscores. If the higher-weighted queue has any callers in it at all, then the queue member will be unreachable from the lower-weighted queue. This has the potential to be really really bad if using a queue strategy, such as leastrecent or fewestcalls, with the potential to call the same member repeatedly. The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works well for this situation. With this set of changes, the logic used becomes: If the member we are going to call is part of another queue, the other queue has a higher weight than the queue we are calling from, and the higher weight queue has at least as many callers as available members, then do not try to contact the queue member. If the higher weighted queue has fewer callers than available members, then there is no reason to deny the call to this member since the other queue can afford to spare a member. Since the fix involved writing a generic function for determining the number of available members in the queue, I also modified the is_our_turn function to make use of the new num_available_members function to determine if it is our turn to try calling a member. There is one small behavior change. Before writing this patch, if you had autofill disabled, then if you were the head caller in a queue, you would automatically be told that it was your turn to try calling a member. This did not take into account whether there were actually any queue members available to take the call. Now we actually make sure there is at least one member available to take the call if autofill is disabled. (closes issue #13220) Reported by: garychen Review: http://reviewboard.digium.com/r/202/ ........ ................ 2009-03-30 14:41 +0000 [r184950] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines Merged revisions 184947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines Improve our handling of T38 in the initial INVITE from a device. We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ ........ ................ 2009-03-30 13:57 +0000 [r184912] Russell Bryant * channels/h323/Makefile.in, /: Merged revisions 184910 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) | 4 lines Fix build error when chan_h323 is not being built. (reported by cai1982 in #asterisk-dev) ........ 2009-03-29 05:52 +0000 [r184840-184845] Russell Bryant * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009) | 13 lines Merged revisions 184842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines Ensure targs variable is fully initialized. (closes issue #14758) Reported by: tim_ringenbach ........ ................ * channels/Makefile, /: Merged revisions 184838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines Simplify chan_h323 build to not require a second run of "make". (closes issue #14715) Reported by: jthurman Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614) Tested by: tzafrir, russell ........ 2009-03-27 19:17 +0000 [r184765] Kevin P. Fleming * channels/chan_iax2.c, main/timing.c, main/channel.c, /, include/asterisk/timing.h, include/asterisk/channel.h: Merged revisions 184762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines Improve timing interface to remember which provider provided a timer The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ ........ 2009-03-27 18:09 +0000 [r184728] Russell Bryant * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure we use the best RNG available. ........ 2009-03-27 15:54 +0000 [r184675] Joshua Colp * /, res/res_agi.c: Merged revisions 184673 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 | file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix speech structure leak in the AGI speech recognition integration. The AGI dialplan applications did not destroy the speech structure automatically if it was not destroyed by the running AGI script. They will now do this. (issue LUMENVOX-15) ........ 2009-03-27 14:04 +0000 [r184631] Russell Bryant * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /, res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions 184630 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 | russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines Change g_eid to ast_eid_default. ........ 2009-03-27 13:22 +0000 [r184587] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | 16 lines Merged revisions 184565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls. If calls were placed using an IP address or hostname the global nat setting was copied over but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP actions. (closes issue #14546) Reported by: acunningham ........ ................ 2009-03-27 02:25 +0000 [r184513-184547] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009) | 20 lines Fix some issues with rwlock corruption that caused deadlock like symptoms. When dvossel and I were doing some load testing last week, we noticed that we could make Asterisk trunk lock up instantly when we started generating a bunch of calls. The backtraces of locked threads were bizarre, and many were stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a number of places where a backtrace would be loaded into an invalid index of the backtrace array. It's an off by one error, which ends up writing over the rwlock itself. 2) Ensure that in the array of held locks, we NULL out an index once it is not being used so that it's not confusing when analyzing its contents. 3) Remove a bunch of logging referring to an rwlock operating being done with "deep reentrancy". It is normal for _many_ threads to hold a read lock on an rwlock. ........ * /, main/file.c: Merged revisions 184515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 | russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines Don't act surprised if we get a -1 indication. ........ * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26 Mar 2009) | 2 lines Pass more useful information through to lock tracking when DEBUG_THREADS is on. ........ 2009-03-26 22:19 +0000 [r184451] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 184448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar 2009) | 9 lines Merged revisions 184447 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar 2009) | 3 lines use new, improved 8kHz prompts ........ ................ 2009-03-26 21:18 +0000 [r184394] David Vossel * /, apps/app_test.c: Merged revisions 184389 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009) | 14 lines Merged revisions 184388 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF 8 app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent. During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up. (closes issue #12442) Reported by: tzafrir ........ ................ 2009-03-25 22:13 +0000 [r184325-184345] Russell Bryant * /, main/event.c: Merged revisions 184344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 | russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines Remove unneeded AST_LIST_ENTRY() and comment on the purpose of ast_event_ref. ........ * channels/chan_iax2.c, channels/chan_dahdi.c, include/asterisk/event.h, channels/chan_skinny.c, res/ais/evt.c, main/event.c, include/asterisk/strings.h, main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c, channels/chan_unistim.c, include/asterisk/devicestate.h, /, channels/chan_sip.c, main/devicestate.c, include/asterisk/_private.h: Merged revisions 184339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) | 35 lines Improve performance of the ast_event cache functionality. This code comes from svn/asterisk/team/russell/event_performance/. Here is a summary of the changes that have been made, in order of both invasiveness and performance impact, from smallest to largest. 1) Asterisk 1.6.1 introduces some additional logic to be able to handle distributed device state. This functionality comes at a cost. One relatively minor change in this patch is that the extra processing required for distributed device state is now completely bypassed if it's not needed. 2) One of the things that I noticed when profiling this code was that a _lot_ of time was spent doing string comparisons. I changed the way strings are represented in an event to include a hash value at the front. So, before doing a string comparison, we do an integer comparison on the hash. 3) Finally, the code that handles the event cache has been re-written. I tried to do this in a such a way that it had minimal impact on the API. I did have to change one API call, though - ast_event_queue_and_cache(). However, the way it works now is nicer, IMO. Each type of event that can be cached (MWI, device state) has its own hash table and rules for hashing and comparing objects. This by far made the biggest impact on performance. For additional details regarding this code and how it was tested, please see the review request. (closes issue #14738) Reported by: russell Review: http://reviewboard.digium.com/r/205/ ........ * /: add reviewboard:url property. 2009-03-25 19:26 +0000 [r184282] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 | file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix issue with a T38 reinvite being sent even if not configured to do so. If we receive a T38 request negotiate control frame we should only attempt to do so if the option is enabled on the dialog. ........ 2009-03-25 15:12 +0000 [r184223] Eliel C. Sardanons * main/asterisk.c, /: Merged revisions 184220 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) | 19 lines Merged revisions 184188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. When moving the cursor backward and pressing TAB to autocomplete, a NULL is put in the line and we are loosing what we have already wrote after the actual cursor position. (closes issue #14373) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) Tested by: lmadsen ........ ................ 2009-03-25 01:55 +0000 [r184149] Russell Bryant * main/timing.c, utils/Makefile, /, include/asterisk/compat.h: Merged revisions 184147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 | russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines Fix build issues on Mac OSX. (closes issue #14714) Reported by: ygor ........ 2009-03-24 22:42 +0000 [r184081] Mark Michelson * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar 2009) | 15 lines Merged revisions 184078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. ........ ................ 2009-03-24 21:47 +0000 [r184039] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low and =medium The default codec configuration for chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as the codec in some test calls, but that no longer happens after this change. ........ 2009-03-24 15:28 +0000 [r183867-183916] Tilghman Lesher * /, configs/voicemail.conf.sample: Merged revisions 183914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500 (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ ................ * /, main/http.c: Merged revisions 183865 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 | tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines Allow browsers to cache images and other static content. (This is a regression over 1.4) ........ 2009-03-23 18:59 +0000 [r183768] Mark Michelson * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar 2009) | 13 lines Merged revisions 183700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines Fix a memory leak in res_monitor.c The only way that this leak would occur is if Monitor were started using the Manager interface and no File: header were given. Discovered while reviewing the ast_channel_ao2 review request. ........ ................ 2009-03-23 18:12 +0000 [r183703] Leif Madsen * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) | 7 lines Fixes a documentation error introduced during the CLI cleanup at AstriDevCon 2008. (closes issue #14655) Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728) Tested by: lmadsen ........ 2009-03-20 17:08 +0000 [r183563] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183560 | russell | 2009-03-20 12:00:58 -0500 (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines Fix a crash in IAX2 registration handling found during load testing with dvossel. ........ ................ 2009-03-19 20:33 +0000 [r183438] David Vossel * include/asterisk/features.h, apps/app_dial.c, /, main/features.c: Merged revisions 183436 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines Merged revisions 183386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ................ 2009-03-19 19:19 +0000 [r183333] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500 (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines Delay signalling progress until a PRI channel really signals progress. (closes issue #13034) Reported by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by tilghman (license 14) patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ 2009-03-19 18:14 +0000 [r183249] Russell Bryant * main/loader.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 183242 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines Merged revisions 183241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ ................ 2009-03-19 18:11 +0000 [r183246] Mark Michelson * apps/app_queue.c, /: Merged revisions 183244 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 | mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 lines Fix a memory leak associated with queues. For every attempt that app_queue made to place an outbound call to a queue member, we would allocate a queue_end_bridge structure. When the bridge for the call had completed, we would free the structure. Unfortunately not all call attempts actually end up bridged to a member, so we need to be more selective of when to allocate the structure. With this change, the allocation occurs in an area where we can guarantee that the call will be bridged. (closes issue #14680) Reported by: caspy Patches: 14680.patch uploaded by mmichelson (license 60) Tested by: caspy ........ 2009-03-19 17:08 +0000 [r183198] David Vossel * include/asterisk/features.h, apps/app_dial.c, /, main/features.c: Merged revisions 183172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ ................ 2009-03-19 16:09 +0000 [r183121] Mark Michelson * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 ........ ................ 2009-03-19 Leif Madsen * Release Asterisk 1.6.1.0-rc3 2009-03-19 15:43 +0000 [r183067-183110] Joshua Colp * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ ........ * main/channel.c, /: Merged revisions 183057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. ........ 2009-03-18 21:19 +0000 [r183030] Jeff Peeler * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ 2009-03-18 14:32 +0000 [r182946] Russell Bryant * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c, configure, apps/app_mp3.c, res/res_agi.c, include/asterisk/poll-compat.h, channels/chan_alsa.c, main/asterisk.c, apps/app_nbscat.c, /, main/Makefile, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/io.h, main/utils.c, include/asterisk/channel.h: Merged revisions 182847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ 2009-03-17 20:52 +0000 [r182724] Jeff Peeler * channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /, channels/h323/ast_h323.cxx, configure, autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h, channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions 182722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ 2009-03-17 15:31 +0000 [r182570] Russell Bryant * main/channel.c, /: Merged revisions 182553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). ........ 2009-03-17 15:00 +0000 [r182527-182533] Kevin P. Fleming * main/channel.c, /: Merged revisions 182530 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 lines correct logic flaw in ast_answer() changes in r182525 ........ * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 182525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ 2009-03-17 05:54 +0000 [r182452] Tilghman Lesher * main/db.c, /: Merged revisions 182450 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) | 14 lines Merged revisions 182449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ ................ 2009-03-16 17:53 +0000 [r182284] David Vossel * channels/chan_iax2.c, /: Merged revisions 182282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ ................ 2009-03-16 17:38 +0000 [r182280] Tilghman Lesher * channels/chan_local.c, /, funcs/func_env.c: Merged revisions 182211,182278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009) | 14 lines Merged revisions 182208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ r182278 | tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution. Previously, FILE() returned one less character than specified, due to the terminating NULL. Both the offset and length parameters now behave identically to the way variable substitution offsets and lengths also work. (closes issue #14670) Reported by: BMC ................ 2009-03-16 14:00 +0000 [r182173] Joshua Colp * main/channel.c, /: Merged revisions 182171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 | file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix a memory leak in the ast_answer / __ast_answer API call. For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. ........ 2009-03-13 21:27 +0000 [r182068-182123] Mark Michelson * apps/app_queue.c, /: Merged revisions 182121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 | mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 lines Change faulty comparison used when announcing average hold minutes and seconds (closes issue #14227) Reported by: caspy ........ * /, main/features.c: Merged revisions 182029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar 2009) | 41 lines Merged revisions 181990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ ................ 2009-03-13 17:29 +0000 [r182042] Joshua Colp * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) ........ 2009-03-13 16:58 +0000 [r181987] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 181985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181985 | kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line improve a bit of suboptimal code ........ 2009-03-12 21:45 +0000 [r181771-181849] Mark Michelson * apps/app_queue.c, /: Merged revisions 181846 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 | mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 lines Run the macro on the queue member's channel when he answers, not the caller's channel. ........ * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ ................ 2009-03-12 18:07 +0000 [r181733] Tilghman Lesher * /, main/translate.c: Merged revisions 181731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines Adjust translation table column widths based upon the translation times. Previously, only 5 columns were displayed, and if a translation time exceeded 99,999 useconds, it would be displayed as 0, instead of its actual time. (closes issue #14532) Reported by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman (license 14) Tested by: pj ........ 2009-03-12 16:58 +0000 [r181614-181667] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 lines Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. ........ ................ * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in previous commit. ........ ................ * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy ........ ................ * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix crash when sleep and retries argument was not given to RetryDial application. (closes issue #14647) Reported by: sherpya ........ 2009-03-12 01:05 +0000 [r181544] Richard Mudgett * /, build_tools/make_version: Merged revisions 181542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) | 1 line Use the correct branch integrated property when generating the version string ........ 2009-03-11 23:21 +0000 [r181521] Michiel van Baak * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk 2009-03-11 22:27 +0000 [r181474] Russell Bryant * main/channel.c, /: Merged revisions 181465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181465 | russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGE_PLAY_SOUND variable thread-safe. ........ 2009-03-11 22:23 +0000 [r181457] Jason Parker * /, configure, configure.ac: Merged revisions 181444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181444 | qwell | 2009-03-11 17:20:13 -0500 (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | 4 lines Allow prefix to set localstatedir (when used and different from the default). This is similar to the /etc change that was made for the non-FreeBSD case. ........ ................ 2009-03-11 22:16 +0000 [r181426-181430] Russell Bryant * main/channel.c, /: Merged revisions 181428 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGEPVTCALLID variable thread-safe. ........ * main/channel.c, /: Merged revisions 181424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ ................ 2009-03-11 17:40 +0000 [r181373] David Vossel * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions 181371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ ................ 2009-03-11 17:29 +0000 [r181298-181359] Joshua Colp * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines Merged revisions 181328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ ................ * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines Merged revisions 181295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ ................ 2009-03-11 15:54 +0000 [r181199-181283] Jeff Peeler * channels/h323/ast_h323.cxx: add missing header file * pbx/pbx_config.c, utils/Makefile, include/asterisk/utils.h, include/asterisk/astmm.h, /, channels/chan_sip.c, channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c: Merged revisions 181135 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ 2009-03-11 01:04 +0000 [r181035] Mark Michelson * /, channels/chan_sip.c: Merged revisions 181032-181033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines Remove unused variables. ........ ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC 3891 ................ 2009-03-10 22:07 +0000 [r180947] Jason Parker * /, configure, configure.ac, autoconf/ast_prog_sed.m4, autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180944 | qwell | 2009-03-10 17:03:41 -0500 (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) | 1 line Make things happier when using autoconf 2.62+ ........ ................ 2009-03-10 14:42 +0000 [r180802] Joshua Colp * main/manager.c, /: Merged revisions 180800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines Reset the thread local string buffer when handling the UserEvent action. (closes issue #14593) Reported by: JimDickenson ........ 2009-03-09 21:22 +0000 [r180740] Jeff Peeler * include/asterisk/heap.h, include/asterisk/http.h, include/asterisk/logger.h, main/tcptls.c, include/asterisk/res_odbc.h, include/asterisk/doxyref.h, include/asterisk/event.h, include/asterisk/audiohook.h, include/asterisk/dsp.h, include/asterisk/lock.h, include/asterisk/udptl.h, include/asterisk/dnsmgr.h, include/asterisk/utils.h, include/asterisk/devicestate.h, /, include/asterisk/taskprocessor.h, include/asterisk/astobj2.h, include/asterisk/channel.h, include/asterisk/tcptls.h, include/asterisk/manager.h, main/enum.c, include/asterisk/callerid.h, include/asterisk/app.h, include/asterisk/linkedlists.h, include/asterisk/sched.h, include/asterisk/datastore.h, include/asterisk/timing.h, include/asterisk/dlinkedlists.h, include/asterisk/pbx.h, include/asterisk/enum.h, include/asterisk/config.h, include/asterisk/rtp.h, include/asterisk/extconf.h, main/devicestate.c: Merged revisions 180719 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 Copied from my review board description: This is a continuation of the API changes documentation started for describing changes between releases. Most of the API changes were pretty simple needing only to be brought to attention via the new "Asterisk API Changes" list. However, if you see anything that needs further explanation feel free to supplement what is there. The current method of documenting is to add (in the header file): \version and then to add the function to the change list in doxyref.h on the AstAPIChanges page. I also made sure all the functions that were newly added were tagged with \since 1.6.1. I think this is a good habit to start both for the historical aspect as well as for the future ability to easily add a "New Asterisk API" page. Review: http://reviewboard.digium.com/r/190/ ........ 2009-03-06 18:26 +0000 [r180585] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600 (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when IMAP storage is enabled. ........ ................ 2009-03-06 17:35 +0000 [r180537] David Vossel * main/enum.c, /: Merged revisions 180534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) | 15 lines Merged revisions 180532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ ........ ................ 2009-03-05 23:28 +0000 [r180425-180467] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600 (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 180383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar 2009) | 31 lines Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ ................ 2009-03-05 18:40 +0000 [r180378] Kevin P. Fleming * include/asterisk/frame.h, main/rtp.c, main/frame.c, /: Merged revisions 180373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ ................ 2009-03-04 Leif Madsen * Released Asterisk 1.6.1.0-rc2 2009-03-04 21:09 +0000 [r180263] Russell Bryant * /, channels/chan_sip.c: Merged revisions 180261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180261 | russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines Resolve object matching issues related to the removal of the sip_user object. Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ ........ 2009-03-04 19:27 +0000 [r180122-180197] Joshua Colp * /, main/callerid.c: Merged revisions 180195 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | 11 lines Merged revisions 180194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 lines Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) ........ ................ * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 | file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) ........ 2009-03-03 23:39 +0000 [r180080] David Vossel * main/channel.c, include/asterisk/app.h, apps/app_read.c, /, main/app.c: Merged revisions 180032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ ........ 2009-03-03 23:31 +0000 [r180077] Steve Murphy * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Merged revisions 179973 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | 33 lines Merged revisions 179807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some work to do to port these changes to trunk; the check_expr stuff hasn't been updated here for quite some time, it appears. I added some more tests to the check_expr2 suite. I had to play around with the makefile a bit, etc. I added STANDALONE2 #ifdefs to ast_expr2.y so as not to conflict structure with aelparse. ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. ........ ................ 2009-03-03 22:49 +0000 [r179939-180009] Mark Michelson * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions 180007 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ ................ * doc/timing.txt (added), /, res/res_timing_dahdi.c: Merged revisions 179937 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179937 | mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 lines Add documentation for timing modules used in Asterisk This document specifies the timing modules available in Asterisk beginning with Asterisk 1.6.1. The document goes into detail about the differences between each and gives a general overview of what timing is used for in Asterisk. There is also a section which can be used to help customize your setup or to troubleshoot timing issues you may have. I also added messages to the DAHDI timing test used in res_timing_dahdi.c that points to this new documentation if people experience problems. Big thanks to all who contributed comments on this. (closes issue #14490) Reported by: mmichelson Patches: timing.txt uploaded by mmichelson (license 60) Review: http://reviewboard.digium.com/r/164/ ........ 2009-03-03 20:09 +0000 [r179905] Russell Bryant * /, apps/app_directed_pickup.c: Merged revisions 179903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar 2009) | 1 line fix a leaked channel lock (and future deadlock) when we try to pick up our own channel ........ 2009-03-03 18:30 +0000 [r179843] Joshua Colp * /, main/features.c: Merged revisions 179841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines Merged revisions 179840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ ................ 2009-03-03 16:48 +0000 [r179744] Russell Bryant * main/channel.c, /: Merged revisions 179742 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ ................ 2009-03-03 14:41 +0000 [r179674] Joshua Colp * main/channel.c, /: Merged revisions 179672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines Move where fdno is set to the default value to *after* the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ ................ 2009-03-03 13:56 +0000 [r179611] Russell Bryant * main/channel.c, /: Merged revisions 179609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ ................ 2009-03-03 00:04 +0000 [r179539] Jeff Peeler * main/channel.c, /: Merged revisions 179537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ ................ 2009-03-02 23:39 +0000 [r179535] Russell Bryant * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) | 48 lines Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ ................ 2009-03-02 23:12 +0000 [r179471] Tilghman Lesher * /, main/app.c: Merged revisions 179469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ ................ 2009-03-02 23:04 +0000 [r179464] Russell Bryant * main/channel.c, /: Merged revisions 179462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ ................ 2009-03-02 20:18 +0000 [r179407] Jason Parker * /, main/editline/configure, main/editline/np/unvis.c, main/editline/sys.h, main/editline/configure.in: Merged revisions 179396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | 9 lines Merged revisions 179395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ ................ 2009-03-02 17:19 +0000 [r179362] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179361 | tilghman | 2009-03-02 11:18:48 -0600 (Mon, 02 Mar 2009) | 2 lines Backport 1.6.0 fix to trunk (failsafe if db is not loaded) ........ 2009-03-02 14:14 +0000 [r179293] Joshua Colp * /, main/audiohook.c: Merged revisions 179291 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179291 | file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix issue where changing the volume of both directions of audio did not work. (closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) ........ 2009-03-01 23:28 +0000 [r179221-179256] Mark Michelson * apps/app_speech_utils.c, /: Merged revisions 179254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar 2009) | 5 lines Swap reversed timevals. This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........ * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines Properly free memory and remove scheduler entries when a transmission failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis ........ 2009-02-27 21:48 +0000 [r179166] Russell Bryant * configs/ais.conf.sample, res/res_ais.c, /, doc/distributed_devstate.txt: Merged revisions 179164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 Feb 2009) | 2 lines Mark res_ais as experimental, as the binary event format is subject to change. ........ 2009-02-27 21:34 +0000 [r179163] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) | 3 lines If config file is blank, don't load module. (Closes issue #14563) ........ 2009-02-27 21:25 +0000 [r179160] Russell Bryant * /, UPGRADE.txt: Merged revisions 179154 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines Add a note about the ordering of entries in sip.conf in 1.6.1. ........ 2009-02-27 19:06 +0000 [r179059] Jason Parker * /, doc/tex/channelvariables.tex: Merged revisions 179057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb 2009) | 8 lines Update documentation for DIALEDTIME and ANSWEREDTIME variables. (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) ........ 2009-02-27 03:56 +0000 [r178988] Steve Murphy * configs/features.conf.sample, /, main/features.c: Merged revisions 178986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ ................ 2009-02-26 17:50 +0000 [r178875] David Vossel * channels/chan_iax2.c, /: Merged revisions 178871 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) | 6 lines IAX2 prune realtime, minor tweak to last fix A return statement was missing which caused unexpected cli output. issue #14479 ........ 2009-02-26 17:38 +0000 [r178869] Steve Murphy * /, main/features.c: Merged revisions 178828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines Merged revisions 178804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom ........ ................ 2009-02-26 16:44 +0000 [r178803] Joshua Colp * /, main/file.c: Merged revisions 178801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 | file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed. (closes issue #14541) Reported by: grant ........ 2009-02-26 16:07 +0000 [r178769] David Vossel * channels/chan_iax2.c, /: Merged revisions 178767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ 2009-02-25 12:46 +0000 [r178511] Russell Bryant * main/asterisk.c, /: Merged revisions 178509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) | 10 lines Merged revisions 178508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines Update the copyright year for the main page of the doxygen documentation. ........ ................ 2009-02-24 23:28 +0000 [r178383-178448] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 178446 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600 (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines Add section about the #exec command in configuration files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) ........ ................ * main/asterisk.c, /: Merged revisions 178381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 | tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines Apparently, a void cast doesn't override warn_unused_result. ........ 2009-02-24 20:44 +0000 [r178379-178380] Russell Bryant * Makefile: revert accidental Makefile change. * main/rtp.c, Makefile, /: Merged revisions 178374 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell ........ ................ 2009-02-24 20:41 +0000 [r178305-178377] Tilghman Lesher * main/asterisk.c, /: Merged revisions 178375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 | tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines The 3 possible errors with pipe(2) are all impossible in this situation. ........ * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init. ........ * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 | tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary. Also, add some documentation supporting the use of astcanary. (closes issue #14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545) ........ 2009-02-24 15:22 +0000 [r178232] Joshua Colp * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines Merged revisions 178205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ ................ 2009-02-23 23:22 +0000 [r178172] Russell Bryant * main/rtp.c, /: Merged revisions 178142 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines Merged revisions 178141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras ........ ................ 2009-02-21 16:04 +0000 [r177945] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 177944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177944 | tilghman | 2009-02-21 09:59:49 -0600 (Sat, 21 Feb 2009) | 2 lines On update, test against the existence of sipregs. ........ 2009-02-21 12:51 +0000 [r177851] Michiel van Baak * /, channels/chan_sip.c: Merged revisions 177849 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177849 | mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines make chan_sip.c compile on OpenBSD again. ........ 2009-02-20 23:05 +0000 [r177789] Tilghman Lesher * main/pbx.c, /: Merged revisions 177787 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009) | 16 lines Merged revisions 177786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines Don't print the CR-NL combination when we aren't outputting to the manager. An embedded CR-NL in a CLI command screws up several AMI parsers that don't expect to see that combination in the middle of output. (Closes issue #14305) Reported by: martins Patch by: tilghman ........ ................ 2009-02-20 22:27 +0000 [r177785] Dwayne M. Hubbard * /, apps/app_fax.c: Merged revisions 177699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177699 | dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines Make app_fax compatible with spandsp-0.0.6pre4 Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred integer to indicate the number of pages transferred (so far) during the fax session. The spandsp-0.0.6pre4 release removed the pages_transferred integer and replaced it with two different integers - pages_tx and pages_rx. This revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards compatibility for previous spandsp releases. ........ 2009-02-20 22:15 +0000 [r177760-177764] Tilghman Lesher * include/asterisk/strings.h: Oops, last merge broke 1.6.1 branch * apps/app_system.c, include/asterisk/app.h, /, main/app.c: Merged revisions 177664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 | tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines Allow semicolons to be escaped, when passing arguments to the System command. (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert ........ * include/asterisk/threadstorage.h, /: Merged revisions 177732 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177732 | tilghman | 2009-02-20 15:25:37 -0600 (Fri, 20 Feb 2009) | 10 lines Merged revisions 177701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed. Fixed for snuff-home on -dev channel. ........ ................ 2009-02-20 20:34 +0000 [r177700] David Vossel * channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.6 only audio codec bits 0-12 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue #14283) Reported by: jcovert 2009-02-20 17:28 +0000 [r177663] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 177661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177661 | tilghman | 2009-02-20 11:22:19 -0600 (Fri, 20 Feb 2009) | 2 lines Oops, merge broke trunk ........ 2009-02-20 00:38 +0000 [r177626] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 177624 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177624 | jpeeler | 2009-02-19 18:35:53 -0600 (Thu, 19 Feb 2009) | 7 lines Set sip_request ast_str data to NULL so ast_str_copy allocates space properly in copy_request (issue #14478) Reported by: erik_dedecker ........ 2009-02-20 00:26 +0000 [r177623] Steve Murphy * /, main/Makefile, main/ast_expr2f.c: Merged revisions 177595 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177595 | murf | 2009-02-19 16:56:50 -0700 (Thu, 19 Feb 2009) | 32 lines Merged revisions 177540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was already pretty 8-bit clean; but I'm still removing the --full from the flex command so everything is uniform. ........ r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines This patch fixes a problem with 8-bit input to the ast_expr2 scanner. The real culprit was the --full argument to flex in the Makefile! This causes a 7-bit scanner to be generated. I reviewed the rules and found one rule where I needed to specifically include 8-bit chars for a token. I tested against the text supplied by ibercom, and all looks very well. This has been there a surprisingly long time! (closes issue #14498) Reported by: ibercom Patches: 14498.patch uploaded by murf (license 17) Tested by: murf ........ ................ 2009-02-19 22:35 +0000 [r177539] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 177537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177537 | tilghman | 2009-02-19 16:33:00 -0600 (Thu, 19 Feb 2009) | 14 lines Merged revisions 177536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads. (closes issue #14253) Reported by: Skavin Patches: 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14) Tested by: Skavin ........ ................ 2009-02-19 16:46 +0000 [r177389] Jeff Peeler * /, include/asterisk/channel.h: Merged revisions 177387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19 Feb 2009) | 3 lines Fix another merge error from 176708 ........ 2009-02-19 16:40 +0000 [r177386] Joshua Colp * apps/app_speech_utils.c, /: Merged revisions 177384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177384 | file | 2009-02-19 12:38:41 -0400 (Thu, 19 Feb 2009) | 10 lines Merged revisions 177383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines If we are able to create a speech structure unset the ERROR variable in case it was previously set. (issue #LUMENVOX-13) ........ ................ 2009-02-19 15:57 +0000 [r177358] Jeff Peeler * /, main/features.c: Merged revisions 177356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177356 | jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev on the asterisk-dev mailing list. Thanks! ........ 2009-02-19 00:17 +0000 [r177294] Steve Murphy * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177286 | murf | 2009-02-18 16:50:57 -0700 (Wed, 18 Feb 2009) | 39 lines Merged revisions 177225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | 34 lines This patch fixes a regression of sorts that was introduced in rev 24425. It basically fixes AST-190/ABE-1782. What was wrong: the user has 6000 extensions in one context; and then 6000 contexts, one per extension. The parser could only handle about 4893 of the 6000 extens in the single context. This was due to the regression I mentioned. To get rid of shift/reduce conflicts, Luigi set up right-recursive lists for globals, context elements, switch lists, and statements. Right recursive lists got rid of the warnings, but instead, they use up a tremendous amount of stack space when the lists are long. I saw this a few years back, and resolved not to fix it until someone complained. That day has arrived! After the changes were made, I ran the regression test suite, and there were no problems. I took the test case the user provided, and added 100,000 extensions to the single context, that already had 6,000 extens in it. (I'll see your 6, and raise you 100!) It takes a few minutes to read it all in, check it and generate code for it, but no problems. So, I think I can say that fundamentally, there are no longer any limits on the number of items you can place in contexts, statement blocks, switches, or globals, beyond your virt mem constraints. ........ ................ 2009-02-18 23:15 +0000 [r177230] Kevin P. Fleming * main/frame.c, /: Merged revisions 177229 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177229 | kpfleming | 2009-02-18 17:09:58 -0600 (Wed, 18 Feb 2009) | 3 lines fix two very minor bugs: if anyone ever uses SLINEAR16 as a format in RTP, ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps. ........ 2009-02-18 23:03 +0000 [r177228] David Vossel * /, main/features.c: Merged revisions 177226 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177226 | dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines Locking issue in action_bridge and bridge_exec action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it. issue# 14296 Review: http://reviewboard.digium.com/r/167/ ........ 2009-02-18 20:16 +0000 [r177164] Jeff Peeler * channels/h323/chan_h323.h, channels/h323/cisco-h225.cxx, channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h, /, channels/h323/caps_h323.cxx, channels/h323/ast_ptlib.h (added), channels/h323/ast_h323.cxx, configure, channels/h323/compat_h323.h, configure.ac, channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4, channels/h323/ast_h323.h: Merged revisions 177162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177162 | jpeeler | 2009-02-18 14:11:57 -0600 (Wed, 18 Feb 2009) | 14 lines Modify h323 to build against PTLib as well as the older PWLib Several changes in PTLib have occurred requiring build time detection. Changes accounted for include the library name change, config option change, install location change, and a boolean type change which is handled by ast_ptlib.h. Also, the sed check has been modified to properly work with autoconf >= 2.62. (closes issue #14224) Reported by: bergolth Patches: asterisk-autoconf-sed.patch uploaded by bergolth (license 661) asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested by: jpeeler ........ 2009-02-18 19:30 +0000 [r177158] Russell Bryant * /, apps/app_meetme.c: Merged revisions 177101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177101 | russell | 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) | 8 lines Re-add 'o' option to MeetMe, reverting rev 62297. Enabling this option by default proved to be a bad idea, as the talker detection is not very reliable. So, make it optional again, and off by default. (issue #13801) Reported by: justdave ........ 2009-02-18 19:09 +0000 [r177100] Tilghman Lesher * /, include/asterisk/config.h: Merged revisions 177098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177098 | tilghman | 2009-02-18 13:05:15 -0600 (Wed, 18 Feb 2009) | 9 lines Merged revisions 177096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) | 2 lines Document the return value of the update method (as requested on -dev list) ........ ................ 2009-02-18 17:26 +0000 [r177037] Doug Bailey * /, main/utils.c: Merged revisions 177035 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 | dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines Fixed error where a check for an zero length, terminated string was needed. ........ 2009-02-18 17:14 +0000 [r177007] Joshua Colp * /, channels/chan_sip.c: Merged revisions 177005 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177005 | file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines Fix ordering of output for a ChannelUpdate manager event. (closes issue #14497) Reported by: vinsik Patches: chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) ........ 2009-02-18 16:20 +0000 [r176962] Doug Bailey * /, main/utils.c: Merged revisions 176948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176948 | dbailey | 2009-02-18 10:09:12 -0600 (Wed, 18 Feb 2009) | 2 lines Need to take into account the \0 terminator of the old string to determine the amount available. ........ 2009-02-18 15:59 +0000 [r176946] Steve Murphy * main/pbx.c, /: Merged revisions 176943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176943 | murf | 2009-02-18 08:35:26 -0700 (Wed, 18 Feb 2009) | 45 lines This patch fixes merge_contexts_and_delete so it does not deadlock when hints are present. Reason: when I re-engineered the merge_and_delete func to reduce its lock time, I failed to notice that the functions it calls still also do locking as before. This leads to deadlocks on dialplan reloads, when there are actually living, subscribed hints registered in the system. While the reporter come across this problem while using AEL, I might note that these deadlocks should also happen if extensions.conf were used. Here I added these routines to pbx.c: ast_add_extension_nolock add_pri_lockopt ast_add_extension2_lockopt find_context add_hint_nolock All of the above routines are static and restricted to be used only within pbx.c, and more specifically within the merge_contexts_and_delete routine. They are pretty much the same as their counterparts except they don't lock contexts or hints. Most of them now do the real work of their name-alike, with optional locking via extra arguments, and are called by their name-alike. The goal was to have the original functions so they would behave exactly as before. Both PJ and I tested these fixes, and the deadlocking problem is no longer encountered. (closes issue #14357) Reported by: pj Patches: 14357.diff uploaded by murf (license 17) Tested by: pj, murf ........ 2009-02-18 06:15 +0000 [r176903-176906] Russell Bryant * include/asterisk/heap.h, /: Merged revisions 176904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176904 | russell | 2009-02-18 00:14:47 -0600 (Wed, 18 Feb 2009) | 2 lines Add example code for a heap traversal. ........ * main/pbx.c, /: Merged revisions 176901 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176901 | russell | 2009-02-18 00:00:40 -0600 (Wed, 18 Feb 2009) | 9 lines Fix a number of incorrect uses of strncpy(). The big problem here is that the 3rd argument provided in these uses of strncpy() did not reserve a byte for the null terminator, leaving the potential for writing one byte past the end of the buffer. Aside from this, there were coding guidelines violations with regards to spacing, as well as hard coded lengths being used instead of sizeof(). ........ 2009-02-18 00:23 +0000 [r176809] Shaun Ruffell * /, codecs/codec_dahdi.c: Merged revisions 176760 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009) | 10 lines Several changes to codec_dahdi to play nice with G723. This commit brings in the changes that were living out on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. codec_dahdi.c now always uses signed linear as the simple codec so that a soft g729 codec will not end up being preferred to the hardware codec. There are also changes to allow codec_dahdi.c to feed packets to the hardware in the native sample size of the codec. This solves problems with choppy audio when using G723. ........ 2009-02-17 22:21 +0000 [r176731] Dwayne M. Hubbard * /, channels/chan_sip.c: Merged revisions 176705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176705 | dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11 lines create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38 This is required to create a UDPTL structure in create_addr_from_peer() to handle the scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but is defined the peer's context. I tested this patch by enabling t38pt_udptl in the [general] section on one system and only enabling t38pt_udptl in a peer's context on the system sending a fax. Without the patch, the sending system will fail to initiate T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure". When this patch is applied the sending side will successfully initiate T38 negotiation. ........ 2009-02-17 22:15 +0000 [r176711] Jeff Peeler * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 176708 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ ................ 2009-02-17 21:41 +0000 [r176699] Mark Michelson * include/asterisk/frame.h, /: Merged revisions 176697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176697 | mmichelson | 2009-02-17 15:40:09 -0600 (Tue, 17 Feb 2009) | 3 lines Clear up documentation of AST_FRIENDLY_OFFSET in frame.h ........ 2009-02-17 21:24 +0000 [r176675] Russell Bryant * main/timing.c, main/channel.c, /, res/res_timing_pthread.c, res/res_timing_dahdi.c, include/asterisk/timing.h: Merged revisions 176666 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176666 | russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ ........ 2009-02-17 21:16 +0000 [r176644] Tilghman Lesher * res/res_odbc.c, channels/chan_local.c, /: Merged revisions 176592,176642 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176592 | tilghman | 2009-02-17 12:49:20 -0600 (Tue, 17 Feb 2009) | 4 lines Add assertions in the quest to track down a refcount leak. (closes issue #14485) Reported by: davevg ........ r176642 | tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines Prior to masquerade, move the group definitions to the channel performing the masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ 2009-02-17 20:57 +0000 [r176559-176637] Russell Bryant * tests/test_heap.c (added), /: Merged revisions 176635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176635 | russell | 2009-02-17 14:56:26 -0600 (Tue, 17 Feb 2009) | 4 lines Add a test module for the heap implementation. Review: http://reviewboard.digium.com/r/160/ ........ * include/asterisk/heap.h (added), /, main/Makefile, main/heap.c (added): Merged revisions 176632 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176632 | russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines Add an implementation of the heap data structure. A heap is a convenient data structure for implementing a priority queue. Code from svn/asterisk/team/russell/heap/. Review: http://reviewboard.digium.com/r/160/ ........ * apps/app_queue.c, main/pbx.c, /: Merged revisions 176557 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176557 | russell | 2009-02-17 11:33:38 -0600 (Tue, 17 Feb 2009) | 12 lines Fix a race condition that caused device states to become incorrect for hints. The problem here is that the hint processing code was subscribed to the wrong event type. So, it started processing state for a hint too soon, before the device state cache had been updated. Also, fix a similar bug in app_queue, as it was also subscribed to the wrong event type. (closes issue #14461) Reported by: alecdavis ........ 2009-02-17 14:48 +0000 [r176461-176503] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 176501 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176501 | tilghman | 2009-02-17 08:39:36 -0600 (Tue, 17 Feb 2009) | 3 lines In this version, we can combine the queries, because we support dropping nonexistent columns. ........ * /, channels/chan_sip.c: Merged revisions 176459 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009) | 17 lines Merged revisions 176426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines After a 'sip reload', qualifies for realtime peers weren't immediately restarted, instead waiting until the next registration. We're now caching the qualify across a reload/restart and starting the qualify immediately upon loading the peer. (closes issue #14196) Reported by: pdf Patches: 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) Tested by: pdf ........ ................ 2009-02-16 23:57 +0000 [r176362] David Vossel * channels/chan_iax2.c, /: Merged revisions 176355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176355 | dvossel | 2009-02-16 17:33:55 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that. issue #13749 ........ ................ 2009-02-16 23:17 +0000 [r176321] Tilghman Lesher * /, channels/chan_skinny.c: Merged revisions 176320 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176320 | tilghman | 2009-02-16 17:14:08 -0600 (Mon, 16 Feb 2009) | 7 lines Use the correct list macros for deleting an item from the middle of a list. (issue #13777) Reported by: pj Patches: 20090203__bug13777.diff.txt uploaded by Corydon76 (license 14) Tested by: pj ........ 2009-02-16 22:00 +0000 [r176259] Kevin P. Fleming * include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 176255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ ................ 2009-02-16 21:50 +0000 [r176257] Mark Michelson * /, apps/app_meetme.c: Merged revisions 176253 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176253 | mmichelson | 2009-02-16 15:40:40 -0600 (Mon, 16 Feb 2009) | 24 lines Merged revisions 176249,176252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it to be nonblocking atomically Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys ........ r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines Remove unused variable and make dev-mode compilation happy ........ ................ 2009-02-16 21:36 +0000 [r176251] David Vossel * channels/chan_iax2.c, /: Merged revisions 176248 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176248 | dvossel | 2009-02-16 15:30:17 -0600 (Mon, 16 Feb 2009) | 11 lines Merged revisions 175597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ ................ 2009-02-16 18:38 +0000 [r176176] Mark Michelson * /, main/logger.c: Merged revisions 176174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176174 | mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11 lines Assist proper thread synchronization when stopping the logger thread. I was finding that on my dev box, occasionally attempting to "stop now" in trunk would cause Asterisk to hang. I traced this to the fact that the logger thread was waiting on a condition which had already been signalled. The logger thread also need to be sure to check the value of the close_logger_thread variable. The close_logger_thread variable is only checked when the list of logmessages is empty. This allows for the logger thread to print and free any pending messages before exiting. ........ 2009-02-16 17:10 +0000 [r176102] Russell Bryant * /, channels/chan_features.c (removed): Merged revisions 176100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176100 | russell | 2009-02-16 11:09:24 -0600 (Mon, 16 Feb 2009) | 4 lines Remove chan_features. Review: http://reviewboard.digium.com/r/161/ ........ 2009-02-16 17:07 +0000 [r176099] Tilghman Lesher * configs/func_odbc.conf.sample: Eliminate mention of a variable which exists only in trunk. (Thanks, jsmith) 2009-02-16 15:38 +0000 [r176032] Joshua Colp * /, channels/chan_sip.c: Merged revisions 176030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) | 16 lines Merged revisions 176029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) ........ ................ 2009-02-16 09:42 +0000 [r176023] Michiel van Baak * include/asterisk/manager.h, doc/unistim.txt, channels/chan_unistim.c, /, channels/chan_sip.c: Merged revisions 175952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines Merged revisions 175921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ ................ 2009-02-15 21:28 +0000 [r175831-175890] Russell Bryant * main/sched.c, /, include/asterisk/sched.h: Merged revisions 175882 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175882 | russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines Make ast_sched_report() and ast_sched_dump() thread safe. ........ * main/sched.c, /, channels/chan_sip.c, include/asterisk/sched.h: Merged revisions 175829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175829 | russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines Fix a number of problems with ast_sched_report(). 1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. ........ 2009-02-13 20:48 +0000 [r175662] David Vossel * channels/chan_iax2.c, configs/iax.conf.sample, channels/iax2.h: Merged revisions 175597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ 2009-02-13 19:52 +0000 [r175593] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 175591 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175591 | mmichelson | 2009-02-13 13:49:38 -0600 (Fri, 13 Feb 2009) | 22 lines Merged revisions 175590 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) ........ ................ 2009-02-13 16:44 +0000 [r175551] Joshua Colp * /, apps/app_record.c: Merged revisions 175549 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian ........ 2009-02-12 21:41 +0000 [r175370] Russell Bryant * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines Remove useless string copy, and make sscanf safe again ........ 2009-02-12 21:27 +0000 [r175342] Tilghman Lesher * main/udptl.c, /: Merged revisions 175334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ ................ 2009-02-12 20:51 +0000 [r175300] Jeff Peeler * /, main/features.c: Merged revisions 175298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ ................ 2009-02-12 20:48 +0000 [r175257-175297] Russell Bryant * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines Avoid using ast_strdupa() in a loop. ........ * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) | 4 lines Don't enable something by default that has a dependency on something _not_ enabled by default. menuselect was not happy with this. ........ 2009-02-12 18:50 +0000 [r175251] Kevin P. Fleming * channels/chan_iax2.c, /: Merged revisions 175250 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb 2009) | 1 line correct warning message to not refer specifically to DAHDI ........ 2009-02-12 18:01 +0000 [r175190] Jeff Peeler * /, main/features.c: Merged revisions 175188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ ................ 2009-02-12 17:09 +0000 [r175130] David Vossel * channels/chan_iax2.c, /: Merged revisions 175127 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175127 | dvossel | 2009-02-12 11:07:17 -0600 (Thu, 12 Feb 2009) | 4 lines Setting key rotation to be off by default Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off. ........ 2009-02-12 17:08 +0000 [r175129] Russell Bryant * main/rtp.c, /: Merged revisions 175125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ ................ 2009-02-12 16:35 +0000 [r174947-175123] Mark Michelson * main/astobj2.c, /, include/asterisk/astobj2.h: