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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-11-09 17:09:31 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-11-09 17:09:31 +0000
commite152502317f839014deb2f38ede411cb348dee1a (patch)
tree82043670dbd7975f47c8c2a43d666770a7788a3f
parent8f4b1b19f06fe8c893e98cc6669335b50276429d (diff)
Importing files for 1.6.2.0-rc5 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.0-rc5@228976 f38db490-d61c-443f-a65b-d21fe96a405b
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-rw-r--r--.version1
-rw-r--r--ChangeLog20196
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diff --git a/.version b/.version
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+1.6.2.0-rc5
diff --git a/ChangeLog b/ChangeLog
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@@ -0,0 +1,20196 @@
+2009-11-09 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc5
+
+2009-11-09 15:40 +0000 [r228900] Leif Madsen <lmadsen@digium.com>
+
+ * main/channel.c: Merged revisions 228897 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009)
+ | 14 lines Merged revisions 228896 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
+ | 6 lines Update WARNING message. Update a WARNING message to
+ give a suggested fix when encountered. (closes issue #16198)
+ Reported by: atis Tested by: atis ........ ................
+
+2009-11-09 14:48 +0000 [r228859] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600
+ (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov
+ 2009) | 8 lines Perform limited bounds checking when destroying
+ ast_mutex_t structures to make sure we don't try to use negative
+ indices. (closes issue #15588) Reported by: zerohalo Patches:
+ 20090820__issue15588.diff.txt uploaded by tilghman (license 14)
+ Tested by: zerohalo ........ ................
+
+2009-11-06 22:37 +0000 [r228694] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 228693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009)
+ | 16 lines Merged revisions 228692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
+ | 9 lines fixes audiohook write crash occuring in chan_spy
+ whisper mode. After writing to the audiohook list in ast_write(),
+ frames were being freed incorrectly. Under certain conditions
+ this resulted in a double free crash. (closes issue #16133)
+ Reported by: wetwired ........ ................
+
+2009-11-06 20:26 +0000 [r228649] Matthew Nicholson <mnicholson@digium.com>
+
+ * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600
+ (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
+ 2009) | 8 lines Properly handle '=' while decoding base64
+ messages and null terminate strings returned from BASE64_DECODE.
+ (closes issue #15271) Reported by: chappell Patches:
+ base64_fix.patch uploaded by chappell (license 8) Tested by:
+ kobaz ........ ................
+
+2009-11-06 18:43 +0000 [r228551] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-rc4
+
+2009-11-06 17:54 +0000 [r228504] Joshua Colp <jcolp@digium.com>
+
+ * doc/tex/localchannel.tex, /: Merged revisions 228499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2
+ lines Fix the localchannel.tex file. ........
+
+2009-11-06 17:24 +0000 [r228421-228447] David Vossel <dvossel@digium.com>
+
+ * codecs/codec_ilbc.c, /: Merged revisions 228441 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 |
+ dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines
+ Fixes merging issue from 1.4, frame data is held in data.ptr in
+ trunk ........
+
+ * codecs/codec_ilbc.c, /: Merged revisions 228420 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009)
+ | 19 lines Merged revisions 228418 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
+ | 13 lines fixes segfault in iLBC For reasons not yet known, it
+ appears possible for an ast_frame to have a datalen greater than
+ zero while the actual data is NULL during Packet Loss
+ Concealment. Most codecs don't support PLC so this doesn't affect
+ them. This patch catches the malformed frame and prevents the
+ crash from occuring. Additional efforts to determine why it is
+ possible for a frame to look like this are still being
+ investigated. (issue #16979) ........ ................
+
+2009-11-06 16:44 +0000 [r228413] Joshua Colp <jcolp@digium.com>
+
+ * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) |
+ 14 lines Merged revisions 228409 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
+ lines Fix a bug caused by a partially invalid frame (from the
+ jitterbuffer) passing through the Asterisk core. (closes issue
+ #15560) Reported by: jvandal (closes issue #15709) Reported by:
+ covici ........ ................
+
+2009-11-06 15:43 +0000 [r228269-228340] David Vossel <dvossel@digium.com>
+
+ * /, main/astfd.c: Merged revisions 228339 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009)
+ | 12 lines Merged revisions 228338 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
+ | 5 lines fixes crash in astfd.c (closes issue #15981) Reported
+ by: slavon ........ ................
+
+ * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06
+ Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c
+ (closes issue #15394) Reported by: boroda Patches:
+ bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790)
+ Tested by: dbrooks, boroda ........
+
+2009-11-05 22:13 +0000 [r228198] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 |
+ tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines
+ Yet another error message in the dialplan (thanks,
+ rmudgett/russellb) ........
+
+2009-11-05 21:27 +0000 [r228195] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 |
+ jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
+ Fix the fix for chanspy option o In 224178, I assumed the
+ uploaded patch was correct as it had received positive feedback.
+ The flags were being checked in the incorrect location. Upon
+ testing the fix this time it was also found that the flags from
+ the dialplan weren't being copied to the
+ chanspy_translation_helper. (closes issue #16167) Reported by:
+ marhbere ........
+
+2009-11-05 21:27 +0000 [r228194] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 |
+ tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines
+ MEETME_INFO should not return a literal error message to the
+ dialplan. (closes issue #15450) Reported by: JimVanM Patches:
+ meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested
+ by: JimVanM ........
+
+2009-11-05 19:42 +0000 [r228148] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600
+ (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009)
+ | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to
+ chan_misdn connection. Patch submitted by gknispel_proformatique,
+ tested by francesco_r. "I have many crash since i have upgraded
+ to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out
+ an ast_frame. (closes issue #16041) Reported by: francesco_r
+ ........ ................
+
+2009-11-05 19:20 +0000 [r228093] Jason Parker <jparker@digium.com>
+
+ * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r228080 | qwell | 2009-11-05 13:16:29 -0600
+ (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) |
+ 8 lines Fix crash on VPB exception when no hardware is present.
+ (closes issue #14970) Reported by: tzafrir Patches:
+ vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+ markwaters ........ ................
+
+2009-11-05 17:14 +0000 [r228017] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_externalivr.c, /: Merged revisions 228015 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009)
+ | 4 lines Don't crash if no arguments are passed. (closes issue
+ #16119) Reported by: thedavidfactor ........
+
+2009-11-04 23:53 +0000 [r227947] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009)
+ | 21 lines Merged revisions 227944 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
+ | 14 lines Fix incorrect filename comparsion after monitor file
+ change The logic to detect if a requested file is indeed a
+ different file from the current file was incorrect. The main
+ issue being confusion of the use of filename_base which was
+ previously set without pathing information and then compared to
+ another full path. Robust file comparison logic has been added to
+ properly check if two files are the same even if symlinks are
+ used. (closes issue #15313) Reported by: caspy Patches:
+ 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
+ 325) but mostly tilghman's work ........ ................
+
+2009-11-04 21:09 +0000 [r227760-227831] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov
+ 2009) | 17 lines Merged revisions 227827 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
+ 2009) | 10 lines This patch modifies the Dial application to
+ monitor the calling channel for hangups while playing back
+ announcements. (closes issue #16005) Reported by: falves11
+ Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: mnicholson, falves11 Review:
+ https://reviewboard.asterisk.org/r/407/ ........ ................
+
+ * channels/chan_sip.c: Modify the SDP parsing code to parse session
+ and media level items separately. With the new code, media level
+ proprieties should no longer be confused with session level
+ proprieties. This change also reorganizes some of the SDP parsing
+ code which should make it easier to manage in the future. (closes
+ issue #14994) Reported by: frawd
+
+2009-11-04 19:28 +0000 [r227733-227748] Joshua Colp <jcolp@digium.com>
+
+ * /, static-http/prototype.js: Merged revisions 227739 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed,
+ 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5
+ lines Fix a security issue where it may be possible for someone
+ to execute a cross-site AJAX request exploit. (AST-2009-009)
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) |
+ 12 lines Merged revisions 227700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
+ lines Fix a security issue where sending a REGISTER with a
+ differing username in the From URI and Authorization header would
+ reveal whether it was valid or not. (AST-2009-008) ........
+ ................
+
+2009-11-03 20:01 +0000 [r227375] Jason Parker <jparker@digium.com>
+
+ * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) |
+ 9 lines Fix some build issues on Solaris. (closes issue #14517)
+ (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded
+ by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell
+ ........
+
+2009-11-03 19:49 +0000 [r227364-227371] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_controlplayback.c, /: Merged revisions 227368 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03
+ Nov 2009) | 8 lines Change warning message to debug message.
+ app_controlplayback outputs a warning, when in fact it is normal.
+ (closes issue #16071) Reported by: atis Patches:
+ controlplayback_warning.patch uploaded by atis (license 242)
+ ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 227361 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03
+ Nov 2009) | 11 lines Additional fixes to the
+ extensions.conf.sample file. Update the extensions.conf.sample
+ [stdexten] context so that we use the variable instead of
+ requiring it to be passed explicitly. Also updated uses of the
+ [stdexten] context throughout. (closes issue #15858) Reported by:
+ pprindeville Patches: stdexten-context-update.txt uploaded by
+ lmadsen (license 10) Tested by: pprindeville ........
+
+2009-11-03 18:15 +0000 [r227280] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
+ | 4 lines Make sure the outgoing flag is cleared if a new channel
+ fails to get created for outgoing calls. This is the relevant
+ portion of asterisk/trunk -r226648 ........
+
+2009-11-03 17:14 +0000 [r227239] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 |
+ dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines
+ user.conf entries in SIP were not having their peer type set.
+ (closes issue #16120) Reported by: jsmith ........
+
+2009-11-03 15:40 +0000 [r227170] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) |
+ 12 lines Merged revisions 227166 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
+ lines Fix a bug where an RPID header could be generated with a
+ blank username in the URI. (closes issue #15909) Reported by:
+ kobaz ........ ................
+
+2009-11-03 15:25 +0000 [r227165] Leif Madsen <lmadsen@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 227162 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03
+ Nov 2009) | 7 lines Update extensions.conf.sample file to fix
+ incorrect extensions. (closes issue #15857) Reported by:
+ pprindeville Patches: stdexten.patch#2 uploaded by pprindeville
+ (license 347) Tested by: pprindeville ........
+
+2009-11-03 13:51 +0000 [r227156] Olle Johansson <oej@edvina.net>
+
+ * Makefile, /, channels/chan_sip.c: Merged revisions 227091 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis,
+ 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
+ lines Use proper response code when violating Contact ACL's.
+ https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
+ quick review. (EDVX-003) ........ ................
+
+2009-11-02 21:06 +0000 [r226978] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_sip.c: SIP channel name uniqueness SIP channel
+ names were supposed to be unique by way of a name suffix derived
+ from the pointer to the channel's private data. Uniqueness was
+ preserved on 32-bit systems, but not on 64-bit systems. This
+ patch, as suggested by kpfleming, replaces this suffix with a
+ simple incremented unsigned int. (closes issue #15152) Reported
+ by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
+
+2009-11-02 18:12 +0000 [r226893] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) |
+ 18 lines Merged revisions 226889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
+ 11 lines Fix a bug where the recorded privacy introduction file
+ would not get removed if the caller hung up while the called
+ party had not yet answered. This was fixed by introducing an
+ argument to the 'n' option which, when enabled, removes the
+ introduction file under all scenarios. This was done to preserve
+ the behavior that has existed for quite some time. (closes issue
+ #14674) Reported by: ulogic Patches: bug14674.patch uploaded by
+ jpeeler (license 325) ........ ................
+
+2009-11-02 17:17 +0000 [r226815] Tilghman Lesher <tlesher@digium.com>
+
+ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600
+ (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
+ | 8 lines Don't allow two separate instances of safe_asterisk
+ when restarting from the init script. (closes issue #14562)
+ Reported by: davidw Patches: Initially
+ 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
+ Modified to 20091030__Issue14562_diff.txt uploaded by davidw
+ (license 780) Tested by: davidw ........ ................
+
+2009-10-29 18:18 +0000 [r226540] Joshua Colp <jcolp@digium.com>
+
+ * doc/tex/localchannel.tex, channels/chan_local.c, /: Merged
+ revisions 226532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) |
+ 13 lines Merged revisions 226531 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
+ lines Add an option to enabling passing music on hold start and
+ stop requests through instead of acting on them in chan_local.
+ (closes issue #14709) Reported by: dimas ........
+ ................
+
+2009-10-28 21:32 +0000 [r226486] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * build_tools/get_documentation, /: remove empty awk pattern (//)
+ Solaris 10 nawk doesn't like the empty pattern such as '//' for
+ 'always'. Just remove that. No pattern at all always matches.
+ Merged revisions 226453 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk
+
+2009-10-28 20:13 +0000 [r226379-226385] Leif Madsen <lmadsen@digium.com>
+
+ * configs/sip.conf.sample: Merged revisions 226384 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500
+ (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009)
+ | 9 lines Update documentation in sip.conf.sample. Update the
+ documentation in sip.conf.sample in order to make it more clear
+ that directmedia/canreinvite do not cause Asterisk to ignore
+ reINVITEs. It is only used to stop Asterisk from generating a
+ reINVITE, but does not stop it from accepting them if necessary.
+ (closes issue #15644) Reported by: lmadsen ........
+ ................
+
+ * doc/tex/channelvariables.tex: Merged revisions 226378 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500
+ (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
+ | 7 lines Update CALLINGSUBADDR channel variable documentation.
+ (closes issue #15734) Reported by: alecdavis Patches:
+ channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
+ Tested by: alecdavis ........ ................
+
+2009-10-28 18:06 +0000 [r226170-226308] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 226305 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500
+ (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28
+ Oct 2009) | 2 lines Fix documentation (pointed out by
+ TheDavidFactor on #-dev) ........ ................
+
+ * main/manager.c, /: Merged revisions 226159 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009)
+ | 14 lines Merged revisions 226138 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
+ | 7 lines Manager output is not always NULL-terminated, so force
+ a NULL at the end of the filestream. (closes issue #15495)
+ Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
+ by tilghman (license 14) Tested by: pdf ........ ................
+
+2009-10-27 17:12 +0000 [r226101] Terry Wilson <twilson@digium.com>
+
+ * res/res_http_post.c, /: Merged revisions 226099 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 |
+ twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines
+ Don't prepend the URI prefix to the post directory ........
+
+2009-10-27 00:16 +0000 [r226055] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, configure, configure.ac: detect ARM Linux EABI OSARCH as
+ linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
+ if host_os is linux-gnueabi * When checking if we are Linux,
+ check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
+ the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
+ sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
+ tested for the value of 'linux-gnu' in one or two places in the
+ tree. This patch also fixes the check libcap to check for $OSARCH
+ rather than $host_os . See also:
+ http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
+ svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4
+ Merged revisions 226018 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk
+
+2009-10-26 19:42 +0000 [r225914] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 |
+ jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines
+ ACL check not present for verifying SIP INVITEs The ACL check in
+ check_peer_ok was missing and has now been restored. The missing
+ check allowed for calls to be made on prohibited networks where
+ an ACL was defined in sip.conf and the allowguest option was set
+ to off. See the AST security advisory below for more information.
+ Merge code associated with AST-2009-007. (closes issue #16091)
+ Reported by: thom4fun ........
+
+2009-10-26 15:56 +0000 [r225871] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_fax.c: Backport audio handling loop fixes from trunk
+ version of app_fax. This backport resolves some issues handling
+ audio frames during FAX processing, and ensures that the FAX
+ application doesn't accidentally get notified of a T.38
+ switchover at the end of a successful FAX. (closes issue #16127)
+
+2009-10-23 14:46 +0000 [r225651] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 |
+ dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines
+ Fixes an iterator memory leak and uninitialized memory ........
+
+2009-10-23 14:08 +0000 [r225585] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 225582 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct
+ 2009) | 17 lines Merged revisions 225581 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
+ 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
+ every build. For some reason the menuselect.makeopts file was
+ listed as PHONY in the Makefile, resulting in 'make' needing to
+ rebuild it for every build. This then resulted in the embedded
+ module rules being rebuilt on every build, which can be slow and
+ is unnecessary. This patch fixes the problem by properly allowing
+ 'make' to know when the menuselect.makeopts file needs to be
+ rebuilt (defining the proper dependencies). ........
+ ................
+
+2009-10-22 22:24 +0000 [r225516] Leif Madsen <lmadsen@digium.com>
+
+ * README, /: Merged revisions 225515 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225515 |
+ lmadsen | 2009-10-22 17:24:03 -0500 (Thu, 22 Oct 2009) | 8 lines
+ Update README documentation. Update the README documentation to
+ correctly describe which CLI command you should use when
+ attempting to get help from the CLI. (closes issue #16064)
+ Reported by: thedavidfactor Patches: readme.patch uploaded by
+ thedavidfactor (license 903) ........
+
+2009-10-22 21:55 +0000 [r225489] David Vossel <dvossel@digium.com>
+
+ * apps/app_externalivr.c, include/asterisk/tcptls.h, main/tcptls.c,
+ /, channels/chan_sip.c: Merged revisions 225445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 |
+ dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines
+ SIP TCP/TLS: move client connection setup/write into tcp helper
+ thread, various related locking/memory fixes. What this patch
+ fixes 1.Moves sip TCP/TLS connection setup into the TCP helper
+ thread: Connection setup takes awhile and before this it was
+ being done while holding the monitor lock. 2.Moves TCP/TLS
+ writing to the TCP helper thread: Through the use of a packet
+ queue and an alert pipe, the TCP helper thread can now be woken
+ up to write data as well as read data. 3.Locking error: sip_xmit
+ returned an XMIT_ERROR without giving up the tcptls_session lock.
+ This lock has been completely removed from sip_xmit and placed in
+ the new sip_tcptls_write() function. 4.Memory leak: When creating
+ a tcptls_client the tls_cfg was alloced but never freed unless
+ the tcptls_session failed to start. Now the session_args for a
+ sip client are an ao2 object which frees the tls_cfg on
+ destruction. 5.Pointer to stack variable: During
+ sip_prepare_socket the creation of a client's
+ ast_tcptls_session_args was done on the stack and stored as a
+ pointer in the newly created tcptls_session. Depending on the
+ events that followed, there was a slight possibility that pointer
+ could have been accessed after the stack returned. Given the new
+ changes, it is always accessed after the stack returns which is
+ why I found it. Notable code changes 1.I broke tcptls.c's
+ ast_tcptls_client_start() function into two functions. One for
+ creating and allocating the new tcptls_session, and a separate
+ one for starting and handling the new connection. This allowed me
+ to create the tcptls_session, launch the helper thread, and then
+ establish the connection within the helper thread. 2.Writes to a
+ tcptls_session are now done within the helper thread. This is
+ done by using an alert pipe to wake up the thread if new data
+ needs to be sent. The thread's sip_threadinfo object contains the
+ alert pipe as well as the packet queue. 3.Since the threadinfo
+ object contains the alert pipe, it must now be accessed outside
+ of the helper thread for every write (queuing of a packet). For
+ easy lookup, I moved the threadinfo objects from a linked list to
+ an ao2_container. (closes issue #13136) Reported by: pabelanger
+ Tested by: dvossel, whys (closes issue #15894) Reported by:
+ dvossel Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/380/ ........
+
+2009-10-22 21:54 +0000 [r225488] Leif Madsen <lmadsen@digium.com>
+
+ * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions
+ 225485 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009)
+ | 19 lines Merged revisions 225484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
+ | 11 lines Clean valgrind output by suppressing false errors.
+ Update valgrind.txt documentation and add valgrind.supp file in
+ order to allow those who are creating valgrind output to have
+ less false errors in the logfile. (closes issue #16007) Reported
+ by: atis Patches: valgrind.txt.diff uploaded by atis (license
+ 242) asterisk2.supp uploaded by atis (license 242) Tested by:
+ atis, amorsen ........ ................
+
+2009-10-22 17:14 +0000 [r225363] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
+ Merged revisions 225360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009)
+ | 11 lines Merged revisions 225105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
+ | 4 lines Fix documentation for ast_softhangup() and correct the
+ misuse thereof. (closes issue #16103) Reported by: majorbloodnok
+ ........ ................
+
+2009-10-21 22:00 +0000 [r225035-225308] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 225307 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500
+ (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009)
+ | 13 lines IAX2: VNAK loop caused by signaling frames with no
+ destination call number It is possible for the PBX thread to
+ queue up signaling frames before a destination call number is
+ received. This can result in signaling frames being sent out with
+ no destination call number. Since recent versions of Asterisk
+ require accurate destination callnumbers for all Full Frames,
+ this can cause a VNAK loop to occur. To resolve this no signaling
+ frames are sent until a destination callnumber is received, and
+ destination call numbers are now only required for iax_pvt
+ matching when the frame is an ACK. Review:
+ https://reviewboard.asterisk.org/r/413/ ........ ................
+
+ * configs/sip.conf.sample, channels/chan_iax2.c,
+ configs/iax.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 225033 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009)
+ | 27 lines Merged revisions 225032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
+ | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
+ id removes '(', ' ', ')', non-trailing '.', and '-' from the
+ string. This means values such as 555.5555 and test-test result
+ in 555555 and testtest. There are instances, such as Skype
+ integration, where a specific value is passed via caller id that
+ must be preserved unmodified. This patch makes the shrinking of
+ caller id optional in chan_sip and chan_iax in order to support
+ such cases. By default this option is on to preserve previous
+ expected behavior. (closes issue #15940) Reported by: dimas
+ Patches: v2-15940.patch uploaded by dimas (license 88)
+ 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
+ Tested by: dvossel Review:
+ https://reviewboard.asterisk.org/r/408/ ........ ................
+
+2009-10-20 22:11 +0000 [r224859] Tilghman Lesher <tlesher@digium.com>
+
+ * main/audiohook.c, funcs/func_speex.c, /: Merged revisions 224856
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500
+ (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
+ | 5 lines Pay attention to the return value of the manipulate
+ function. While this looks like an optimization, it prevents a
+ crash from occurring when used with certain audiohook callbacks
+ (diagnosed with SVN trunk, backported to 1.4 to keep the source
+ consistent across versions). ........ ................
+
+2009-10-20 17:50 +0000 [r224777] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 224774 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) |
+ 12 lines Merged revisions 224773 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
+ lines Add support for relaying early media in the features
+ attended transfer option. (closes issue #14828) Reported by:
+ licedey ........ ................
+
+2009-10-20 00:00 +0000 [r224674] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp.c, /: Merged revisions 224671 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct
+ 2009) | 14 lines Merged revisions 224670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct
+ 2009) | 7 lines Correct timestamp calculations when RTP sample
+ rates over 8kHz are used. While testing some endpoints that
+ support 16kHz and 32kHz sample rates, some log messages were
+ generated due to calc_rxstamp() computing timestamps in a way
+ that produced odd results, so this patch sanitizes the result of
+ the computations. ........ ................
+
+2009-10-19 19:54 +0000 [r224571] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) |
+ 12 lines Merged revisions 224565 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
+ lines Do not attempt early media bridging (ie: direct RTP setup)
+ if options are enabled that should prevent it. (closes issue
+ #14763) Reported by: cupotka ........ ................
+
+2009-10-19 19:41 +0000 [r224563] Kevin P. Fleming <kpfleming@digium.com>
+
+ * formats/format_siren14.c, /: Merged revisions 224562 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r224562 | kpfleming | 2009-10-19 14:40:26 -0500 (Mon, 19 Oct
+ 2009) | 1 line Remove useless debugging message. ........
+
+2009-10-19 00:13 +0000 [r224447-224451] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009)
+ | 3 lines Allow ODBC storage to be queried with multiple
+ mailboxes, and remove multiple goto's. This corrects an issue
+ reported on the -users list. ........
+
+ * configs/res_odbc.conf.sample, /: Merged revisions 224446 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18
+ Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias
+ for "autocommit", but instead controls the default disposition of
+ uncommitted transactions. ........
+
+2009-10-17 01:58 +0000 [r224334] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500
+ (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
+ | 13 lines Fix stale caller id data from being reported in AMI
+ NewChannel event The problem here is that chan_dahdi is designed
+ in such a way to set certain values in the dahdi_pvt only once.
+ One of those such values is the configured caller id data in
+ chan_dahdi.conf. For PRI, the configured caller id data could be
+ overwritten during a call. Instead of saving the data and
+ restoring, it was decided that for all non-analog channels it was
+ simply best to not set the configured caller id in the first
+ place and also clear it at the end of the call. (closes issue
+ #15883) Reported by: jsmith ........ ................
+
+2009-10-16 20:58 +0000 [r224264] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500
+ (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
+ | 18 lines Never released PRI channels when using Busy() or
+ Congestion() dialplan apps. When the Busy() or Congestion()
+ application is used towards ISDN (an ISDN progress is sent), the
+ responding ISDN Disconnect or Release may contain the ISDN cause
+ user busy or one of the congestion causes. In chan_dahdi.c these
+ causes will only set the needbusy or needcongestion flags and not
+ activate the softhangup procedure. Unfortunately only the latter
+ can interrupt the endless wait loop of Busy()/Congestion().
+ Result: PRI channels staying in state busy for the rest of
+ asterisk life or until the other end times out and forces the
+ call to clear. (in issue 0014292) Reported by: tomaso Patches:
+ disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
+ patch is unrelated to the issue.) ........ ................
+
+2009-10-15 15:58 +0000 [r224181] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 |
+ jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines
+ Readd removed ability to allow listening to one side of the call
+ in app_chanspy (Option o) (closes issue #15675) Reported by:
+ john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks
+ (license 790) Tested by: jgutierrez on users list:
+ http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
+ ........
+
+2009-10-12 23:55 +0000 [r223835] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009)
+ | 15 lines Merged revisions 223804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
+ | 8 lines Ensure ringing continues for branched calls after
+ progress is received While waiting for an answer, don't send
+ progress for branched calls for which ringing was sent. (closes
+ issue #15028) Reported by: fnordian ........ ................
+
+2009-10-12 21:01 +0000 [r223757] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009)
+ | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2
+ options SWP-151 ........
+
+2009-10-12 14:37 +0000 [r223655] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12
+ Oct 2009) | 13 lines Remove automatic switching from T.38 to
+ voice mode in chan_sip. chan_sip has some code to automatically
+ switch from T.38 mode to voice mode when a voice frame is written
+ to the channel while it is in T.38 mode; this was intended to
+ handle the situation when a FAX transmission has ended and the
+ channel is not yet hung up, but is causing problems at the
+ beginning of FAX sessions as well when there are still voice
+ frames 'in flight' at the time the T.38 negotiation completes.
+ This patch removes the automatic switchover, and changes app_fax
+ to explicitly switch off T.38 mode when the FAX transmission
+ process ends. (closes issue #16025) Reported by: jamicque
+ ........
+
+2009-10-11 17:32 +0000 [r223490] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c, /: Merged revisions 223487 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009)
+ | 17 lines Merged revisions 223485-223486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009)
+ | 6 lines Don't use data outside of its scope. The purpose of
+ this code was to have a hangup frame put on the list of deferred
+ frames. However, the code that read the hangup frame was outside
+ of the scope of where the hangup frame was declared. ........
+ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009)
+ | 2 lines Remove some unnecessary code. ........ ................
+
+2009-10-09 23:12 +0000 [r223406] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation
+ of PRIREDIRECTIONREASON set by chan_sip. This commit is the
+ simplest way to solve a problem that has already been solved in
+ trunk with the "COLP/CONP and Redirecting party information into
+ Asterisk" commit. In trunk the redirection reason is translated
+ into a generic redirect reason. I would have had to do the same
+ fix except chan_sip never reads PRIREDIRECTREASON. So both
+ chan_dahdi and chan_h323 have been modified to interpret the one
+ different redirect reason of "no-answer" properly and set the
+ ISDN reason code 2 of "no reply". (closes issue #15033) Reported
+ by: steinwej
+
+2009-10-09 21:01 +0000 [r223333] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 |
+ kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10
+ lines Initiate T.38 switchover when acting as called party,
+ regardless of FAX direction. SendFAX() and ReceiveFAX() can be
+ given options to indicate whether they should act as the calling
+ or called party; this mode should be used to decide whether to
+ initiate a switchover to T.38, not the direction that the FAX
+ transfer will take place. (closes issue #16039) Reported by:
+ jamicque ........
+
+2009-10-09 18:53 +0000 [r223286] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /: Merged revisions 223273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct
+ 2009) | 14 lines Merged revisions 223225 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct
+ 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING
+ when originating calls. (closes issue #15104) Reported by:
+ nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
+ (license 96) Tested by: nblasgen, mnicholson ........
+ ................
+
+2009-10-09 18:29 +0000 [r223257] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct
+ 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri,
+ 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c
+ ........ ................
+
+2009-10-09 17:55 +0000 [r223208] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009)
+ | 16 lines Merged revisions 223205 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009)
+ | 10 lines fixes sip registration using authuser in user.conf
+ (closes issue #14954) Reported by: tornblad Tested by:
+ mmichelson, tornblad, dvossel ........ ................
+
+2009-10-09 17:27 +0000 [r223173] Matthew Nicholson <mnicholson@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct
+ 2009) | 8 lines Don't close the sqlite database when reloading.
+ Only close the database when unloading. (closes issue #15953)
+ Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by
+ frawd (license 610) Tested by: frawd ........
+
+2009-10-09 17:09 +0000 [r223089-223133] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 |
+ dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines
+ 'auth=' did not parse md5 secret correctly (closes issue #15949)
+ Reported by: ebroad Patches: authparsefix.patch uploaded by
+ ebroad (license 878) 15949_trunk.diff uploaded by dvossel
+ (license 671) Tested by: ebroad ........
+
+ * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 |
+ dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines
+ p->peerauth is always empty in transmit_register() When using
+ callbackextension or specifing the peer name in a registration
+ string, the peer's specific auth settings set by the "auth="
+ strings within the peer definition are not used by the
+ registration. Thanks to ebroad for reporting the issue and
+ providing the patch. (closes issue #15955) Reported by: ebroad
+ Patches: regauthfix.patch uploaded by ebroad (license 878)
+ ........
+
+2009-10-08 20:00 +0000 [r222883] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c,
+ /, main/file.c: Merged revisions 222880 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009)
+ | 51 lines Merged revisions 222878 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009)
+ | 44 lines Make filestream frame handling safer by isolating
+ frames before returning them. This patch is related to a number
+ of issues on the bug tracker that show crashes related to freeing
+ frames that came from a filestream. A number of fixes have been
+ made over time while trying to figure out these problems, but
+ there re still people seeing the crash. (Note that some of these
+ bug reports include information about other problems. I am
+ specifically addressing the filestream frame crash here.) I'm
+ still not clear on what the exact problem is. However, what is
+ _very_ clear is that we have seen quite a few problems over time
+ related to unexpected behavior when we try to use embedded frames
+ as an optimization. In some cases, this optimization doesn't
+ really provide much due to improvements made in other areas. In
+ this case, the patch modifies filestream handling such that the
+ embedded frame will not be returned. ast_frisolate() is used to
+ ensure that we end up with a completely mallocd frame. In
+ reality, though, we will not actually have to malloc every time.
+ For filestreams, the frame will almost always be allocated and
+ freed in the same thread. That means that the thread local frame
+ cache will be used. So, going this route doesn't hurt. With this
+ patch in place, some people have reported success in not seeing
+ the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609)
+ Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt
+ uploaded by russell (license 2) Tested by: aragon, russell
+ (closes issue #15817) Reported by: zerohalo Tested by: zerohalo
+ (closes issue #15845) Reported by: marhbere Review:
+ https://reviewboard.asterisk.org/r/386/ ........ ................
+
+2009-10-08 19:41 +0000 [r222874] David Vossel <dvossel@digium.com>
+
+ * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions
+ 222873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 |
+ dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines
+ fixes an ast_netsock_list memory leak. ABE-1998 Review:
+ https://reviewboard.asterisk.org/r/395/ ........
+
+2009-10-08 16:51 +0000 [r222695-222802] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500
+ (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009)
+ | 12 lines Fix memory leak if chan_misdn config parameter is
+ repeated. Memory leak when the same config option is set more
+ than once in an misdn.conf section. Why must this be considered?
+ Templates! Defining a template with default port options and
+ later adding to or overriding some of them. Patches:
+ memleak-misdn.patch JIRA ABE-1998 ........ ................
+
+ * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500
+ (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009)
+ | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf:
+ astdtmf must be set to "yes". With "no", buffer loss does not
+ occur. The translated frame "f2" when passing through
+ ast_dsp_process() is not freed whenever it is not used further in
+ process_ast_dsp(). Then in the end it is never ever freed.
+ Patches: translate.patch JIRA ABE-1993 ........ ................
+
+2009-10-07 18:06 +0000 [r222549] Jason Parker <jparker@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 222548 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct
+ 2009) | 5 lines Remove 'keepstats' queue option from sample
+ config, as it's no longer used.
+ https://reviewboard.asterisk.org/r/115/ (closes issue #15820)
+ Reported by: kshumard ........
+
+2009-10-07 18:00 +0000 [r222547] Sean Bright <sean@malleable.com>
+
+ * funcs/func_strings.c: Fix merge error.
+
+2009-10-07 17:45 +0000 [r222544] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009)
+ | 14 lines Merged revisions 222542 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009)
+ | 8 lines crash on transfer handle_invite_replaces() attempts to
+ uplock a pvt's owner channel without first verifing that it
+ exists. (issue #16027) ........ ................
+
+2009-10-06 23:59 +0000 [r222354-222466] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500
+ (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009)
+ | 8 lines Add missing unlock(s) in dahdi_read (two cases in
+ trunk, and 1.6.2) (closes issue #15683) Reported by: alecdavis
+ ........ ................
+
+ * channels/chan_dahdi.c: Fix potential crash when entire span
+ request is received. The variable index used in this scenario for
+ accessing the dahdi_pvts was wrong and was most likely copied
+ from the several other places it is used correctly. (closes issue
+ #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch
+ uploaded by tsearle (license 373)
+
+ * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009)
+ | 9 lines Fix 222298 (crash during destruction of second channel
+ when variable set with setvar). I mistakenly reasoned that setvar
+ would be used on all channels. Since it can be set per channel,
+ give each dahdi channel a copy of the variable. (related to
+ #15899) ........
+
+2009-10-06 19:41 +0000 [r222311] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c, res/res_config_pgsql.c, /: Merged revisions
+ 222309 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 |
+ tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10
+ lines Change schema query to involve the use of an optional
+ schema parameter. This change is done in such a way as to allow
+ the driver to continue to function with older databases which
+ don't have these features. (closes issue #16000) Reported by:
+ jamicque Patches: 20091002__issue16000.diff.txt uploaded by
+ tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+
+2009-10-06 19:27 +0000 [r222304] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009)
+ | 9 lines Fix crash during destruction of second channel when
+ variable set with setvar. The setvar line in chan_dahdi.conf is
+ shared among all the channels, so make sure to only free the
+ resources only when the last channel is destroyed. (closes issue
+ #15899) Reported by: tzafrir ........
+
+2009-10-06 19:22 +0000 [r222289] Tilghman Lesher <tlesher@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 222273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 |
+ tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines
+ When we call a gosub routine, the variables should be scoped to
+ avoid contaminating the caller. This affected the ~~EXTEN~~ hack,
+ where a subroutine might have changed the value before it was
+ used in the caller. Patch by myself, tested by ebroad on
+ #asterisk ........
+
+2009-10-06 Leif Madsen <lmadsen@digium.com>
+
+ * Released Asterisk 1.6.2.0-rc3
+
+2009-10-06 01:39 +0000 [r222113-222187] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c,
+ res/res_clialiases.c, /, channels/chan_sip.c,
+ funcs/func_dialgroup.c, include/asterisk/astobj2.h,
+ res/res_phoneprov.c: Merged revisions 222176 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct
+ 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05
+ Oct 2009) | 20 lines Fix ao2_iterator API to hold references to
+ containers being iterated. See Mantis issue for details of what
+ prompted this change. Additional notes: This patch changes the
+ ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
+ instead of a macro, with a name that fits our naming policy;
+ also, it is now necessary to call ao2_iterator_destroy() on any
+ iterator that has been created. Currently this only releases the
+ reference to the container being iterated, but in the future this
+ could also release other resources used by the iterator, if the
+ iterator implementation changes to use additional resources.
+ (closes issue #15987) Reported by: kpfleming Review:
+ https://reviewboard.asterisk.org/r/383/ ........ ................
+
+ * configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c,
+ configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 222110
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05
+ Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be
+ supportable via configuration option. Many T.38 endpoints
+ incorrectly send the maximum IFP frame size they can accept as
+ the T38FaxMaxDatagram value in their SDP, when in fact this value
+ is supposed to be the maximum UDPTL payload size (datagram size)
+ they can accept. If the value they supply is small enough (a
+ commonly supplied value is '72'), T.38 UDPTL transmissions will
+ likely fail completely because the UDPTL packets will not have
+ enough room for a primary IFP frame and the redundancy used for
+ error correction. If this occurs, the Asterisk UDPTL stack will
+ emit log messages warning that data loss may occur, and that the
+ value may need to be overridden. This patch extends the
+ 't38pt_udptl' configuration option in sip.conf to allow the
+ administrator to override the value supplied by the remote
+ endpoint and supply a value that allows T.38 FAX transmissions to
+ be successful with that endpoint. In addition, in any SIP call
+ where the override takes effect, a debug message will be printed
+ to that effect. This patch also removes the T38FaxMaxDatagram
+ configuration option from udptl.conf.sample, since it has not
+ actually had any effect for a number of releases. In addition,
+ this patch cleans up the T.38 documentation in sip.conf.sample
+ (which incorrectly documented that T.38 support was passthrough
+ only). (issue #15586) Reported by: globalnetinc ........
+
+2009-10-02 17:35 +0000 [r222032] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500
+ (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
+ Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
+ memcpy. ........ ................
+
+2009-10-02 17:01 +0000 [r221923-221974] Tilghman Lesher <tlesher@digium.com>
+
+ * main/astobj2.c, /: Merged revisions 221971 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009)
+ | 9 lines Merged revisions 221970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009)
+ | 2 lines Ensure the result of the hash function is positive.
+ Negative array offsets suck. ........ ................
+
+ * /, main/logger.c: Merged revisions 221920 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 |
+ tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines
+ Initialize a variable that we check immediately upon startup.
+ (closes issue #15973) Reported by: atis ........
+
+2009-10-02 01:35 +0000 [r221879] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
+ Merged revisions 221844 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009)
+ | 33 lines Merged revisions 221769 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
+ | 26 lines Occasionally losing use of B channels in chan_misdn. I
+ have not been able to reproduce the problem of losing channels.
+ However, I have seen in the code a reentrancy problem that might
+ give these symptoms. The reentrancy patch does several things: 1)
+ Guards B channel and B channel structure allocation. 2) Makes the
+ B channel structure find routines more precise in locating
+ records. 3) Never leave a B channel allocated if we received
+ cause 44. The last item may cause temporary outgoing call
+ problems, but they should clear when the line becomes idle.
+ (closes issue #15490) Reported by: slutec18 Patches:
+ issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
+ Reported by: FabienToune Patches:
+ issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: FabienToune, rmudgett, slutec18 ........
+ ................
+
+2009-10-02 00:07 +0000 [r221744-221780] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions
+ 221777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009)
+ | 9 lines Merged revisions 221776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
+ | 2 lines Fix a bunch of off-by-one errors ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 |
+ tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines
+ Revision 220906 (a merge from 1.4) was not merged correctly,
+ causing a problem with non-dynamic peers. ........
+
+2009-10-01 19:35 +0000 [r221698] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 |
+ dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines
+ outbound tls connections were not defaulting to port 5061 (closes
+ issue #15854) Reported by: dvossel Patches:
+ sip_port_config_trunk.diff uploaded by dvossel (license 671)
+ Tested by: dvossel ........
+
+2009-10-01 16:57 +0000 [r221660] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221554,221589 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu,
+ 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE
+ constructs when it's just TRUE or FALSE. ................ r221589
+ | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9
+ lines Merged revisions 221588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
+ 2009) | 2 lines Use unsigned ints for portinuri flags. ........
+ ................
+
+2009-10-01 16:25 +0000 [r221622] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged
+ revisions 221592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 |
+ kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12
+ lines Remove ability to control T.38 FAX error correction from
+ udptl.conf. chan_sip has had the ability to control T.38 FAX
+ error correction mode on a per-peer (or global) basis for a
+ couple of releases now, which is where it should have been all
+ along. This patch removes the ability to configure it in
+ udptl.conf, but issues a warning if the user tries to do, telling
+ them to look at sip.conf.sample for how to configure it now. For
+ any SIP peers that are T.38 enabled in sip.conf, there is already
+ a default for FEC error correction even if the user does not
+ specify any mode, so this change will not turn off error
+ correction by default, it will have the same default value that
+ has been in the udptl.conf sample file. ........
+
+2009-09-30 23:07 +0000 [r221477-221485] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 |
+ mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2
+ lines Cleaned up merge from r221432 ........
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 221432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep
+ 2009) | 17 lines Merged revisions 221360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
+ 2009) | 10 lines Fix SRV lookup and Request-URI generation in
+ chan_sip. This patch adds a new field "portinuri" to the sip
+ dialog struct and the sip peer struct. That field is used during
+ RURI generation to determine if the port should be included in
+ the RURI. It is also used in some places to determine if an SRV
+ lookup should occur. (closes issue #14418) Reported by: klaus3000
+ Tested by: klaus3000, mnicholson Review:
+ https://reviewboard.asterisk.org/r/369/ ........ ................
+
+2009-09-30 21:46 +0000 [r221371-221472] Matthias Nick <mnick@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 221436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 |
+ mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines
+ Prevents from division by zero ........
+
+ * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
+ revisions 221368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) |
+ 23 lines Merged revisions 221153,221157,221303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
+ 2 lines check bounds - prevents for buffer overflow ........
+ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
+ 8 lines added a new dialplan function 'CSV_QUOTE' and changed the
+ cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
+ Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
+ mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
+ 30 Sep 2009) | 2 lines changed the prototype definition of
+ csv_quote ........ ................
+
+2009-09-30 19:15 +0000 [r221304] Terry Wilson <twilson@digium.com>
+
+ * configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c,
+ include/asterisk/rtp.h: Merged revisions 221266 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009)
+ | 32 lines Merged revisions 221086 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
+ | 25 lines Change the SSRC by default when our media stream
+ changes Be default, change SSRC when doing an audio stream
+ changes Asterisk doesn't honor marker bit when reinvited to
+ already-bridged RTP streams,resulting in far-end stack discarding
+ packets with "old" timestamps that areactually part of a new
+ stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
+ a reinvite, unless the 'constantssrc' is set to true in sip.conf.
+ The original issue reported to Digium support detailed the
+ following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
+ Application Server Call comes in fromITSP, Asterisk dials the app
+ server which sends a re-invite back toAsterisk--not to negotiate
+ to send media directly to the ITSP, but to indicatethat it's
+ changing the stream it's sending to Asterisk. The app
+ servergenerates a new SSRC, sequence numbers, timestamps, and
+ sets the marker bit on the new stream. Asterisk passes through
+ the teimstamp of the new stream, butdoes not reset the SSRC,
+ sequence numbers, or set the marker bit. When the timestamp on
+ the new stream is older than the timestamp on the originalstream,
+ the ITSP (which doesn't know there has been any change) discards
+ the newframes because it thinks they are too old. This patch
+ addresses this by changing the SSRC on a stream update unless
+ constantssrc=true is set in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/374/ ........ ................
+
+2009-09-30 16:57 +0000 [r221204] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 221201 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009)
+ | 14 lines Merged revisions 221200 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
+ | 7 lines Avoid a potential NULL dereference. (closes issue
+ #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
+ uploaded by tilghman (license 14) Tested by: kobaz ........
+ ................
+
+2009-09-30 14:57 +0000 [r221089] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep
+ 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a()
+ option. We require box numbers, not names as the documentation
+ implies. (issue #14740) Reported by: pj Patches:
+ __20090729-app_voicemail-documentation.patch uploaded by lmadsen
+ (license 10) Tested by: seanbright, lmadsen ........
+
+2009-09-30 04:41 +0000 [r221027-221047] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_lock.c: Recorded merge of revisions 221044 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29
+ Sep 2009) | 8 lines Allow locks to be inherited through a
+ masquerade without causing starvation. (closes issue #14859)
+ Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
+ by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
+ uploaded by tilghman (license 14) Tested by: atis, tilghman
+ ........
+
+ * include/asterisk/smdi.h, include/asterisk/optional_api.h
+ (removed), apps/app_voicemail.c, include/asterisk/agi.h,
+ include/asterisk/monitor.h: Remove optional_api from 1.6.2
+ branch, since it is not currently working. This is a blocking
+ issue for the 1.6.2 release. (closes issue #15914) Reported by:
+ mbeckwell Branch:
+ http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162
+ Tested by: mbeckwell
+
+ * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009)
+ | 16 lines Merged revisions 220873 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
+ | 9 lines Reduce CPU usage related to building a peer merely for
+ devicestates. This fixes a 100% CPU problem in the SIP driver,
+ found by profiling the driver while the problem was occurring.
+ (closes issue #14309) Reported by: pkempgen Patches:
+ 20090924__issue14309.diff.txt uploaded by tilghman (license 14)
+ Tested by: pkempgen, vrban ........ ................
+
+2009-09-29 20:24 +0000 [r220905-220934] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
+ spyee is masqueraded and chanspy_ds_chan_fixup() is called with
+ the channel locked. (closes issue #15965) Reported by: atis
+ Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: atis
+
+ * /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep
+ 2009) | 5 lines Fix options 'm' and 's'. They were swapped in the
+ code. Also document the fact that app_confbridge does not
+ automatically answer the channel. (closes issue #15964) Reported
+ by: shrift ........
+
+2009-09-29 17:06 +0000 [r220836] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009)
+ | 12 lines Make deletion of temporary greetings work properly
+ with IMAP_STORAGE When imapgreetings was set to yes, the message
+ was being deleted but wasn't actually being expunged. When
+ imapgreetings was set to no, the file based message was not being
+ deleted at all. All good now! (closes issue #14949) Reported by:
+ noahisaac Patches: vm_tempgreeting_removal.patch uploaded by
+ noahisaac (license 748), modified by me ........
+
+2009-09-28 19:13 +0000 [r220725] Sean Bright <sean@malleable.com>
+
+ * /, Makefile.rules: Merged revisions 220721 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep
+ 2009) | 10 lines Merged revisions 220717 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
+ 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
+ explicitly pass -O0 to the compiler so we override any default
+ optimization levels for a particular install. ........
+ ................
+
+2009-09-28 19:11 +0000 [r220722] Jeff Peeler <jpeeler@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 |
+ jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines
+ Fix building of registration entry in build_peer when using
+ callbackextension Check for remotesecret option was
+ unintentionally always true, which therefore caused the secret
+ option to never be used. Thanks to dvossel for pointing out the
+ exact fix. (closes issue #15943) Reported by: tpsast ........
+
+2009-09-27 20:45 +0000 [r220632] Michiel van Baak <michiel@vanbaak.info>
+
+ * funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009)
+ | 3 lines add name argument for the CALLERID dialplan function to
+ the xml documentation. Pointed out to me on IRC by snuff-home.
+ Thanks ........
+
+2009-09-26 15:12 +0000 [r220589] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009)
+ | 2 lines Allow AES to compile, when OpenSSL is not present.
+ ........
+
+2009-09-24 20:38 +0000 [r220369] David Vossel <dvossel@digium.com>
+
+ * main/tcptls.c, /: Merged revisions 220365 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 |
+ dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines
+ fixes tcptls_session memory leak caused by ref count error
+ (closes issue #15939) Reported by: dvossel Review:
+ https://reviewboard.asterisk.org/r/375/ ........
+
+2009-09-24 19:42 +0000 [r220292] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged
+ revisions 220289 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009)
+ | 13 lines Merged revisions 220288 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
+ | 6 lines Implicitly sending a progress signal breaks some
+ applications. Call Progress() in your dialplan if you explicitly
+ want progress to be sent. (Reverts change 216430, closes issue
+ #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
+ list
+ http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
+ ........ ................
+
+2009-09-24 18:22 +0000 [r220103-220221] Sean Bright <sean@malleable.com>
+
+ * Makefile, /: Merged revisions 220217 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep
+ 2009) | 9 lines Merged revisions 220213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
+ 2009) | 1 line Resolve parallel build warnings. Reported by Klaus
+ Darilion on the asterisk-dev mailing list. ........
+ ................
+
+ * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400
+ (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu,
+ 24 Sep 2009) | 2 lines Remove the remaining bashisms in the
+ Makefile/mkpkgconfig ........ ................
+
+2009-09-24 08:43 +0000 [r220031] Michiel van Baak <michiel@vanbaak.info>
+
+ * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200
+ (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009)
+ | 7 lines mkpkgconfig does not need bash so make it use /bin/sh
+ This fixes building on all systems that don't have bash at
+ /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
+ #asterisk-dev ........ ................
+
+2009-09-24 07:45 +0000 [r219989] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_directory.c, /: Merged revisions 219987 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009)
+ | 8 lines Fix two possible crashes, one only in 1.6.1 and one in
+ 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg
+ Patches: 20090914__issue15739.diff.txt uploaded by tilghman
+ (license 14) 20090922__issue15739.diff.txt uploaded by tilghman
+ (license 14) Tested by: DLNoah, jeffg ........
+
+2009-09-22 21:48 +0000 [r219821] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500
+ (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009)
+ | 10 lines When IMAP variables were changed during a reload,
+ Voicemail did not use the new values. This change introduces a
+ configuration version variable, which ensures that connections
+ with the old values are not reused but are allowed to expire
+ normally. (closes issue #15934) Reported by: viniciusfontes
+ Patches: 20090922__issue15934.diff.txt uploaded by tilghman
+ (license 14) Tested by: viniciusfontes ........ ................
+
+2009-09-21 17:01 +0000 [r219722] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500
+ (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
+ Sep 2009) | 3 lines Reverting merge 219520. This change was not
+ necessary. ........ ................
+
+2009-09-20 18:21 +0000 [r219669] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/file.c: Merged revisions 219654 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009)
+ | 15 lines Merged revisions 219653 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
+ | 8 lines Really stop the stream, when ast_closestream() is
+ called. (closes issue #15129) Reported by: bmh Patches:
+ 20090918__issue15129.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/372/ ........
+ ................
+
+2009-09-19 03:14 +0000 [r219590] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219587 | russell | 2009-09-18 21:59:52 -0500
+ (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009)
+ | 6 lines Make sure the iax_pvt exists before dereferencing it.
+ This fixes the latest crash posted on issue 15609. (issue #15609)
+ ........ ................
+
+2009-09-18 23:21 +0000 [r219452-219521] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500
+ (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009)
+ | 9 lines iax2 frame double free The iax frame's retrans sched id
+ was written over right before iax2_frame_free was called. In
+ iax2_frame_free that retrans id is used to delete the sched item.
+ By writing over the retrans field before the sched item could be
+ deleted, it was possible for a retransmit to occur on a freed
+ frame. ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009)
+ | 20 lines Merged revisions 219450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
+ | 14 lines via-header branches not updated correctly on INVITE
+ INVITE requests must always contain a new unique branch id. When
+ a new branch id is created for an INVITE, the dialog's
+ invite_branch variable must be updated so CANCEL requests use the
+ correct branch id. (closes issue #15262) Reported by: maniax
+ Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
+ (license 608) invite_new_branch_trunk.diff uploaded by dvossel
+ (license 671) Tested by: maniax, dvossel ........
+ ................
+
+2009-09-18 13:57 +0000 [r219415] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009)
+ | 6 lines Missing value setting line for maxsecs/maxmessage
+ (closes issue #15696) Reported by: fhackenberger Patches:
+ maxsecs.patch uploaded by fhackenberger (license 592) ........
+
+2009-09-17 22:38 +0000 [r219376] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 |
+ dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines
+ fixes deadlock when performing directed pickup w Invite/replaces
+ (closes issue #15340) Reported by: lmsteffan Patches:
+ deadlock.patch uploaded by lmsteffan (license 779) Tested by:
+ lmsteffan ........
+
+2009-09-17 22:37 +0000 [r219370] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep
+ 2009) | 12 lines Merged revisions 219320 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
+ 2009) | 6 lines Send a 100 Trying response when we detect a
+ spiral. This was problematic during spiral tests at SIPit...
+ along with some other things as well. ........ ................
+
+2009-09-17 22:06 +0000 [r219307] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009)
+ | 27 lines Merged revisions 219303 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
+ | 21 lines INVITE w/Replaces deadlock fix This patch cleans up
+ the locking logic in chan_sip.c's handle_invite_replaces()
+ function as well as making use of ast_do_masquerade() rather than
+ forcing the masquerade on an ast_read(). The code had several
+ redundant unlocks that would result in 'freed more times than
+ we've locked!' errors. I cleaned these up as well as moving all
+ the unlock logic to the end of the function. This patch should
+ also resolve the issue people were having with the replacecall
+ channel never being unlocked with one legged calls. (closes issue
+ #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
+ uploaded by dvossel (license 671) Tested by: irroot, dvossel
+ Review: https://reviewboard.asterisk.org/r/371/ ........
+ ................
+
+2009-09-17 19:58 +0000 [r219267] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 <lmadsen@digium.com>
+
+ * Released Asterisk 1.6.2.0-rc2
+
+2009-09-17 15:38 +0000 [r219194] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h,
+ include/asterisk/channel.h: Merged revisions 219139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500
+ (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
+ 2009) | 10 lines Prevent a potential race condition and crash
+ when hanging up a channel by removing the channel from the
+ channel list before begining channel tear down. This fix may
+ potentially cause problems with CDR backends that access the
+ channel a CDR is associated with via the channel list. This fix
+ makes the channel unavabile at the time when the CDR backend is
+ invoked. This has been documented in include/asterisk/cdr.h.
+ (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/362/ ........
+ ................
+
+2009-09-16 23:52 +0000 [r219063] Tilghman Lesher <tlesher@digium.com>
+
+ * main/config.c, configs/extensions.conf.sample, /: Merged
+ revisions 219061 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009)
+ | 15 lines Merged revisions 219023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
+ | 8 lines Properly deal with quotes in the arguments of '#exec'
+ includes. (closes issue #15583) Reported by: pkempgen Patches:
+ 20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+ 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+ 169) Tested by: pkempgen ........ ................
+
+2009-09-16 19:40 +0000 [r218938] David Brooks <dbrooks@digium.com>
+
+ * main/pbx.c, /: Merged revisions 218868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009)
+ | 20 lines Merged revisions 218867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
+ | 13 lines Fixes CID pattern matching behavior to mirror that of
+ extension pattern matching. Pattern matching for extensions uses
+ a type of scoring system, giving values for specificity to each
+ character in the pattern. Unfortunately, this is done character
+ by character, in order. This does lead to some less specific
+ patterns being first in line for matching, but it will usually
+ get the job done. This patch merely brings CID matching to the
+ same level as extension matching. This patch does not attempt to
+ tackle the problem shared by extension matching. (closes issue
+ #14708) Reported by: klaus3000 ........ ................
+
+2009-09-16 19:29 +0000 [r218937] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 |
+ mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12
+ lines Reverse order of args to fread. This way, we don't always
+ write a null byte into byte 1 of the buffer (closes issue #15905)
+ Reported by: ebroad Patches: freadfix.patch uploaded by ebroad
+ (license 878) Tested by: ebroad ........
+
+2009-09-16 19:25 +0000 [r218934] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 |
+ file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On
+ TCP and TLS connections do not attempt to stop retransmission of
+ the packet internally. This was preventing responses from being
+ properly processed because the packet was not being found causing
+ handle_response to return prematurely. ........
+
+2009-09-16 13:38 +0000 [r218802] Russell Bryant <russell@digium.com>
+
+ * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
+ revisions 218799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009)
+ | 16 lines Merged revisions 218798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
+ | 9 lines Remove the IAXy firmware from Asterisk. The firmware
+ can now be found on downloads.digium.com, where the rest of our
+ binary downloads live. This was the last part of our Asterisk
+ tarballs that was considered non-free by Debian. :-) (closes
+ issue #15838) Reported by: paravoid ........ ................
+
+2009-09-15 22:46 +0000 [r218733] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500
+ (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009)
+ | 6 lines If the user enters the same password as before, don't
+ signal an error when the change does nothing. (closes issue
+ #15492) Reported by: cbbs70a Patches:
+ 20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2009-09-15 19:24 +0000 [r218688] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 |
+ dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines
+ upward bound checking for port string to int conversion ........
+
+2009-09-15 16:18 +0000 [r218590] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep
+ 2009) | 15 lines Merged revisions 218578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
+ 2009) | 8 lines Send request contact header field with response
+ to registrer queries instead of the address of record. (closes
+ issue #14438) Reported by: ravindrad Patches: regquerypatch
+ uploaded by ravindrad (license 684) Tested by: ravindrad ........
+ ................
+
+2009-09-15 16:06 +0000 [r218582] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_followme.c, /: Merged revisions 218579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009)
+ | 16 lines Merged revisions 218577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
+ | 9 lines Ensure FollowMe sets language in channels it creates.
+ Also, not in the original bug report, but related fields are
+ accountcode and musicclass, and the inheritance of datastores.
+ (closes issue #15372) Reported by: Romik Patches:
+ 20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+ Tested by: cervajs ........ ................
+
+2009-09-15 15:59 +0000 [r218576] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 218430 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500
+ (Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
+ | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
+ crash in do_monitor. After talking to rmudgett about some of his
+ recent iflist locking changes, it was determined that the only
+ place that would destroy a channel without being explicitly to do
+ so was in handle_init_event. The loop to walk the interface list
+ has been modified to wait to destroy the channel until the
+ dahdi_pvt of the channel to be destroyed is no longer needed.
+ (closes issue #15378) Reported by: samy ........ ................
+
+2009-09-15 15:42 +0000 [r218507-218575] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 |
+ mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4
+ lines Use a better method of ensuring null-termination of the
+ buffer while reading the SDP when using TCP. ........
+
+ * /, channels/chan_sip.c: Merged revisions 218499,218504 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue,
+ 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent
+ over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53
+ -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP
+ socket is null-terminated. ........
+
+2009-09-15 15:05 +0000 [r218503] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/Makefile, /: Merged revisions 218500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep
+ 2009) | 9 lines Merged revisions 218497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
+ 2009) | 1 line Use proper hostname for downloading sound files.
+ ........ ................
+
+2009-09-14 19:49 +0000 [r218364] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile, apps/app_voicemail.c, /,
+ configs/voicemail.conf.sample: Merged revisions 218361 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500
+ (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
+ | 4 lines Don't say "Please try again" if we don't give the user
+ another chance to try again. (issue #15055, SWP-129) Reported by:
+ jthurman ........ ................
+
+2009-09-14 18:18 +0000 [r218300] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: Merged revisions 218295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 |
+ file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do
+ not attempt to add a parking extension if an error occurred while
+ reading the configuration. ........
+
+2009-09-14 15:20 +0000 [r218238] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 218224 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500
+ (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
+ 2009) | 8 lines Ensure we don't pickup ourselves when doing
+ pickup by exten. (closes issue #15100) Reported by: lmsteffan
+ Patches: (modified) pickup.patch uploaded by lmsteffan (license
+ 779) ........ ................
+
+2009-09-13 22:12 +0000 [r218219] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0
+ that annoys gcc This memset doesn't write beyond the end of the
+ buffer. (tmpbuf has size of 4). Merged revisions 218184 via
+ svnmerge from http://svn.digium.com/svn/asterisk/trunk
+
+2009-09-13 05:59 +0000 [r218151] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 218150 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1
+ line get rid of mfcr2 monitor thread condition, is problematic
+ ........
+
+2009-09-11 06:00 +0000 [r217926-218055] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 218050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 |
+ tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines
+ Check the origination priority for more matches, not the current
+ priority. Found by Pavel Troller on the -dev list. ........
+
+ * apps/app_queue.c, /: Merged revisions 217990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009)
+ | 10 lines Merged revisions 217989 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
+ | 3 lines Don't ring another channel, if there's not enough time
+ for a queue member to answer. (Fixes AST-228) ........
+ ................
+
+ * channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /,
+ channels/chan_sip.c: Merged revisions 217916 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 |
+ tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines
+ Make calltoken support work with realtime users and peers.
+ ........
+
+2009-09-10 21:21 +0000 [r217821] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 217807 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500
+ (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009)
+ | 22 lines IAX2 encryption regression The IAX2 Call Token
+ security patch inadvertently broke the use of encryption due to
+ the reorganization of code in the socket_process() function. When
+ encryption is used, an incoming full frame must first be
+ decrypted before the information elements can be parsed. The
+ security release mistakenly moved IE parsing before decryption in
+ order to process the new Call Token IE. To resolve this,
+ decryption of full frames is once again done before looking into
+ the frame. This involves searching for an existing callno,
+ checking the pvt to see if encryption is turned on, and
+ decrypting the packet before the internal fields of the full
+ frame are accessed. (closes issue #15834) Reported by: karesmakro
+ Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
+ (license 671) Tested by: dvossel, karesmakro Review:
+ https://reviewboard.asterisk.org/r/355/ ........ ................
+
+2009-09-10 19:56 +0000 [r217739] mnick <mnick@localhost>:
+
+ * res/res_musiconhold.c, /: Merged revisions 217730 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) |
+ 17 lines Sets the correct musicclass after an announcement
+ (closes issue #15279) Reported by: mbeckwell Patches: patch.txt
+ uploaded by mnick (license ) Tested by: mnick (closes issue
+ #15832) Reported by: mbeckwell Patches: patch.txt uploaded by
+ mnick (license 874) Tested by: mnick ........
+
+2009-09-10 18:40 +0000 [r217665] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 216805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 |
+ oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
+ Since it's possible to have more than 999 calls, I'm changing the
+ call counter roof to something higher. ........
+
+2009-09-10 18:19 +0000 [r217647] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 217638 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 |
+ tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines
+ Verify support for wide ODBC character types before using them.
+ (closes issue #15870) Reported by: nic_bellamy ........
+
+2009-09-10 15:14 +0000 [r217632] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 217524 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1
+ line ast_log replaced for ast_verbose in MFCR2 event
+ notifications ........
+
+2009-09-10 12:09 +0000 [r217594] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 |
+ oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines
+ Include ActionID in all events that are responsed to AMI Action
+ SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy
+ Patches: manager_SIPshowregistry_actionid.patch uploaded by nic
+ bellamy (license 299) ........
+
+2009-09-09 20:37 +0000 [r217519] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
+
+ * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc
+ 4.4 has more strict rules for aliasing. It doesn't like a struct
+ sockaddr_in pointer pointing to a struct sockaddr. So we make it
+ a union. Merged revisions 217445 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk
+
+2009-09-09 10:58 +0000 [r217369] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 |
+ oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not
+ having any TLS session to write to is a serious XMIT_ERROR.
+ ........
+
+2009-09-08 22:20 +0000 [r217299] Sean Bright <sean@malleable.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 |
+ seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4
+ lines Fix compilation of app_meetme. Reported by ebroad in
+ #asterisk-bugs ........
+
+2009-09-08 20:33 +0000 [r217217] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009)
+ | 14 lines Merged revisions 217156 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
+ | 7 lines When MOH is playing on the channel, announcements sent
+ through the conference are not heard. (closes issue #14588)
+ Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
+ tilghman ........ ................
+
+2009-09-08 16:39 +0000 [r217077] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 217074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 |
+ kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9
+ lines Ensure that the default autoconf CFLAGS are not used. A
+ recent change to the configure script that allows the user to
+ specify CFLAGS and/or LDFLAGS to the script had the unfortunate
+ side effect of letting autoconf's default CFLAGS (-g -O2) feed in
+ to the rest of the build system, thereby overriding the
+ DONT_OPTIMIZE setting in menuselect. That problem is now
+ corrected. ........
+
+2009-09-08 15:36 +0000 [r217036] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_limit.c: Merged revisions 217033 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 |
+ tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines
+ Remove what appears to be an unnecessary define. (closes issue
+ #15851) Reported by: tzafrir ........
+
+2009-09-08 14:27 +0000 [r216994] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 |
+ dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
+ caller id number empty parse_uri was not being given the correct
+ scheme's, as a result, uri parsing did not parse the username
+ correctly. One of the side effects of this is an empty caller id.
+ (closes issue #15839) Reported by: ebroad Patches:
+ blank_cidv2.patch uploaded by ebroad (license 878)
+ parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
+ ebroad, dvossel ........
+
+2009-09-07 16:43 +0000 [r216647-216845] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 |
+ oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
+ Make sure we reset global_exclude_static at channel reload
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 |
+ oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If
+ there is no session timer in the INVITE, set it to default value
+ (not unset minimum = -1) Patch by oej closes issue #15621
+ Reported by: fnordian Tested by: atis ........
+
+ * CHANGES, UPGRADE.txt: Add docs
+
+ * configs/sip.conf.sample, apps/app_playback.c, main/pbx.c, /,
+ channels/chan_sip.c, apps/app_disa.c: Merged revisions 216438 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre,
+ 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
+ lines Make apps send PROGRESS control frame for early media and
+ fix too early media issue in SIP The issue at hand is that some
+ legacy (dying) PBX systems send empty media frames on PRI links
+ *before* any call progress. The SIP channel receives these frames
+ and by default signals 183 Session progress and starts sending
+ media. This will cause phones to play silence and ignore the
+ later 180 ringing message. A bad user experience. The fix is
+ twofold: - We discovered that asterisk apps that support early
+ media ("noanswer") did not send any PROGRESS frame to indicate
+ early media. Fixed. - We introduce a setting in chan_sip so that
+ users can disable any relay of media frames before the outbound
+ channel actually indicates any sort of call progress. In 1.4,
+ 1.6.0 and 1.6.1, this will be disabled for backward
+ compatibility. In later versions of Asterisk, this will be
+ enabled. We don't assume that it will change your Asterisk phone
+ experience - only for the better. We encourage third-party
+ application developers to make sure that if they have
+ applications that wants to send early media, add a PROGRESS
+ control frame transmission to make sure that all channel drivers
+ actually will start sending early media. This has not been the
+ default in Asterisk previous to this patch, so if you got
+ inspiration from our code, you need to update accordingly. Sorry
+ for the trouble and thanks for your support. This code has been
+ running for a few months in a large scale installation (over 250
+ servers with PRI and/or BRI links to old PBX systems). That's no
+ proof that this is an excellent patch, but, well, it's tested :-)
+ ........ ................
+
+2009-09-04 19:42 +0000 [r216598] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 |
+ dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines
+ sip peer matching by address only with TCP/TLS This patch removes
+ the contact header matching logic and adds logic to match all
+ tcp/tls connections by ip only Review:
+ https://reviewboard.asterisk.org/r/354/ ........
+
+2009-09-04 19:32 +0000 [r216597] Sean Bright <sean@malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep
+ 2009) | 1 line Use ast_free() instead of free(). ........
+
+2009-09-04 17:53 +0000 [r216550-216553] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009)
+ | 2 lines Fix trunk breakage. ........
+
+ * UPGRADE-1.6.txt, main/pbx.c, /: Merged revisions 216547 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04
+ Sep 2009) | 3 lines Enable turning off the application delimiter
+ warning with the 'dontwarn' option. Suggested on the -dev list,
+ and implemented in an alternate way by me. ........
+
+2009-09-04 15:11 +0000 [r216469-216509] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, main/utils.c: Merged revisions 216506 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009)
+ | 9 lines Merged revisions 216435 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make asterisk compile under devmode with DEBUG_THREADS
+ enabled on OpenBSD ........ ................
+
+ * /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make sure canlog is set so we can compile with
+ DEBUG_THREADS enabled on OpenBSD ........
+
+2009-09-04 13:57 +0000 [r216267-216436] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 |
+ russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines
+ Do not treat every SIP peer as if they were configured with
+ insecure=port. There was a problem in the function responsible
+ for doing peer matching by IP address and port number such that
+ during the second pass for checking for a peer configured with
+ insecure=port, it would end up treating every peer as if it had
+ been configured that way. These changes fix the logic in the peer
+ IP and port comparison callback to handle insecure=port checking
+ properly. This problem was introduced when SIP peers were
+ converted to astobj2. Many thanks to dvossel for noticing this
+ while working on another peer matching issue. ........
+
+ * doc/IAX2-security.txt (added), /: Merged revisions 216264 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216264 | russell | 2009-09-04 05:48:44 -0500
+ (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216263 | russell | 2009-09-04 05:48:00 -0500
+ (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
+ Sep 2009) | 2 lines Add a plain text version of the IAX2 security
+ document. ........ ................ ................
+
+2009-09-04 06:14 +0000 [r216225] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/astobj2.c, /: Merged revisions 216222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 |
+ mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
+ make sure 'start' is always initialized. Makes asterisk compile
+ with --enable-dev-mode ........
+
+2009-09-03 19:44 +0000 [r216014-216099] Russell Bryant <russell@digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009)
+ | 16 lines Merged revisions 216085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216085 | russell | 2009-09-03 14:36:46 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
+ ........ ................ ................
+
+ * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216009 | russell | 2009-09-03 13:45:54 -0500
+ (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216008 | russell | 2009-09-03 13:44:58 -0500
+ (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
+ Sep 2009) | 2 lines Add IAX2 security document related to
+ AST-2009-006. ........ ................ ................
+
+2009-09-03 18:42 +0000 [r216007] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c, channels/iax2-parser.c, main/astobj2.c,
+ configs/iax.conf.sample, include/asterisk/acl.h,
+ channels/iax2-parser.h, /, include/asterisk/astobj2.h,
+ channels/iax2.h, main/acl.c: Merged revisions 215955 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009)
+ | 6 lines Merge code associated with AST-2009-006 (closes issue
+ #12912) Reported by: rathaus Tested by: tilghman, russell,
+ dvossel, dbrooks ........
+
+2009-09-03 14:21 +0000 [r215887-215929] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 215891 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 |
+ oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add
+ known internal IP address when autodomain=yes (closes issue
+ #14573) Reported by: pj Patches: sip-internip-autodomain1.diff
+ uploaded by mnicholson (license 96) modified by oej Tested by: pj
+ ........
+
+ * main/rtp.c, channels/chan_sip.c: Fix bad reports in "sip show
+ channelstats". Not directly mergeable in svn trunk, needs more
+ tests, therefore committed directly to 1.6.2. (closes issue
+ #15819) Reported by: klaus3000 Patches:
+ asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded
+ by klaus3000 (license 65) Tested by: klaus3000, oej
+
+2009-09-03 06:02 +0000 [r215841] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/manager_1_1.txt, /: Merged revisions 215838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215838 |
+ mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines
+ Document that SIPshowpeer and SKINNYshowline now include the
+ configured parkinglot in their response. Prodded by snuff-work on
+ #asterisk-dev IRC channel ........
+
+2009-09-03 03:44 +0000 [r215802] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215801 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 |
+ tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines
+ Default the callback extension to "s". This is a regression.
+ (closes issue #15764) Reported by: elguero Change-type: bugfix
+ ........
+
+2009-09-03 00:34 +0000 [r215795] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215758 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009)
+ | 25 lines Merged revisions 215682 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009)
+ | 18 lines Re-send non-100 provisional responses to prevent
+ cancellation From section 13.3.1.1 of RFC 3261: If the UAS
+ desires an extended period of time to answer the INVITE, it will
+ need to ask for an "extension" in order to prevent proxies from
+ canceling the transaction. A proxy has the option of canceling a
+ transaction when there is a gap of 3 minutes between responses in
+ a transaction. To prevent cancellation, the UAS MUST send a
+ non-100 provisional response at every minute, to handle the
+ possibility of lost provisional responses. (closes issue #11157)
+ Reported by: rjain Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/315/ ........ ................
+
+2009-09-02 21:46 +0000 [r215683] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215681 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 |
+ dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines
+ port string to int conversion using sscanf There are several
+ instances where a port is parsed from a uri or some other source
+ and converted to an int value using atoi(), if for some reason
+ the port string is empty, then a standard port is used. This
+ logic is used over and over, so I created a function to handle it
+ in a safer way using sscanf(). ........
+
+2009-09-02 21:37 +0000 [r215647-215680] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_sip.c, channels/chan_skinny.c: Merged revisions
+ 215665 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 |
+ mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines
+ add Parkinglot info to sip show peer <foo> and skinny show line
+ <foo> If we had this from the start, debugging the 'parking not
+ using configured parkinglot' bug would have been easier. ........
+
+ * /, main/features.c: Merged revisions 215622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215622 |
+ mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines
+ - lock channel before looking for a channel variable - Init the
+ parkings list member of struct parkinglot. Thanks Sean for the
+ explanation why this should be here. ........
+
+2009-09-02 18:52 +0000 [r215569-215570] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/Makefile, main/app.c: Merged revisions 215567 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r215567 | tilghman | 2009-09-02 13:37:25 -0500 (Wed, 02
+ Sep 2009) | 9 lines Close up to the soft open file limit (same on
+ Linux, but varies drastically on OS X). Also, a Makefile fix for
+ Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches:
+ 20090901__issue14542.diff.txt uploaded by tilghman (license 14)
+ Tested by: jtodd, tilghman Change-type: bugfix ........
+
+ * /, channels/chan_sip.c: Merged revisions 215222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 |
+ tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines
+ Fix register such that lines with a transport string, but without
+ an authuser, parse correctly. (AST-228) ........
+
+2009-09-02 17:35 +0000 [r215523] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215522 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 |
+ dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines
+ SIP uri parsing cleanup Now, the scheme passed to parse_uri can
+ either be a single scheme, or a list of schemes ',' delimited.
+ This gets rid of the whole problem of having to create two
+ buffers and calling parse_uri twice to check for separate
+ schemes. Review: https://reviewboard.asterisk.org/r/343/ ........
+
+2009-09-02 16:35 +0000 [r215512] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_skinny.c: Merged revisions 215479 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009)
+ | 3 lines like in chan_sip's sip_new skinny should copy the
+ configured parkinglot from a line to the newly created channel.
+ This makes callparking honor the configured parkinglot for skinny
+ lines as well. ........
+
+2009-09-02 16:09 +0000 [r215467] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 215466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 |
+ dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines
+ SIP support for keep-alive event keep-alive events are used by
+ Sipura/Linksys for NAT keepalive. There currently don't appear to
+ be any problems with NAT, but everytime a keep-alive event is
+ received, Asterisk responds with a "489 Bad event". This error
+ may indicate to a user that NAT problems exist just because this
+ even is not supported. Now, rather than respond with an error,
+ the packet is consumed and a "200 ok" is sent just to indicate we
+ received the packet. (issue #15084) Patches:
+ chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
+ ........
+
+2009-09-02 16:07 +0000 [r215465] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, channels/chan_sip.c: Merged revisions 215462 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 |
+ mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12
+ lines Honor configured parkinglot when parking and retrieving
+ parked calls Thank oej for pointing out the fact that sip_new did
+ not copy parkinglot from the peer into the newly created channel.
+ (closes issue #15538) Reported by: gracedman Patches:
+ 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak
+ (license 7) With mod by me to also fix callparking as well (this
+ uploaded patch only fixed retrieving a parked call) Tested by:
+ gracedman, mvanbaak ........
+
+2009-09-02 01:49 +0000 [r215376] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * /, apps/app_softhangup.c: Merged revisions 215338 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500
+ (Tue, 01 Sep 2009) | 18 lines Merged revisions 215270 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009)
+ | 12 lines Use strrchr() so SoftHangup will correctly truncate
+ multi-hyphen channel names In general channel names are in the
+ form Foo/Bar-Z, but the channel name could have multiple hyphens
+ and look like Foo/B-a-r-Z. Use strrchr to truncate the channel
+ name at the last hyphen. (closes issue #15810) Reported by:
+ dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard
+ (license 733) ........ ................
+
+2009-09-01 20:00 +0000 [r215165] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c, /: Merged revisions 215161 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215161 |
+ kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3
+ lines Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS
+ frames are properly decoded. ........
+
+2009-08-31 16:22 +0000 [r214822-214960] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 214945 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500
+ (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009)
+ | 7 lines Also unlock the "other" channel, when returning, due to
+ glare. (closes issue #15787) Reported by: tim_ringenbach Patches:
+ chan_local.diff uploaded by tim ringenbach (license 540) Tested
+ by: tim_ringenbach ........ ................
+
+ * Makefile, /: Merged revisions 214898 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214898 |
+ tilghman | 2009-08-30 17:10:35 -0500 (Sun, 30 Aug 2009) | 2 lines
+ Force Darwin on ppc platforms to compile with a target level that
+ supports aliasing. ........
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c: Merged revisions 214819 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 |
+ tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines
+ If lua is detected with the lua5.1 prefix (or not), adjust the
+ include path accordingly. Based upon feedback to a release
+ announcement on the -users list. See
+ http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html
+ ........
+
+2009-08-29 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.6.2.0-rc1 released.
+
+2009-08-28 20:17 +0000 [r214707] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 214702 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009)
+ | 15 lines Merged revisions 214701 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009)
+ | 8 lines Modify comment to be a bit more accurate. We have kept
+ this comment around long enough, that it's pretty clear that
+ we're keeping the code, because changing the code would require a
+ pretty fundamental architectural shift. We've also taken
+ criticism in some quarters, because it was believed that it was
+ referring to the code being nasty. No, the code isn't nasty, just
+ the operation itself is rather odd. Fixed for eternity (probably
+ not). ........ ................
+
+2009-08-28 20:05 +0000 [r214700] Kevin P. Fleming <kpfleming@digium.com>
+
+ * makeopts.in, Makefile, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 214696 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 |
+ kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9
+ lines Ensure that CFLAGS and/or LDFLAGS provided to configure
+ script are preserved. Cross-compilation environments want to
+ provide 'defaults' for compiler and linker options, and
+ frequently do this by specifying CFLAGS and LDFLAGS in the
+ environment or as command-line arguments to the configure script.
+ This patch modifies the configure script and Makefile to preserve
+ these settings and ensure they are used in the build process.
+ ........
+
+2009-08-28 18:43 +0000 [r214653] Mark Michelson <mmichelson@digium.com>
+
+ * /, include/asterisk/sched.h: Merged revisions 214650 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug
+ 2009) | 3 lines Fix some incorrect documentation of sched_thread
+ functions. ........
+
+2009-08-27 21:49 +0000 [r214202-214521] Tilghman Lesher <tlesher@digium.com>
+
+ * autoconf/libcurl.m4 (added), /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 214518 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009)
+ | 14 lines Merged revisions 214517 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009)
+ | 7 lines Use autoconf to detect libcurl, as this enables
+ cross-compilation checks, something we didn't allow before.
+ (closes issue #15714) Reported by: pprindeville Patches:
+ 20090813__issue15714.diff.txt uploaded by tilghman (license 14)
+ Tested by: pprindeville ........ ................
+
+ * main/manager.c, /: Merged revisions 214514 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214514 |
+ tilghman | 2009-08-27 16:26:37 -0500 (Thu, 27 Aug 2009) | 7 lines
+ Ensure that we check for the special value
+ CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by:
+ a_villacis Patches:
+ asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch
+ uploaded by a villacis (license 660) (Plus a few of my own, to
+ catch the remaining places within manager.c where it could have
+ been a problem) ........
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
+ 214466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009)
+ | 9 lines Merged revisions 214436 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009)
+ | 2 lines One more build system change, to make the descriptions
+ look better, if we have better information. ........
+ ................
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 214360 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500
+ (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009)
+ | 3 lines Make autoheader descriptions render correctly in our
+ autoconfig.h file. (Figured out while working with issue #14906)
+ ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 214199 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 |
+ tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines
+ Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue
+ #15362) Reported by: klaus3000 Patches:
+ chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license
+ 65) ........
+
+2009-08-26 16:39 +0000 [r214196] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, /: Merged revisions 214195 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009)
+ | 25 lines Merged revisions 214194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009)
+ | 19 lines ast_write() ignores ast_audiohook_write() results In
+ ast_write(), if a channel has a list of audiohooks, those lists
+ are written to and the resulting frame is what ast_write() should
+ continue with. The problem was the returned audiohook frame was
+ not being handled at all, and the original frame passed into it
+ did not contain the mixed audio, so essentially audio was being
+ lost. One result of this was chan_spy's whisper mode no longer
+ worked. To complicate the issue, frames passed into ast_write may
+ either be a single frame, or a list of frames. So, as the list of
+ frames is processed in the audiohook_write, the returned frames
+ had to be added to a new list. (closes issue #15660) Reported by:
+ corruptor Tested by: dvossel ........ ................
+
+2009-08-25 22:43 +0000 [r213903-214155] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 214152 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 |
+ tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines
+ Not all versions of gnu-linux use glibc, which contains iconv.
+ Some (especially embedded systems) don't have iconv at all.
+ (closes issue #15169) Reported by: pprindeville ........
+
+ * /, main/say.c: Merged revisions 214071 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009)
+ | 17 lines Merged revisions 214068-214069 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009)
+ | 6 lines Fix pronunciation of German dates. (closes issue
+ #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
+ by Benjamin Kluck (license 803) ........ r214069 | tilghman |
+ 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should
+ always compile before committing... ........ ................
+
+ * /, pbx/pbx_dundi.c: Merged revisions 213975 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213975 |
+ tilghman | 2009-08-25 01:51:12 -0500 (Tue, 25 Aug 2009) | 6 lines
+ DUNDILOOKUP function in 1.6 should use comma delimiters. (closes
+ issue #15322) Reported by: chappell Patches:
+ dundilookup-0015322.patch uploaded by chappell (license 8)
+ ........
+
+ * main/pbx.c, /: Merged revisions 213971 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009)
+ | 14 lines Merged revisions 213970 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009)
+ | 7 lines Improve error message by informing user exactly which
+ function is missing a parethesis. (closes issue #15242) Reported
+ by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
+ dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
+ loloski (license 68) ........ ................
+
+ * Makefile, /: Merged revisions 213904 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213904 |
+ tilghman | 2009-08-24 21:54:07 -0500 (Mon, 24 Aug 2009) | 6 lines
+ The DTD should be installed in the same path as the rest of the
+ XML documentation. (closes issue #15344) Reported by: tzafrir
+ Patches: makefile_appdocs_dtd.diff uploaded by tzafrir (license
+ 46) ........
+
+ * Makefile, /: Merged revisions 213900 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r213900 | tilghman | 2009-08-24 21:41:17 -0500 (Mon, 24 Aug 2009)
+ | 11 lines Merged revisions 213899 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009)
+ | 4 lines Use the default runlevels for Debian derivatives,
+ instead of making up our own. (closes issue #14730) Reported by:
+ pkempgen ........ ................
+
+2009-08-24 16:49 +0000 [r213836] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 213833 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009)
+ | 14 lines Fix storage of greetings when using IMAP_STORAGE The
+ store macro was not getting called preventing storage of IMAP
+ greetings at all. This has been corrected along with fixing
+ checking if the imapgreetings option is turned on to store the
+ greeting in IMAP. Lastly, the attachment filename was incorrectly
+ using the full path instead of just the basename, which was
+ causing problems with retrieval of the greeting. (closes issue
+ #14950) Reported by: noahisaac (closes issue #15729) Reported by:
+ lmadsen ........
+
+2009-08-24 04:54 +0000 [r213791] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 213790 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213790 | moy | 2009-08-24 00:46:28 -0400 (Mon, 24 Aug 2009) | 1
+ line improve handling of openr2_chan_disconnect_call API failure,
+ unlikely, but happened on openr2 library bug ........
+
+2009-08-21 22:54 +0000 [r213739] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 213738 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 |
+ tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines
+ Clarifying comments in sip_register, and removing a dead section
+ ........
+
+2009-08-21 22:23 +0000 [r213721] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 213716 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 |
+ dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines
+ Register request line contains wrong address when user domain and
+ register host differ (closes issue #15539) Reported by:
+ Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by
+ Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel
+ (license 671) Tested by: Nick_Lewis, dvossel ........
+
+2009-08-21 21:44 +0000 [r213698] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 213697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug
+ 2009) | 12 lines Ensure that realtime mailboxes properly report
+ status on subscription. This patch modifies app_voicemail's
+ response to mailbox status subscriptions (via the internal event
+ system) to ensure that a subscription triggers an explicit poll
+ of the mailbox, so the subscriber can get an immediate cached
+ event with that status. Previously, the cache was only populated
+ with the status of non-realtime mailboxes. (closes issue #15717)
+ Reported by: natmlt ........
+
+2009-08-21 21:12 +0000 [r213636] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 213635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 |
+ dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines
+ fixes sip register parsing when user@domain is used (issue
+ #15008) (issue #15672) ........
+
+2009-08-21 16:55 +0000 [r213563] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk.h, /: Merged revisions 213560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009)
+ | 14 lines Merged revisions 213559 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009)
+ | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files.
+ (closes issue #15698) Reported by: slavon Patches:
+ 20090817__issue15698.diff.txt uploaded by tilghman (license 14)
+ Tested by: slavon, tilghman ........ ................
+
+2009-08-21 16:06 +0000 [r213497] Jason Parker <jparker@digium.com>
+
+ * /, configs/queues.conf.sample: Merged revisions 213494 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r213494 | qwell | 2009-08-21 11:04:21 -0500
+ (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) |
+ 5 lines Clarify queues.conf comments to specify that variables
+ should be set in the dialplan. (closes issue #15755) Reported by:
+ trendboy ........ ................
+
+2009-08-21 04:25 +0000 [r213475-213481] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 213454 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213454 | moy | 2009-08-21 00:09:26 -0400 (Fri, 21 Aug 2009) | 1
+ line increment the mfcr2 monitor count when clearing the call
+ request ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 213216 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213216 | moy | 2009-08-19 23:26:59 -0400 (Wed, 19 Aug 2009) | 1
+ line fixed bug caused by calling ast_request without calling
+ ast_call on an R2 channel, ie, CHANISAVAIL ........
+
+2009-08-21 03:53 +0000 [r213453] Terry Wilson <twilson@digium.com>
+
+ * main/loader.c, /: Merged revisions 213450 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 |
+ twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines
+ Make LOAD_ORDER actually work ........
+
+2009-08-20 21:50 +0000 [r213413] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 213404 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009)
+ | 12 lines Fix greeting retrieval from IMAP Properly check for
+ the current voicemail state and if it doesn't exist, create it.
+ (closes issue #14597) Reported by: wtca Patches: 14597_v2.patch
+ uploaded by mmichelson (license 60) Tested by: jpeeler ........
+
+2009-08-20 20:37 +0000 [r213350] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/features.c: Merged revisions 213327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 |
+ mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7
+ lines Fix a crash by checking the proper pointer for validity
+ before deferencing it. (closes issue #15751) Reported by: atis
+ Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license
+ 242) ........
+
+2009-08-19 22:41 +0000 [r213182] Jason Parker <jparker@digium.com>
+
+ * main/alaw.c, main/ulaw.c, /: Merged revisions 213179 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) |
+ 5 lines Fix compile when certain G711 menuselect options are
+ enabled. (closes issue #15697) Reported by: slavon ........
+
+2009-08-19 21:25 +0000 [r213128] David Vossel <dvossel@digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 213113 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500
+ (Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009)
+ | 8 lines Fixes memory leak caused by incorrectly freeing
+ mixmonitor (closes issue #15699) Reported by: edantie Patches:
+ mixmonitor.patch uploaded by edantie (license 862) ........
+ ................
+
+2009-08-19 21:22 +0000 [r213095-213117] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
+ 213098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 |
+ tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines
+ Better parsing for the "register" line Allows characters that are
+ otherwise used as delimiters to be used within certain fields
+ (like the secret). (closes issue #15008, closes issue #15672)
+ Reported by: tilghman Patches: 20090818__issue15008.diff.txt
+ uploaded by tilghman (license 14) Tested by: lmadsen, tilghman
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 213093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 |
+ tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines
+ If we have realtime caching enabled, 'sip reload' must purge
+ users/peers, even if the config files haven't changed. (closes
+ issue #12869) Reported by: bcnit Patches:
+ 20090819__issue12869__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: lasko ........
+
+2009-08-19 15:35 +0000 [r213047] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 213046 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r213046 |
+ russell | 2009-08-19 10:32:18 -0500 (Wed, 19 Aug 2009) | 4 lines
+ Don't blow up on a NULL cdr. Reported in #asterisk-dev. ........
+
+2009-08-18 20:34 +0000 [r212931-212944] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: Merged revisions 212939 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212939 |
+ kpfleming | 2009-08-18 15:33:34 -0500 (Tue, 18 Aug 2009) | 1 line
+ Remove some accidentally-committed properties. ........
+
+ * sounds/Makefile, doc/tex/asterisk.tex, CREDITS, /,
+ UPGRADE-1.4.txt, sounds/sounds.xml, build_tools/prep_tarball:
+ Merged revisions 212922 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212922 |
+ kpfleming | 2009-08-18 15:29:37 -0500 (Tue, 18 Aug 2009) | 6
+ lines Convert this branch to Opsound music-on-hold. For more
+ details:
+ http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
+ ........
+
+2009-08-18 19:28 +0000 [r212866] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/extconfig.conf.sample: Merged revisions 212857 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18
+ Aug 2009) | 4 lines Make the default extconfig.conf match entries
+ with the sample res_mysql.conf. This eliminates a future source
+ of possible confusion with the configuration of 1.6.1 and higher.
+ ........
+
+2009-08-18 16:56 +0000 [r212769] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, /: Merged revisions 212758 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500
+ (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18
+ Aug 2009) | 1 line Removed some deadwood and added some doxygen
+ comments. ........ ................
+
+2009-08-18 16:41 +0000 [r212767] Sean Bright <sean@malleable.com>
+
+ * main/manager.c, /: Merged revisions 212764 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r212764 | seanbright | 2009-08-18 12:38:36 -0400 (Tue, 18 Aug
+ 2009) | 18 lines Merged revisions 212763 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug
+ 2009) | 11 lines Delay the creation of temporary files until we
+ have a valid manager command to handle. Without this patch,
+ asterisk creates a temporary file before determining if the
+ specified command is valid. If invalid, we weren't properly
+ cleaning up the file. (closes issue #15730) Reported by: zmehmood
+ Patches: M15730.diff uploaded by junky (license 177) Tested by:
+ zmehmood ........ ................
+
+2009-08-17 20:01 +0000 [r212631] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 212627 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009)
+ | 4 lines Check the return value of opendir(3), or we may crash.
+ (closes issue #15720) Reported by: tobias_e ........
+
+2009-08-17 18:56 +0000 [r212580-212584] Sean Bright <sean@malleable.com>
+
+ * /, channels/chan_agent.c: Merged revisions 212581 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug
+ 2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in
+ chan_agent. (closes issue #15668) Reported by: davidw ........
+
+ * main/logger.c: Merged revisions 212574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212574 |
+ seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8
+ lines Correct the return value check for ast_safe_system. The
+ logic here was reversed as ast_safe_system returns -1 on error
+ and not on success. Fix suggested by reporter. (closes issue
+ #15667) Reported by: loic ........
+
+2009-08-17 16:52 +0000 [r212509] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/misdn_config.c, /: Merged revisions 212506 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500
+ (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009)
+ | 12 lines Fix segfault when reloading chan_misdn. If more ports
+ were specified than configured in misdn.conf a reload would crash
+ asterisk. The problem was the unconfigured port was using data
+ from the previously configured port. When the data for an
+ unconfigured port was freed a crash would result from the double
+ free. (closes issue #12113) Reported by: agupta Patches:
+ bug12113.patch uploaded by jpeeler (license 325) ........
+ ................
+
+2009-08-17 15:51 +0000 [r212434] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 212431 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500
+ (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix
+ uninitialized variable causing random MWI indications. (closes
+ issue #15727) Reported by: doda Patches: dahdi_changes.patch
+ uploaded by doda (license 853) ........ r212430 | rmudgett |
+ 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix
+ uninitialized variable. ........ ................
+
+2009-08-14 17:37 +0000 [r212250] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_curl.c, /: Merged revisions 212249 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212249 |
+ tilghman | 2009-08-14 12:36:40 -0500 (Fri, 14 Aug 2009) | 2 lines
+ Add SSL_VERIFYPEER, as requested on the -users list ........
+
+2009-08-13 15:47 +0000 [r212116] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 212113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 |
+ kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3
+ lines Ensure that T38FaxVersion is put into outgoing SDP in the
+ proper case. ........
+
+2009-08-13 13:56 +0000 [r212070] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 212067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 |
+ file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines
+ Check an actual populated variable when seeing if we need to do
+ video or not. ........
+
+2009-08-13 11:47 +0000 [r212030] Gavin Henry <ghenry@suretecsystems.com>
+
+ * contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, /: Merged revisions 212027 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r212027 | ghenry | 2009-08-13 12:37:12 +0100 (Thu, 13
+ Aug 2009) | 6 lines Fixed typo (closes issue #15710) Reported by:
+ suretec ........
+
+2009-08-12 23:16 +0000 [r211951-211959] Matthew Nicholson <mnicholson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 211957 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug
+ 2009) | 17 lines Merged revisions 211953 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug
+ 2009) | 10 lines This patch adds additional checking when
+ generating queue log TRANSFER events. The additional checks
+ prevent generation of false TRANSFER events in certain
+ situations. (closes issue #14536) Reported by: aragon Patches:
+ queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, mnicholson ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 211876 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 |
+ mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11
+ lines Make asterisk handle 423 Interval Too Short messages
+ better. This change uses separate values for the acceptable
+ minimum expiry provided by the 423 error and the expiry value
+ stored in the configuration file. Previously, the value pulled
+ from the configuration file would be overwritten. (closes issue
+ #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff
+ uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch
+ uploaded by Nick (license 657) Tested by: mnicholson ........
+
+2009-08-12 16:21 +0000 [r211785] Gavin Henry <ghenry@suretecsystems.com>
+
+ * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema,
+ contrib/scripts/asterisk.ldif, /: Merged revisions 211767 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r211767 | ghenry | 2009-08-12 17:00:46 +0100 (Wed, 12
+ Aug 2009) | 33 lines Added three new attributes and applied a
+ patch to res_config_ldap.c attributetype (
+ AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC
+ 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR
+ caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
+ attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC
+ 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR
+ caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
+ attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent'
+ DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch
+ SUBSTR caseIgnoreSubstringsMatch SYNTAX
+ 1.3.6.1.4.1.1466.115.121.1.15) and patch
+ fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725)
+ Reported by: macogeek Patches:
+ fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license
+ 863) Tested by: suretec ........
+
+2009-08-10 19:51 +0000 [r211580-211585] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
+ (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
+ Aug 2009) | 1 line Conversion specifiers, not format specifiers
+ ........ ................
+
+ * apps/app_queue.c, apps/app_talkdetect.c, agi/eagi-sphinx-test.c,
+ res/res_config_curl.c, channels/chan_usbradio.c,
+ channels/chan_misdn.c, res/snmp/agent.c, apps/app_sms.c,
+ apps/app_verbose.c, apps/app_stack.c, apps/app_mixmonitor.c,
+ main/asterisk.c, main/dsp.c, main/timing.c,
+ doc/CODING-GUIDELINES, funcs/func_speex.c, main/frame.c,
+ utils/muted.c, apps/app_meetme.c, apps/app_alarmreceiver.c,
+ cdr/cdr_pgsql.c, res/res_musiconhold.c, channels/chan_iax2.c,
+ apps/app_followme.c, main/enum.c, main/indications.c,
+ res/res_config_sqlite.c, channels/misdn_config.c, utils/frame.c,
+ main/cli.c, pbx/pbx_loopback.c, channels/chan_phone.c,
+ funcs/func_enum.c, res/res_smdi.c, channels/chan_skinny.c,
+ funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c,
+ res/res_config_ldap.c, apps/app_adsiprog.c,
+ funcs/func_dialplan.c, main/pbx.c, main/dnsmgr.c,
+ funcs/func_sprintf.c, funcs/func_timeout.c, channels/chan_sip.c,
+ apps/app_privacy.c, res/res_limit.c, apps/app_waitforsilence.c,
+ codecs/codec_speex.c, agi/eagi-test.c, apps/app_morsecode.c,
+ funcs/func_cut.c, channels/chan_oss.c, main/netsock.c,
+ apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
+ pbx/pbx_dundi.c, utils/extconf.c, pbx/pbx_config.c,
+ apps/app_chanspy.c, res/res_odbc.c, apps/app_voicemail.c,
+ apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, /,
+ apps/app_record.c, main/utils.c, cdr/cdr_adaptive_odbc.c,
+ res/res_http_post.c, main/config.c, res/ael/pval.c, main/cdr.c,
+ main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
+ main/manager.c, apps/app_setcallerid.c, apps/app_osplookup.c,
+ main/features.c, main/http.c, channels/xpmr/xpmr.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, res/res_config_pgsql.c,
+ channels/chan_agent.c, funcs/func_math.c, apps/app_waituntil.c,
+ apps/app_disa.c, main/acl.c, apps/app_originate.c,
+ channels/iax2-provision.c: AST-2009-005
+
+2009-08-10 14:15 +0000 [r211350] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
+ file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
+ retrieval of the port used for the video stream when adding SDP
+ to a SIP message. (closes issue #15121) Reported by: jsmith
+ ........
+
+2009-08-09 15:43 +0000 [r211235-211278] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/astfd.c: Merged revisions 211275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
+ | 9 lines Merged revisions 211274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
+ | 2 lines Small oops. Clear the flags which have been checked.
+ ........ ................
+
+ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
+ tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
+ Check for NULL frame, before dereferencing pointer. (closes issue
+ #15617) Reported by: rain ........
+
+2009-08-07 20:18 +0000 [r211122] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
+ | 11 lines Recorded merge of revisions 211112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
+ | 4 lines Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936) ........ ................
+
+2009-08-07 18:20 +0000 [r211051] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 211040 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
+ | 21 lines Merged revisions 211038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
+ | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
+ not the membername. This is a partial revert of revision 82590,
+ which was an attempted cleanup, but in reality, it broke
+ QUEUE_MEMBER_LIST, which has always been intended as a method by
+ which component interfaces could be queried from the queue.
+ Membername isn't useful here, because that field cannot be used
+ to obtain further information about the member. See the
+ documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+ QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+ member argument for further justification. (closes issue #15664)
+ Reported by: rain Patches: app_queue-queue_member_list.diff
+ uploaded by rain (license 327) ........ ................
+
+2009-08-07 13:10 +0000 [r210995] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /: Merged revisions 210992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
+ kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
+ lines Workaround broken T.38 endpoints that offer tiny
+ MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
+ the maximum IFP size that should be sent to them, rather than the
+ maximum packet payload size. If such an endpoint also requests
+ UDPRedundancy as the error correction mode, we'll end up
+ calculating a tiny maximum IFP size, so small as to be unusable.
+ This patch sets a lower bound on what we'll consider the remote's
+ maximum IFP size to be, assuming that endpoints that do this
+ really can accept larger packets than they've offered to accept.
+ (closes issue #15649) Reported by: dazza76 ........
+
+2009-08-06 21:47 +0000 [r210911-210917] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 210914 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
+ | 14 lines Merged revisions 210913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
+ | 7 lines Because channel information can be accessed outside of
+ the channel thread, we must lock the channel prior to modifying
+ it. (closes issue #15397) Reported by: caspy Patches:
+ 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy ........ ................
+
+ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
+ revisions 210908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
+ tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
+ Allow Gosub to recognize quote delimiters without consuming them.
+ (closes issue #15557) Reported by: rain Patches:
+ 20090723__issue15557.diff.txt uploaded by tilghman (license 14)
+ Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
+ ........
+
+2009-08-06 17:49 +0000 [r210820] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
+ file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
+ Accept additional T.38 reinvites after an initial one has been
+ handled. Discussion of this subject has yielded that it is not
+ actually acceptable to change T.38 parameters after the initial
+ reinvite but declining is harsh and can cause the fax to fail
+ when it may be possible to allow it to continue. This patch
+ changes things so that additional T.38 reinvites are accepted but
+ parameter changes ignored. This gives the fax a fighting chance.
+ (closes issue #15610) Reported by: huangtx2009 ........
+
+2009-08-05 20:43 +0000 [r210686] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
+ (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
+ | 14 lines Dialplan starts execution before the channel setup is
+ complete. * Issue 15655: For the case where dialing is complete
+ for an incoming call, dahdi_new() was asked to start the PBX and
+ then the code set more channel variables. If the dialplan hungup
+ before these channel variables got set, asterisk would likely
+ crash. * Fixed potential for overlap incoming call to erroneously
+ set channel variables as global dialplan variables if the
+ ast_channel structure failed to get allocated. * Added missing
+ set of CALLINGSUBADDR in the dialing is complete case. (closes
+ issue #15655) Reported by: alecdavis ........ ................
+
+2009-08-05 18:56 +0000 [r210565-210566] Leif Madsen <lmadsen@digium.com>
+
+ * /: Merged revisions 210564 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009)
+ | 19 lines Merged revisions 210563 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+ | 11 lines Update imapstorage.txt documentation. Updated the
+ imapstorage.txt documentation to reflect that issues with
+ c-client versions older than 2007 seem to cause crashing issues
+ that are not seen with more recent versions. Documentation has
+ been updated to reflect this. (closes issue #14496) Reported by:
+ vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks ........ ................
+
+ * doc/tex/imapstorage.tex: Merged revisions 210564 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500
+ (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+ | 11 lines Update imapstorage.txt documentation. Updated the
+ imapstorage.txt documentation to reflect that issues with
+ c-client versions older than 2007 seem to cause crashing issues
+ that are not seen with more recent versions. Documentation has
+ been updated to reflect this. (closes issue #14496) Reported by:
+ vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks ........ ................
+
+2009-08-04 14:55 +0000 [r210191-210241] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 210238 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug
+ 2009) | 16 lines Merged revisions 210237 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
+ 2009) | 10 lines Eliminate spurious compiler warnings from system
+ headers on *BSD platforms. Ensure that system headers located in
+ /usr/local/include are actually treated as system headers by the
+ compiler, and not as local headers which are subject to warnings
+ from the -Wundef compiler option and others. (closes issue
+ #15606) Reported by: mvanbaak ........ ................
+
+ * configs/sip.conf.sample, configs/skinny.conf.sample, main/rtp.c,
+ channels/chan_mgcp.c, doc/chan_sip-perf-testing.txt,
+ contrib/scripts/realtime_pgsql.sql, /, channels/chan_sip.c,
+ channels/chan_skinny.c, configs/mgcp.conf.sample,
+ doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt,
+ configs/res_ldap.conf.sample: Merged revisions 210190 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03
+ Aug 2009) | 11 lines Rename 'canreinvite' option to
+ 'directmedia', with backwards compatibility. It is clear from
+ multiple mailing list, forum, wiki and other sorts of posts that
+ users don't really understand the effects that the 'canreinvite'
+ config option actually has, and that in some cases they think
+ that setting it to 'no' will actually cause various other
+ features (T.38, MOH, etc.) to not work properly, when in fact
+ this is not the case. This patch changes the proper name of the
+ option to what it should have been from the beginning
+ ('directmedia'), but preserves backwards compatibility for
+ existing configurations. ........
+
+2009-08-01 11:33 +0000 [r209837-209906] Russell Bryant <russell@digium.com>
+
+ * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209887 | russell | 2009-08-01 06:29:25 -0500
+ (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
+ | 5 lines Resolve a valgrind warning about a read from
+ uninitialized memory. (issue #15396) Reported by: aragon ........
+ ................
+
+ * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r209839 | russell | 2009-08-01 06:02:07 -0500
+ (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009)
+ | 13 lines Modify how Playtones() is used in Milliwatt() to
+ resolve gain issue. When Milliwatt() was changed internally to
+ use Playtones() so that the proper tone was used, it introduced a
+ drop in gain in the output signal. So, use the playtones API
+ directly and specify a volume argument such that the output
+ matches the gain of the original Milliwatt() code. (closes issue
+ #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff
+ uploaded by russell (license 2) Tested by: rue_mohr ........
+ ................
+
+ * /, main/event.c: Merged revisions 209835 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 |
+ russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines
+ Fix ast_event_queue_and_cache() to actually do the cache() part.
+ (closes issue #15624) Reported by: ffossard Tested by: russell
+ ........
+
+2009-08-01 01:34 +0000 [r209816] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx/pbx_config.c, channels/misdn/isdn_lib.c, utils/frame.c,
+ main/pbx.c, /, main/Makefile, channels/misdn/ie.c: Merged
+ revisions 209760-209761 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul
+ 2009) | 13 lines Merged revisions 209759 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
+ 2009) | 7 lines Minor changes inspired by testing with latest
+ GCC. The latest GCC (what will become 4.5.x) has a few new
+ warnings, that in these cases found some either downright buggy
+ code, or at least seriously poorly designed code that could be
+ improved. ........ ................ r209761 | kpfleming |
+ 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert
+ accidental Makefile change. ................
+
+2009-07-31 22:01 +0000 [r209715] Russell Bryant <russell@digium.com>
+
+ * /, main/event.c: Merged revisions 209711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 |
+ russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines
+ Fix some places where ast_event_type was used instead of
+ ast_event_ie_type. ........
+
+2009-07-30 18:51 +0000 [r209594] David Brooks <dbrooks@digium.com>
+
+ * channels/chan_console.c, include/asterisk/abstract_jb.h,
+ apps/app_forkcdr.c, channels/chan_dahdi.c,
+ contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
+ codecs/lpc10/pitsyn.c: Merged revisions 209554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
+ dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
+ Fixes numerous spelling errors. Patch submitted by alecdavis.
+ (closes issue #15595) Reported by: alecdavis ........
+
+2009-07-30 14:40 +0000 [r209518] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 |
+ mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8
+ lines Fix a crash that can result if text codecs are allowed but
+ textsupport is disabled. (closes issue #15596) Reported by:
+ fabled Patches: sip-red.patch uploaded by fabled (license 448)
+ ........
+
+2009-07-28 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-beta4
+
+2009-07-28 00:19 +0000 [r209328] Tilghman Lesher <tlesher@digium.com>
+
+ * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
+ | 9 lines Merged revisions 209315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
+ | 2 lines Publish French extra sounds ........ ................
+
+2009-07-27 21:44 +0000 [r209265-209282] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
+ kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
+ lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
+ messages about T.38 negotiation in debug level 1 messages, clean
+ up some looping logic, and correct an improper use of ast_free()
+ for freeing an ast_frame. ........
+
+ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
+ kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
+ lines Make T.38 switchover in ReceiveFAX synchronous. In receive
+ mode, if the channel that ReceiveFAX is running on supports T.38,
+ we should *always* attempt to switch T.38, rather than listening
+ for an incoming CNG tone and only triggering on that. The channel
+ may be using a low-bitrate codec that distorts the CNG tone, the
+ sending FAX endpoint may not send CNG at all, or there could be a
+ variety of other reasons that we don't detect it, but in all
+ those cases if T.38 is available we certainly want to use it.
+ ........
+
+2009-07-27 20:58 +0000 [r209238] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, /: Merged revisions 209235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 |
+ mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5
+ lines Gracefully handle malformed RTP text packets. AST-2009-004
+ ........
+
+2009-07-27 20:33 +0000 [r209234] David Brooks <dbrooks@digium.com>
+
+ * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c,
+ channels/chan_vpb.cc, res/res_smdi.c, /,
+ include/asterisk/module.h, main/features.c, res/res_agi.c: Merged
+ revisions 209098 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
+ dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
+ Fixing typos. Replaces "recieved" with "received" and "initilize"
+ with "initialize" (closes issue #15571) Reported by: alecdavis
+ ........
+
+2009-07-27 20:23 +0000 [r209135-209222] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 209197 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
+ 2009) | 9 lines Honor channel's music class when using realtime
+ music on hold. (closes issue #15051) Reported by: alexh Patches:
+ 15051.patch uploaded by mmichelson (license 60) Tested by: alexh
+ ........
+
+ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+ 209132 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
+ 2009) | 24 lines Merged revisions 209131 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+ 2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+ if desired. There are some VoIP providers out there that will not
+ accept SDP offers with odd numbered UDPTL ports. While it is my
+ personal opinion that these VoIP providers are misinterpreting
+ RFC 2327, it really is not a big deal to play along with their
+ silly little games. Of course, since restricting UDPTL ports to
+ only even numbers reduces the range of available ports by half,
+ so the option to use only even port numbers is off by default. A
+ user can enable the behavior by setting use_even_ports=yes in
+ udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+ 15182.patch uploaded by mmichelson (license 60) Tested by:
+ CGMChris ........ ................
+
+2009-07-27 16:07 +0000 [r209063] David Brooks <dbrooks@digium.com>
+
+ * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing
+ typos "recieved" with "received". From issue #15360, forgot to
+ apply to trunk and other branches.
+
+2009-07-27 15:40 +0000 [r209059] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 209056 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
+ kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
+ lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
+ underscore-variants to sub-makes. During the recent Makefile
+ improvements I made, it seemed the 'make' was automatically
+ carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
+ I removed the explict export of them. However, there are some
+ circumstances where make does this, and some where it does not,
+ so I've brought them back to ensure they are always exported. I
+ also removed an extraneous double setting of _ASTLDFLAGS on *BSD
+ platforms. ........
+
+2009-07-27 01:23 +0000 [r208927] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, main/translate.c: Merged revisions
+ 208924 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009)
+ | 9 lines Merged revisions 208923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
+ | 2 lines Fix logic errors from 208746 ........ ................
+
+2009-07-26 14:07 +0000 [r208889] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208886 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26
+ Jul 2009) | 2 lines add OpenBSD to the install_prereq script
+ ........
+
+2009-07-25 12:31 +0000 [r208816-208853] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208848 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25
+ Jul 2009) | 2 lines libxml2-dev is needed as well by default.
+ ........
+
+ * main/cli.c, /, configs/cli_aliases.conf.sample: Merged revisions
+ 208813 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 |
+ mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10
+ lines add default alias reload to run module reload. Requiring
+ 'module reload' to reload everything, including core etc makes
+ russell very unhappy. The default configuration already loads the
+ 'friendly' aliases template. Added 'reload=module reload' to that
+ template. Also removed the comment in main/cli.c that reload
+ should come back. ........
+
+2009-07-25 06:26 +0000 [r208755] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_iax2.c, /, channels/chan_skinny.c,
+ main/translate.c: Merged revisions 208749 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009)
+ | 13 lines Merged revisions 208746 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
+ | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
+ trivial changes, but I did not know of any other way to fix the
+ "dereferencing type-punned pointer will break strict-aliasing
+ rules" error without creating a tmp variable in chan_skinny.
+ ........ ................
+
+2009-07-24 21:13 +0000 [r208695-208710] Russell Bryant <russell@digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 208709 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208709 |
+ russell | 2009-07-24 16:12:43 -0500 (Fri, 24 Jul 2009) | 2 lines
+ Remove trailing whitespace. ........
+
+ * main/cli.c, /: Merged revisions 208706 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208706 |
+ russell | 2009-07-24 15:54:37 -0500 (Fri, 24 Jul 2009) | 6 lines
+ Note that "reload" needs to be added back. I keep getting annoyed
+ at having to type "module reload" to reload everything, so I'm
+ adding a note that we need to add "reload" back. "module reload"
+ doesn't really make sense as the command to reload everything,
+ including the core. ........
+
+ * main/cli.c, /: Merged revisions 208693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208693 |
+ russell | 2009-07-24 15:25:23 -0500 (Fri, 24 Jul 2009) | 2 lines
+ Don't log a warning for something that does not affect operation.
+ ........
+
+2009-07-24 19:42 +0000 [r208664] Mark Michelson <mmichelson@digium.com>
+
+ * /: Fixing trunk-blocked property.
+
+2009-07-24 18:56 +0000 [r208596] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009)
+ | 14 lines Merged revisions 208592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
+ | 7 lines Do not log an ERROR if autoservice_stop() returns -1.
+ This does not indicate an error. A return of -1 just means that
+ the channel has been hung up. (reported in #asterisk-dev)
+ ........ ................
+
+2009-07-24 18:32 +0000 [r208591] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul
+ 2009) | 16 lines Merged revisions 208587 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
+ 2009) | 10 lines Only send a BYE when hanging up a channel that
+ is up. For cases where Asterisk sends an INVITE and receives a
+ non 2XX final response, Asterisk would follow the INVITE
+ transaction by immediately sending a BYE, which was unnecessary.
+ (closes issue #14575) Reported by: chris-mac ........
+ ................
+
+2009-07-24 15:06 +0000 [r208551] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+ Merged revisions 208548 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 |
+ kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8
+ lines Resolve a T.38 negotiation issue left over from the
+ udptl-updates merge. The udptl-updates branch that was merged
+ yesterday failed to properly send back T.38 SDP responses with
+ the correct error correction mode, if the incoming SDP from the
+ other end caused us to change error correction modes. This patch
+ corrects that situation. ........
+
+2009-07-24 14:39 +0000 [r208545] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/install_prereq, /: Merged revisions 208542 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24
+ Jul 2009) | 13 lines use aptitude for debian based systems The
+ function to check wether we need to install packages was using
+ dpkg-query which was gives wrong output on Debian 5 Also, the
+ apt-get has been replaced with aptitude because aptitude is now
+ the preferred way to handle packages on Debian (closes issue
+ #15570) Reported by: mvanbaak Patches:
+ 2009072400_installprereq-aptitude.diff uploaded by mvanbaak
+ (license 7) ........
+
+2009-07-23 22:31 +0000 [r208501] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/frame.h, main/rtp.c, main/channel.c,
+ main/udptl.c, main/frame.c, /, channels/chan_sip.c,
+ apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h: Merged
+ revisions 208464 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 |
+ kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46
+ lines Rework of T.38 negotiation and UDPTL API to address
+ interoperability problems Over the past couple of months, a
+ number of issues with Asterisk negotiating (and successfully
+ completing) T.38 sessions with various endpoints have been found.
+ This patch attempts to address many of them, primarily focused
+ around ensuring that the endpoints' MaxDatagram size is honored,
+ and in addition by ensuring that T.38 session parameter
+ negotiation is performed correctly according to the ITU T.38
+ Recommendation. The major changes here are: 1) T.38 applications
+ in Asterisk (app_fax) only generate/receive IFP packets, they do
+ not ever work with UDPTL packets. As a result of this, they
+ cannot be allowed to generate packets that would overflow the
+ other endpoints' MaxDatagram size after the UDPTL stack adds any
+ error correction information. With this patch, the application is
+ told the maximum *IFP* size it can generate, based on a
+ calculation using the far end MaxDatagram size and the active
+ error correction mode on the T.38 session. The same is true for
+ sending *our* MaxDatagram size to the remote endpoint; it is
+ computed from the value that the application says it can accept
+ (for a single IFP packet) combined with the active error
+ correction mode. 2) All treatment of T.38 session parameters as
+ 'capabilities' in chan_sip has been removed; these parameters are
+ not at all like audio/video stream capabilities. There are strict
+ rules to follow for computing an answer to a T.38 offer, and
+ chan_sip now follows those rules, using the desired parameters
+ from the application (or channel) that wants to accept the T.38
+ negotiation. 3) chan_sip now stores and forwards
+ ast_control_t38_parameters structures for tracking 'our' and
+ 'their' T.38 session parameters; this greatly simplifies
+ negotiation, especially for pass-through calls. 4) Since T.38
+ negotiation without specifying parameters or receiving the final
+ negotiated parameters is not very worthwhile, the AST_CONTROL_T38
+ control frame has been removed. A note has been added to
+ UPGRADE.txt about this removal, since any out-of-tree
+ applications that use it will no longer function properly until
+ they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
+ https://reviewboard.asterisk.org/r/310/ ........
+
+2009-07-23 19:36 +0000 [r208391] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul
+ 2009) | 24 lines Merged revisions 208386 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
+ 2009) | 17 lines Fix a problem where a 491 response could be sent
+ out of dialog. This generalizes the fix for issue 13849. The
+ initial fix corrected the problem that Asterisk would reply with
+ a 491 if a reinvite were received from an endpoint and we had not
+ yet received an ACK from that endpoint for the initial INVITE it
+ had sent us. This expansion also allows Asterisk to appropriately
+ handle an INVITE with authorization credentials if Asterisk had
+ not received an ACK from the previous transaction in which
+ Asterisk had responded to an unauthorized INVITE with a 407.
+ (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
+ uploaded by mmichelson (license 60) Tested by: klaus3000 ........
+ ................
+
+2009-07-23 19:25 +0000 [r208387] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500
+ (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009)
+ | 6 lines Only set the priindication setting when not performing
+ a reload (closes issue #14696) Reported by: fdecher ........
+ ................
+
+2009-07-23 16:30 +0000 [r208266-208320] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul
+ 2009) | 9 lines Merged revisions 208312 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
+ 2009) | 3 lines Remove inaccurate XXX comment. ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul
+ 2009) | 15 lines Merged revisions 208262 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
+ 2009) | 8 lines Properly handle 183 responses which do not
+ contain an SDP. (closes issue #15442) Reported by: ffloimair
+ Patches: 15442.patch uploaded by mmichelson (license 60) Tested
+ by: tkarl, ffloimair ........ ................
+
+2009-07-22 21:46 +0000 [r208116] Jason Parker <jparker@digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 |
+ qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines
+ Restore an int declaration on PPC platforms. This x is one crafty
+ little bugger... It was used for 2 different things (one of which
+ was only done on PPC) in 1.4. One of the uses were removed in
+ trunk, and with it went the declaration. (closes issue #14038)
+ Reported by: ffloimair ........
+
+2009-07-22 16:52 +0000 [r207949-208053] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_realtime.c: Merged revisions 208052 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r208052 |
+ tilghman | 2009-07-22 11:49:42 -0500 (Wed, 22 Jul 2009) | 7 lines
+ Clarify documentation on 'realtime update2' to show more than one
+ condition. (closes issue #15357) Reported by: snuffy Patches:
+ bug_fix_doc_update2.diff uploaded by snuffy (license 35)
+ (slightly modified by me) ........
+
+ * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500
+ (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009)
+ | 8 lines Force an error if a blank is passed to QUOTE (because
+ the documentation states the argument is not optional). This
+ change makes URIENCODE and QUOTE behave similarly, since the
+ documentation states that the argument is not optional, for both.
+ (closes issue #15439) Reported by: pkempgen Patches:
+ 20090706__issue15439.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+2009-07-21 22:23 +0000 [r207930] Russell Bryant <russell@digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 207925 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r207925 | russell | 2009-07-21 17:22:18 -0500 (Tue, 21 Jul 2009)
+ | 4 lines Note that we use tabs instead of spaces for
+ indentation. I'm surprised this was never actually in here...
+ ........
+
+2009-07-21 20:30 +0000 [r207785-207862] Jeff Peeler <jpeeler@digium.com>
+
+ * 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 r207638, this approach could
+ potentially block for an unacceptable amount of time.
+
+2009-07-21 14:32 +0000 [r207727] Mark Michelson <mmichelson@digium.com>
+
+ * 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 13:56 +0000 [r207685] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 04:51 +0000 [r207638] Jeff Peeler <jpeeler@digium.com>
+
+ * 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 22:14 +0000 [r207523] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 16:41 +0000 [r207364] Russell Bryant <russell@digium.com>
+
+ * 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 04:19 +0000 [r207327] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, /: Merged 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 03:50 +0000 [r207315] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c: 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 22:31 +0000 [r207226-207257] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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: Merged 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 19:40 +0000 [r207104-207159] Jeff Peeler <jpeeler@digium.com>
+
+ * 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 17:52 +0000 [r206869-207030] David Vossel <dvossel@digium.com>
+
+ * /, 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 [r206811] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 22:04 +0000 [r206770] David Vossel <dvossel@digium.com>
+
+ * /, 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 21:50 +0000 [r206765] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 20:20 +0000 [r206703] David Vossel <dvossel@digium.com>
+
+ * /, 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 <sip:123.123.123.123> 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 16:04 +0000 [r206639] Sean Bright <sean@malleable.com>
+
+ * 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 20:26 +0000 [r206598] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_meetme.c, contrib/scripts/meetme.sql: Merged
+ 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 19:49 +0000 [r206565] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 14:59 +0000 [r206389] Russell Bryant <russell@digium.com>
+
+ * 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 01:59 +0000 [r206373] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 23:27 +0000 [r206281] David Vossel <dvossel@digium.com>
+
+ * /, 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 [r206187] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 21:46 +0000 [r205986] David Vossel <dvossel@digium.com>
+
+ * /, 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 [r205942] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 17:54 +0000 [r205882] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 16:47 +0000 [r205841] David Vossel <dvossel@digium.com>
+
+ * /, 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 16:01 +0000 [r205781] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, apps/app_fax.c: Merged revisions 205780 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205780 |
+ kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11
+ lines Eliminate extraneous LOG_DEBUG messages generated by
+ app_fax. The transmit_audio() and transmit_t38() functions in
+ app_fax have processing loops that are supposed to wait for
+ frames to arrive on the channel and then handle them, but they
+ also have short timeouts so that the loops can have watchdog
+ timers and do other required processing. This commit changes the
+ loops to not actually call ast_read() and attempt to process the
+ returned frame unless a frame actually arrived, eliminating
+ hundreds of LOG_DEBUG messages and slightly improving
+ performance. ........
+
+2009-07-10 15:58 +0000 [r205779] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 [r205773] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, 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 23:56 +0000 [r205731] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 [r205699] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 16:19 +0000 [r205595-205603] David Vossel <dvossel@digium.com>
+
+ * 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 08:34 +0000 [r205535] Michiel van Baak <michiel@vanbaak.info>
+
+ * /, 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 22:15 +0000 [r205411-205413] David Vossel <dvossel@digium.com>
+
+ * include/asterisk/pbx.h, include/asterisk/devicestate.h,
+ main/pbx.c, /, main/devicestate.c: 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. ........
+ ................
+
+ * /, main/devicestate.c: Merged revisions 205410 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r205410 |
+ dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines
+ missing comma in devstatestring array ........
+
+2009-07-08 19:28 +0000 [r205353] Mark Michelson <mmichelson@digium.com>
+
+ * 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 18:22 +0000 [r205302] Jason Parker <jparker@digium.com>
+
+ * 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 18:18 +0000 [r205287] David Brooks <dbrooks@digium.com>
+
+ * /, 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 17:00 +0000 [r205223] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: oops, fixing build
+
+2009-07-08 16:55 +0000 [r205217] David Vossel <dvossel@digium.com>
+
+ * 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 16:30 +0000 [r205207] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 [r205148-205154] Russell Bryant <russell@digium.com>
+
+ * /, 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. ........
+
+ * include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c,
+ /, main/Makefile, res/res_crypto.c, main/ssl.c (added): 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 13:41 +0000 [r204951] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 22:06 +0000 [r204838] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 16:12 +0000 [r204711] David Vossel <dvossel@digium.com>
+
+ * 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 [r204611] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 18:55 +0000 [r204478] Jason Parker <jparker@digium.com>
+
+ * /, 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 [r204473] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 ........
+ ................
+
+2009-06-30 17:22 +0000 [r204442] Russell Bryant <russell@digium.com>
+
+ * configs/res_config_sqlite.conf (removed),
+ configs/res_config_sqlite.conf.sample (added), /: Merged
+ revisions 204440 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 |
+ russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines
+ Rename res_config_sqlite.conf to res_config_sqlite.conf.sample
+ (missing .sample). ........
+
+2009-06-29 22:53 +0000 [r204250-204304] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 09:55 +0000 [r203961] Russell Bryant <russell@digium.com>
+
+ * CHANGES, /: Merged revisions 203960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 |
+ russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines
+ Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt.
+ ........
+
+2009-06-27 01:24 +0000 [r203941] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 22:14 +0000 [r203857] Jeff Peeler <jpeeler@digium.com>
+
+ * 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 21:27 +0000 [r203782-203828] Russell Bryant <russell@digium.com>
+
+ * /, 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 [r203731] David Brooks <dbrooks@digium.com>
+
+ * 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 19:49 +0000 [r203715] Russell Bryant <russell@digium.com>
+
+ * 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 19:48 +0000 [r203712] David Vossel <dvossel@digium.com>
+
+ * 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 19:42 +0000 [r203709] Jeff Peeler <jpeeler@digium.com>
+
+ * 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 ........
+
+2009-06-26 19:38 +0000 [r203705] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c,
+ main/channel.c, main/frame.c, /, channels/chan_sip.c,
+ apps/app_fax.c: 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. ........
+
+2009-06-25 21:46 +0000 [r203445] David Vossel <dvossel@digium.com>
+
+ * 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 21:21 +0000 [r203400] Terry Wilson <twilson@digium.com>
+
+ * 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 21:08 +0000 [r203379] Russell Bryant <russell@digium.com>
+
+ * /, 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 [r203276] Jason Parker <jparker@digium.com>
+
+ * 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 16:08 +0000 [r203119] Russell Bryant <russell@digium.com>
+
+ * /, 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 21:27 +0000 [r203077] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 [r202970] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 18:11 +0000 [r202928] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 23:58 +0000 [r202842] Sean Bright <sean@malleable.com>
+
+ * doc/tex/cdrdriver.tex, /, doc/tex/billing.tex: Merged revisions
+ 202840-202841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r202840 |
+ seanbright | 2009-06-23 19:53:45 -0400 (Tue, 23 Jun 2009) | 1
+ line Remove some trailing whitespace before making content
+ changes. ........ r202841 | seanbright | 2009-06-23 19:57:07
+ -0400 (Tue, 23 Jun 2009) | 1 line Change some section names in
+ the CDR tex documentation. ........
+
+2009-06-23 22:47 +0000 [r202805] Russell Bryant <russell@digium.com>
+
+ * doc/tex/cdrdriver.tex, /: Merged revisions 202804 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r202804 | russell | 2009-06-23 17:47:26 -0500 (Tue, 23 Jun 2009)
+ | 2 lines Clean up section hierarchy for the CDR chapter.
+ ........
+
+2009-06-23 22:12 +0000 [r202765] Matthew Fredrickson <creslin@digium.com>
+
+ * 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 16:33 +0000 [r202673] David Vossel <dvossel@digium.com>
+
+ * /, 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 20:19 +0000 [r202495-202511] Russell Bryant <russell@digium.com>
+
+ * 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 ........ ................
+
+ * /, 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 16:31 +0000 [r202473] Sean Bright <sean@malleable.com>
+
+ * 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 15:37 +0000 [r202411] David Vossel <dvossel@digium.com>
+
+ * 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 15:17 +0000 [r202340-202346] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 16:16 +0000 [r202261-202265] Russell Bryant <russell@digium.com>
+
+ * 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 [r202186] Sean Bright <sean@malleable.com>
+
+ * /, 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 [r202007] Matthew Nicholson <mnicholson@digium.com>
+
+ * 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 [r201995] David Vossel <dvossel@digium.com>
+
+ * 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 15:49 +0000 [r201785-201906] Tilghman Lesher <tlesher@digium.com>
+
+ * 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: Merged revisions 201829 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009)
+ | 13 lines Merged revisions 201828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009)
+ | 6 lines 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 16:44 +0000 [r201679] David Vossel <dvossel@digium.com>
+
+ * 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 15:40 +0000 [r201614] Russell Bryant <russell@digium.com>
+
+ * 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 15:23 +0000 [r201595] David Vossel <dvossel@digium.com>
+
+ * /, 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 21:33 +0000 [r201533] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 20:12 +0000 [r201461-201465] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 [r201447-201454] David Vossel <dvossel@digium.com>
+
+ * 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 19:49 +0000 [r201446] David Brooks <dbrooks@digium.com>
+
+ * /, 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 15:25 +0000 [r201360] David Vossel <dvossel@digium.com>
+
+ * /, 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 12:06 +0000 [r201265] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, 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 22:30 +0000 [r201224] David Vossel <dvossel@digium.com>
+
+ * /, 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 19:47 +0000 [r200990-201097] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 16:28 +0000 [r200984] David Vossel <dvossel@digium.com>
+
+ * /, 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 16:05 +0000 [r200948] Michiel van Baak <michiel@vanbaak.info>
+
+ * 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 12:55 +0000 [r200842] Eliel C. Sardanons <eliels@gmail.com>
+
+ * res/res_smdi.c, /: Merged revisions 200841 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200841 |
+ eliel | 2009-06-16 08:32:00 -0400 (Tue, 16 Jun 2009) | 6 lines
+ Show the interface name on error, if it is not found. If the
+ smdiport specified is not found, show the interface name instead
+ of '(null)'. ........
+
+2009-06-16 02:41 +0000 [r200807] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 200799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200799 |
+ moy | 2009-06-15 21:24:30 -0500 (Mon, 15 Jun 2009) | 2 lines keep
+ backwards compatible chan_dahdi with older openr2 versions by not
+ using the new skip category feature unless supported ........
+
+2009-06-16 01:30 +0000 [r200690-200765] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, 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: Merged revisions 200689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r200689 |
+ kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 11
+ 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 [r200517] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 ........ ................
+
+2009-06-14 06:33 +0000 [r200512] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
+ build_tools/menuselect-deps.in: Merged revisions 200477 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r200477 | moy | 2009-06-14 01:13:48 -0500 (Sun, 14 Jun
+ 2009) | 3 lines added openr2 to menuselect-deps.in, recent commit
+ in menuselect made me realize this was never done but was working
+ anyways also added support for skip category request feature of
+ openr2 and updated chan_dahdi.conf.sample ........
+
+2009-06-12 19:08 +0000 [r200364] Mark Michelson <mmichelson@digium.com>
+
+ * 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 ........ ................
+
+2009-06-12 02:20 +0000 [r200198-200255] Sean Bright <sean@malleable.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 200254 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r200254 | seanbright | 2009-06-11 22:20:19 -0400 (Thu,
+ 11 Jun 2009) | 5 lines Call chgrp instead of chown when setting
+ run directory group ownership. (issue #13153) Reported by:
+ pabelanger ........
+
+ * 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#. ........
+
+ * /: Undo block of revision 199782 (will be merging it momentarily)
+
+2009-06-11 21:35 +0000 [r200172] Terry Wilson <twilson@digium.com>
+
+ * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null
+
+2009-06-11 21:18 +0000 [r200154] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-beta3
+
+2009-06-11 12:19 +0000 [r200051] Leif Madsen <lmadsen@digium.com>
+
+ * 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 20:37 +0000 [r199998] David Brooks <dbrooks@digium.com>
+
+ * 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 [r199966] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 [r199860] Sean Bright <sean.bright@gmail.com>
+
+ * 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 20:48 +0000 [r199744-199819] David Vossel <dvossel@digium.com>
+
+ * /, 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,
+ res/res_timing_timerfd.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 [r199634] Sean Bright <sean.bright@gmail.com>
+
+ * 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 17:42 +0000 [r199591] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 ........
+
+2009-06-06 21:39 +0000 [r199369] Russell Bryant <russell@digium.com>
+
+ * Makefile, /: Merged revisions 199368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r199368 |
+ russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines
+ Switch from "echo -n" to printf. On my mac, the -n was just
+ getting printed out. ........
+
+2009-06-05 21:25 +0000 [r199299] David Vossel <dvossel@digium.com>
+
+ * 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 13:52 +0000 [r199230] Mark Michelson <mmichelson@digium.com>
+
+ * 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 19:15 +0000 [r199140] David Vossel <dvossel@digium.com>
+
+ * 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 [r199054] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk/_private.h, main/asterisk.c, main/loader.c, /:
+ 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 15:24 +0000 [r198827-198886] David Vossel <dvossel@digium.com>
+
+ * 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 13:51 +0000 [r198794] Joshua Colp <jcolp@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: 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 21:04 +0000 [r198730] Russell Bryant <russell@digium.com>
+
+ * channels/iax2-parser.c, /: Merged revisions 198729 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009)
+ | 2 lines Tell the IAX2 parser about more control frame types.
+ ........
+
+2009-06-01 18:44 +0000 [r198629] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 17:53 +0000 [r198471] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_strings.c: Merged revisions 198470 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009)
+ | 2 lines Fix documentation for FIELDQTY. ........
+
+2009-05-31 01:48 +0000 [r198440] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, 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) ........
+
+2009-05-31 01:21 +0000 [r198436] Russell Bryant <russell@digium.com>
+
+ * 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 20:22 +0000 [r198297-198397] Sean Bright <sean.bright@gmail.com>
+
+ * 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) ........
+ ................
+
+ * 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 02:35 +0000 [r198250] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 [r198148-198188] Russell Bryant <russell@digium.com>
+
+ * /, 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 19:26 +0000 [r198111] Eliel C. Sardanons <eliels@gmail.com>
+
+ * CREDITS, /: Merged revisions 198083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r198083 |
+ eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines
+ Apply anti-spam obfuscation to an email address. ........
+
+2009-05-29 19:14 +0000 [r198075] Matthew Nicholson <mnicholson@digium.com>
+
+ * 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 18:40 +0000 [r198066] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 18:18 +0000 [r198008] Sean Bright <sean.bright@gmail.com>
+
+ * 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 16:29 +0000 [r197994] Russell Bryant <russell@digium.com>
+
+ * /, 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 23:54 +0000 [r197894] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_mixmonitor.c, /: Merged revisions 197828 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009)
+ | 8 lines Update documentation in MixMonitor. Updated the
+ MixMonitor documentation for the 'b' option so that it is more
+ obvious that you must not optimize away the Local channel when
+ using this option. (closes issue #14829) Reported by: licedey
+ Tested by: mmichelson, licedey, lmadsen ........
+
+2009-05-28 18:50 +0000 [r197703] Joshua Colp <jcolp@digium.com>
+
+ * 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 16:15 +0000 [r197625] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, 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 15:44 +0000 [r197548-197619] Mark Michelson <mmichelson@digium.com>
+
+ * 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 ........ ................
+
+ * main/audiohook.c, apps/app_chanspy.c, /,
+ include/asterisk/audiohook.h: 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 14:56 +0000 [r197471-197542] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 [r197441] Gavin Henry <ghenry@suretecsystems.com>
+
+ * 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 23:49 +0000 [r197375] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/xml.c: Merged revisions 197374 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r197374 |
+ tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines
+ Revert commit 192032. This define is needed on Mac OS X. ........
+
+2009-05-27 22:23 +0000 [r197336] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/agi.h, /: Merged revisions 197335 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May
+ 2009) | 3 lines Ensure that this header includes xmldoc.h, since
+ it depends on it. ........
+
+2009-05-27 20:11 +0000 [r197263] Sean Bright <sean.bright@gmail.com>
+
+ * 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 19:30 +0000 [r197247] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 17:28 +0000 [r197176] Jeff Peeler <jpeeler@digium.com>
+
+ * 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 [r196950-197092] Sean Bright <sean.bright@gmail.com>
+
+ * 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 ........
+
+ * /, 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)
+ ........
+
+ * main/xmldoc.c, /: Merged revisions 196948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196948 |
+ seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8
+ lines Reset the terminal to the correct fg/bg after XML
+ documenation is rendered. (closes issue #15200) Reported by:
+ ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright
+ (license 71) Tested by: ajohnson ........
+
+ * main/manager.c, /: Merged revisions 196945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196945 |
+ seanbright | 2009-05-26 18:38:05 -0400 (Tue, 26 May 2009) | 13
+ lines Add ActionID to CoreShowChannel event. There is
+ inconsistency in how we handle manager responses that are lists
+ of items and, unfortunately, third parties have come to rely on
+ ActionID being on every event within those lists instead of just
+ keeping track of the ActionID for the current response. This
+ change makes CoreShowChannels include the ActionID with each
+ CoreShowChannel event generated as a result of it being called.
+ (closes issue #15001) Reported by: sum Patches:
+ patchactionid2.patch uploaded by sum (license 766) ........
+
+2009-05-26 22:44 +0000 [r196870-196949] Russell Bryant <russell@digium.com>
+
+ * /, 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 16:39 +0000 [r196793] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_queue.c, /: Merged revisions 196792 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196792 |
+ seanbright | 2009-05-26 12:38:54 -0400 (Tue, 26 May 2009) | 2
+ lines Add a missing unref for queues in handle_statechange.
+ ........
+
+2009-05-26 13:47 +0000 [r196661-196724] Joshua Colp <jcolp@digium.com>
+
+ * /, 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)
+ ........ ................
+
+2009-05-23 05:29 +0000 [r196487] Moises Silva <moises.silva@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 196456 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r196456 | moy | 2009-05-22 23:27:47 -0500 (Fri, 22 May 2009) | 1
+ line set MFCR2_CATEGORY just when starting the pbx ........
+
+2009-05-22 21:59 +0000 [r196452] David Vossel <dvossel@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: 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 19:48 +0000 [r196378] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, apps/app_minivm.c: Merged revisions 196377 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r196377 |
+ eliel | 2009-05-22 15:38:33 -0400 (Fri, 22 May 2009) | 11 lines
+ Unregister every registered application by MiniVM. The MinivmMWI
+ application was not being unregistered on unload and we were not
+ able to load again the module or reload it. (closes issue #15174)
+ Reported by: junky Patches: unregister_minivm_mwi.diff uploaded
+ by junky (license 177) ........
+
+2009-05-22 13:59 +0000 [r196120] Joshua Colp <jcolp@digium.com>
+
+ * 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 19:15 +0000 [r196000] David Vossel <dvossel@digium.com>
+
+ * 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 15:57 +0000 [r195883] Matthew Nicholson <mnicholson@digium.com>
+
+ * 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 [r195842] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 17:35 +0000 [r195639-195707] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 20:19 +0000 [r195531] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 14:49 +0000 [r195452] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 21:25 +0000 [r195405] Eliel C. Sardanons <eliels@gmail.com>
+
+ * 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 20:55 +0000 [r195359-195373] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c, include/asterisk/smdi.h, res/res_monitor.c,
+ apps/app_voicemail.c, res/res_smdi.c, /,
+ include/asterisk/monitor.h: Merged 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 19:01 +0000 [r195319] Mark Michelson <mmichelson@digium.com>
+
+ * 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 15:57 +0000 [r195212] Joshua Colp <jcolp@digium.com>
+
+ * 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 14:54 +0000 [r195164] Eliel C. Sardanons <eliels@gmail.com>
+
+ * 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 14:00 +0000 [r195099] Joshua Colp <jcolp@digium.com>
+
+ * 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 ........ ................
+
+2009-05-18 13:50 +0000 [r195093-195094] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, main/xml.c: Merged revisions 195075 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r195075 |
+ eliel | 2009-05-18 09:30:34 -0400 (Mon, 18 May 2009) | 3 lines Do
+ not avoid loading the XML documentation if not XInclude
+ substitution is done. ........
+
+ * doc/appdocsxml.dtd, Makefile, /, main/xml.c: Merged revisions
+ 194982 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194982 |
+ eliel | 2009-05-16 16:01:22 -0400 (Sat, 16 May 2009) | 20 lines
+ Allow to include sections of other parts of the xml
+ documentation. Avoid duplicating xml documentation by allowing to
+ include other parts of the xml documentation using XInclude.
+ Example: <xi:include
+ xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
+ (Insert this line to include the synopsis of the CHANNEL function
+ xml documentation). It is also possible to include documentation
+ from other files in the 'documentation/' directory using the
+ href="" attribute inside a xinclude element. (closes issue
+ #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling
+ ........
+
+2009-05-18 13:39 +0000 [r195092] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 13:14 +0000 [r195024] Russell Bryant <russell@digium.com>
+
+ * 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 ........ ................
+
+2009-05-16 18:43 +0000 [r194946] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/pbx.c, /: Merged revisions 194945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194945 |
+ eliel | 2009-05-16 14:32:11 -0400 (Sat, 16 May 2009) | 8 lines
+ Fix a missing unlock in case of error, and a missing free().
+ Always free the allocated memory for a string field, because we
+ are always using it (not only when xmldocs are enabled). Also if
+ there is an error allocating memory for the string field remember
+ to unlock the list of registered applications, before returning.
+ ........
+
+2009-05-15 22:48 +0000 [r194836-194877] David Vossel <dvossel@digium.com>
+
+ * 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 [r194717-194768] Russell Bryant <russell@digium.com>
+
+ * 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. ........
+
+2009-05-15 13:47 +0000 [r194650] Michiel van Baak <michiel@vanbaak.info>
+
+ * CREDITS, /: Merged revisions 194649 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194649 |
+ mvanbaak | 2009-05-15 15:43:24 +0200 (Fri, 15 May 2009) | 2 lines
+ add eliel ........
+
+2009-05-15 13:42 +0000 [r194648] Eliel C. Sardanons <eliels@gmail.com>
+
+ * doc/appdocsxml.dtd, main/xmldoc.c, /: Merged revisions 194635 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r194635 | eliel | 2009-05-15 09:23:37 -0400 (Fri, 15 May
+ 2009) | 16 lines Allow to specify an enumlist inside an enum. It
+ was not possible to use an enumlist inside an enum: <enumlist>
+ <enum name="aa"> <enumlist> ... </enumlist> </enum> </enumlist>
+ Now we will be able to insert as many levels as we want. (closes
+ issue #15112) Reported by: lmadsen ........
+
+2009-05-14 22:31 +0000 [r194545] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /: 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 [r194510] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 [r194437] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 ........
+
+2009-05-14 16:23 +0000 [r194431] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 194430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r194430 |
+ tilghman | 2009-05-14 11:22:14 -0500 (Thu, 14 May 2009) | 7 lines
+ If the timing ended on a zero, then we would loop forever.
+ (closes issue #14983) Reported by: teox Patches:
+ 20090513__issue14983.diff.txt uploaded by tilghman (license 14)
+ Tested by: teox ........
+
+2009-05-13 13:42 +0000 [r194213] Joshua Colp <jcolp@digium.com>
+
+ * 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 [r194141] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 22:48 +0000 [r194059] Matthew Nicholson <mnicholson@digium.com>
+
+ * 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 [r193962] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 [r193823-193959] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 22:12 +0000 [r193719] Russell Bryant <russell@digium.com>
+
+ * /, res/res_timing_timerfd.c: Merged revisions 193718 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r193718 | russell | 2009-05-11 17:04:40 -0500 (Mon, 11 May 2009)
+ | 12 lines Fix some timer state corruption. In res_timer_timerfd,
+ handle the case that set_rate gets called while a timer is still
+ in continuous mode. In this case, we want to remember the
+ configured rate, but not actually set it until continuous mode
+ has been disabled. Thanks to dvossel for finding and helping to
+ debug the problem. (closes issue #15080) Reported by: dvossel
+ Tested by: dvossel ........
+
+2009-05-11 19:17 +0000 [r193617] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 18:59 +0000 [r193612] Leif Madsen <lmadsen@digium.com>
+
+ * /, funcs/func_channel.c: Update CHANNEL(transfercapabilities)
+ documentation. (closes issue #15073) Reported by: pkempgen
+ Patches: 20090511__issue15073__trunk.diff.txt uploaded by
+ tilghman (license 14)
+
+2009-05-10 17:08 +0000 [r193503] Joshua Colp <jcolp@digium.com>
+
+ * main/bridging.c, /: Merged revisions 193502 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193502 |
+ file | 2009-05-10 14:07:46 -0300 (Sun, 10 May 2009) | 2 lines Fix
+ a bug where receiving a control frame of subclass -1 would cause
+ certain channels to get hung up. ........
+
+2009-05-09 11:33 +0000 [r193462] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/event.h, /: Merged revisions 193461 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r193461 | russell | 2009-05-09 06:33:09 -0500 (Sat, 09 May 2009)
+ | 2 lines Minor documentation update for ast_event_queue().
+ ........
+
+2009-05-08 20:52 +0000 [r193390] David Vossel <dvossel@digium.com>
+
+ * /, 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 19:51 +0000 [r193350] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 193349 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r193349 |
+ mmichelson | 2009-05-08 14:50:44 -0500 (Fri, 08 May 2009) | 12
+ lines Reset the members' call counts when resetting queue
+ statistics. This helps to prevent odd scenarios where a queue
+ will claim to have taken 0 calls, but the members appear to have
+ taken a non-zero amount. (closes issue #15068) Reported by: sum
+ Patches: patchreset.patch uploaded by sum (license 766) Tested
+ by: sum ........
+
+2009-05-08 15:36 +0000 [r193336] Sean Bright <sean.bright@gmail.com>
+
+ * 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 14:55 +0000 [r193266] David Vossel <dvossel@digium.com>
+
+ * 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 14:12 +0000 [r193197] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 [r193123] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 22:51 +0000 [r193080] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 17:53 +0000 [r192936-193008] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 ........ ................
+
+2009-05-07 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-beta2
+
+2009-05-06 22:20 +0000 [r192874] Jeff Peeler <jpeeler@digium.com>
+
+ * /, 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 17:57 +0000 [r192813] Matthew Fredrickson <creslin@digium.com>
+
+ * 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 17:41 +0000 [r192637-192810] Joshua Colp <jcolp@digium.com>
+
+ * 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 ........
+
+ * res/res_clialiases.c, /: Merged revisions 192736 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192736 | file | 2009-05-06 13:09:27 -0300 (Wed, 06 May 2009) | 4
+ lines Make the code that prevents an infinite loop from happening
+ into a case insensitive check. (thanks eliel) ........
+
+ * res/res_clialiases.c, /: Merged revisions 192700 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r192700 | file | 2009-05-06 11:35:47 -0300 (Wed, 06 May 2009) | 5
+ lines Fix an infinite loop with tab completion of CLI aliases
+ that reference themselves. (closes issue #15020) Reported by:
+ junky ........
+
+ * /, 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 [r192528] Sean Bright <sean.bright@gmail.com>
+
+ * /, 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 18:27 +0000 [r192402-192480] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 13:43 +0000 [r192298-192360] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 [r192217] David Vossel <dvossel@digium.com>
+
+ * 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 19:34 +0000 [r192175] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 19:31 +0000 [r192135-192173] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, res/res_agi.c: Merged 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 ........
+
+ * 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 17:45 +0000 [r192097] Leif Madsen <lmadsen@digium.com>
+
+ * apps/app_forkcdr.c, /: Merged revisions 192096 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192096 |
+ lmadsen | 2009-05-04 13:42:56 -0400 (Mon, 04 May 2009) | 4 lines
+ Commit documentation changes related to issue #14801. (issue
+ #14801) ........
+
+2009-05-04 15:54 +0000 [r192033] Eliel C. Sardanons <eliels@gmail.com>
+
+ * /, main/xml.c: Merged revisions 192032 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r192032 |
+ eliel | 2009-05-04 11:35:35 -0400 (Mon, 04 May 2009) | 3 lines Do
+ not re-define _POSIX_C_SOURCE if it was already defined. ........
+
+2009-05-04 10:01 +0000 [r191958] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, 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. ........
+
+2009-05-03 14:06 +0000 [r191885] Russell Bryant <russell@digium.com>
+
+ * Makefile, /: Merged revisions 191884 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r191884 |
+ russell | 2009-05-03 09:05:10 -0500 (Sun, 03 May 2009) | 2 lines
+ Remove unnecessary compiler flag ........
+
+2009-05-02 18:48 +0000 [r191779] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, 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 [r191703] Sean Bright <sean.bright@gmail.com>
+
+ * 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 [r191554-191563] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 [r191508] Jeff Peeler <jpeeler@digium.com>
+
+ * 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. ........ ................
+
+2009-04-30 17:46 +0000 [r191224-191370] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 [r191139] David Brooks <dbrooks@digium.com>
+
+ * 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 08:59 +0000 [r190994] Russell Bryant <russell@digium.com>
+
+ * main/indications.c, /: Merged revisions 190993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190993 |
+ russell | 2009-04-29 03:58:39 -0500 (Wed, 29 Apr 2009) | 7 lines
+ Log an error message if indications.conf is not found. (closes
+ issue #14990) Reported by: tzafrir Patches: indications_err.diff
+ uploaded by tzafrir (license 46) ........
+
+2009-04-29 06:38 +0000 [r190985] TransNexus OSP Development <support@transnexus.com>
+
+ * 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 [r190907] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 14:17 +0000 [r190732-190869] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, /: Merged revisions 190865 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190865 |
+ kpfleming | 2009-04-28 09:15:47 -0500 (Tue, 28 Apr 2009) | 5
+ lines Build XML documention from *only* the source files that
+ have docs in them Change the build process so that
+ doc/core-en_US.xml is dependent solely on the source files that
+ have documentation in them, not on all source files. ........
+
+ * /, 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 [r190729] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 ........
+
+2009-04-27 19:15 +0000 [r190666] Russell Bryant <russell@digium.com>
+
+ * res/res_smdi.c, /: Merged revisions 190663 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009)
+ | 22 lines Merged revisions 190661-190662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009)
+ | 9 lines Resolve a crash in res_smdi when used with chan_dahdi.
+ When chan_dahdi goes to get an SMDI message, it provides no
+ search criteria. It just grabs the next message that arrives.
+ This code was written with the SMDI dialplan functions in mind,
+ since that is now the preferred method of using SMDI. However,
+ this broke support of it being used from chan_dahdi. (closes
+ AST-212) ........ r190662 | russell | 2009-04-27 14:03:59 -0500
+ (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661. ........
+ ................
+
+2009-04-27 16:28 +0000 [r190625] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 190622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190622 |
+ mmichelson | 2009-04-27 11:26:14 -0500 (Mon, 27 Apr 2009) | 3
+ lines Update warning message to not have pipes and contain all
+ options. ........
+
+2009-04-23 21:23 +0000 [r190383] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 190371 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ ........
+
+2009-04-23 20:44 +0000 [r190355] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: Merged revisions 190352 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190352 |
+ tilghman | 2009-04-23 15:42:11 -0500 (Thu, 23 Apr 2009) | 7 lines
+ Labels are sometimes (most of the time?) NULL for extensions.
+ (closes issue #14895) Reported by: chris-mac Patches:
+ 20090423__bug14895__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: lmadsen ........
+
+2009-04-23 19:18 +0000 [r190297] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 190287 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r190287 | file | 2009-04-23 16:15:30 -0300 (Thu,
+ 23 Apr 2009) | 13 lines Merged revisions 190286 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6
+ lines Fix a bug in chan_local glare hangup detection. If both
+ sides of a Local channel were hung up at around the same time it
+ was possible for one thread to destroy the local private
+ structure and have the other thread immediately try to remove the
+ already freed structure from the local channel list. ........
+ ................
+
+2009-04-23 17:47 +0000 [r190253] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 190250 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r190250 |
+ mmichelson | 2009-04-23 12:45:35 -0500 (Thu, 23 Apr 2009) | 9
+ lines Fix reversed behavior of leavewhenempty option in
+ queues.conf. (closes issue #14650) Reported by: alecdavis
+ Patches: 14650.patch uploaded by mmichelson (license 60) Tested
+ by: mmichelson, lmadsen ........
+
+2009-04-22 21:43 +0000 [r190096] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/lock.h: Merged revisions 190093 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r190093 | tilghman | 2009-04-22 16:38:15 -0500
+ (Wed, 22 Apr 2009) | 14 lines Merged revisions 190092 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009)
+ | 7 lines Detect availability of pthread_rwlock_timedwrlock()
+ before using it. (closes issue #14930) Reported by: tilghman
+ Patches: 20090420__bug14930.diff.txt uploaded by tilghman
+ (license 14) Tested by: mvanbaak, tilghman ........
+ ................
+
+2009-04-22 21:18 +0000 [r189997-190066] Jeff Peeler <jpeeler@digium.com>
+
+ * main/cli.c, funcs/func_groupcount.c, /, main/app.c,
+ include/asterisk/channel.h: Merged revisions 190057 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009)
+ | 9 lines Fix building of chan_h323 with gcc-3.3 There seems to
+ be a bug with old versions of g++ that doesn't allow a structure
+ member to use the name list. Rename list member to group_list in
+ ast_group_info and change the few places it is used. (closes
+ issue #14790) Reported by: stuarth ........
+
+ * channels/h323/chan_h323.h, /, channels/h323/ast_h323.cxx,
+ channels/chan_h323.c: Merged revisions 189993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189993 |
+ jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines
+ Make chan_h323 respect packetization settings and fix small
+ reload issue. Previously, packetization settings were ignored and
+ now they are not. A new config option 'autoframing' has been
+ added to mirror the way chan_sip handles it. Turning on the
+ autoframing option (available both as a global option or per
+ peer) overrides the local settings with the remote packetization
+ settings. Testing was performed with varying packetization levels
+ with the following codecs: ulaw, alaw, gsm, and g729. Also, an
+ unrelated config reload issue has been fixed in the case of the
+ config file not changing. (closes issue #12415) Reported by: pj
+ Patches: 2009012200_h323packetization.diff.txt uploaded by
+ mvanbaak (license 7), modified by me ........
+
+2009-04-22 18:01 +0000 [r189986] Russell Bryant <russell@digium.com>
+
+ * /, main/features.c: Merged revisions 189951 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189951 |
+ russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines
+ Fix call parking callback. Pipes -> Commas. ........
+
+2009-04-22 16:04 +0000 [r189816-189914] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_unistim.c, /: Merged revisions 189911 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009)
+ | 7 lines Do not continue to receive DTMF, when the channel is
+ hungup and about to be destroyed. (closes issue #14858) Reported
+ by: barryf Patches: 20090421__bug14858.diff.txt uploaded by
+ tilghman (license 14) Tested by: barryf ........
+
+ * /, configure, configure.ac: Merged revisions 189813 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r189813 | tilghman | 2009-04-22 01:33:08 -0500 (Wed, 22 Apr 2009)
+ | 3 lines Detect liblua on SuSE, and add libm for linking for
+ Fedora. (Reported via the -dev list, Subject: Compiling Asterisk
+ with LUA) ........
+
+2009-04-21 20:45 +0000 [r189775] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 189771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r189771 |
+ dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines
+ Fixes segfault when switching UDP to TCP in sip.conf after
+ reload. If transport in sip.conf is switched from UDP to TCP,
+ Asterisk segfaults right after issuing a sip reload. The problem
+ is the socket type is changed to TCP but the fd may still be
+ present for UDP. Later, when the TCP session should be created or
+ set using an existing one, it isn't because the old file
+ descriptor is still present. Now every time transport is changed
+ during a sip.conf reload, the file descriptor is set to -1,
+ signifying it must be created or found. (closes issue #14727)
+ Reported by: pj Tested by: dvossel Review:
+ http://reviewboard.digium.com/r/229/ ........
+
+2009-04-20 22:11 +0000 [r189540] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c, /: Merged revisions 189539 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r189539 | tilghman | 2009-04-20 17:10:25 -0500 (Mon, 20 Apr 2009)
+ | 3 lines Use nanosleep instead of poll. This is not just because
+ mmichelson suggested it, but also because Mac OS X puked on my
+ poll(). ........
+
+2009-04-20 21:41 +0000 [r189536] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 189495,189516 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r189495 | twilson | 2009-04-20 16:24:34 -0500
+ (Mon, 20 Apr 2009) | 9 lines Merged revisions 189463 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20
+ Apr 2009) | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL
+ ........ ................ r189516 | twilson | 2009-04-20 16:29:29
+ -0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189465 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009)
+ | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is
+ set ........ ................
+
+2009-04-20 21:36 +0000 [r189533] Sean Bright <sean.bright@gmail.com>
+
+ * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189464 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r189464 | seanbright | 2009-04-20 17:09:59 -0400
+ (Mon, 20 Apr 2009) | 20 lines Merged revisions 189462 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr
+ 2009) | 13 lines Properly handle @s within hints in AEL. AEL was
+ not handling the case of a device hint containing an @ symbol,
+ which caused parking hints (e.g. hint(park:exten@context)) to
+ error out the parser. This patch makes AEL treat the @ the same
+ way it treats colon and ampersand now, meaning the characters are
+ included in verbatim. (closes issue #14941) Reported by: bpgoldsb
+ Patches: bug14941.patch uploaded by seanbright (license 71)
+ Tested by: bpgoldsb ........ ................
+
+2009-04-20 17:11 +0000 [r189353] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 14:07 +0000 [r189281] Mark Michelson <mmichelson@digium.com>
+
+ * 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 01:42 +0000 [r189207-189208] David Vossel <dvossel@digium.com>
+
+ * 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/ ........ ................
+
+ * /, 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 21:56 +0000 [r189140] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 [r189105] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 19:47 +0000 [r189081] Sean Bright <sean.bright@gmail.com>
+
+ * 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 17:31 +0000 [r189068] Matthew Nicholson <mnicholson@digium.com>
+
+ * 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 14:50 +0000 [r188941-188950] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 [r188777-188839] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 20:20 +0000 [r188474-188598] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 17:46 +0000 [r188259-188416] Joshua Colp <jcolp@digium.com>
+
+ * 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 05:47 +0000 [r188209-188213] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 [r188105] Mark Michelson <mmichelson@digium.com>
+
+ * 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 16:34 +0000 [r188070] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 [r188039] Mark Michelson <mmichelson@digium.com>
+
+ * 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 [r187916] Jeff Peeler <jpeeler@digium.com>
+
+ * 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 17:31 +0000 [r187769] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 [r187724] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, 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 16:05 +0000 [r187679] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 16:01 +0000 [r187677] Russell Bryant <russell@digium.com>
+
+ * tests/test_sched.c, 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 03:57 +0000 [r187601] Tilghman Lesher <tlesher@digium.com>
+
+ * main/audiohook.c, main/bridging.c, 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: 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 21:09 +0000 [r187564] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merge revision 187488 from trunk.
+
+2009-04-09 19:29 +0000 [r187531-187546] David Vossel <dvossel@digium.com>
+
+ * 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) | 6 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) ........
+
+ * /, 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/ ........
+ ................
+
+2009-04-09 19:15 +0000 [r187496] Mark Michelson <mmichelson@digium.com>
+
+ * 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 18:55 +0000 [r187051-187487] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 ........ ................
+
+ * /, 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. ........ ................
+
+ * /, funcs/func_odbc.c: Merged revisions 187050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r187050 |
+ tilghman | 2009-04-08 12:08:43 -0500 (Wed, 08 Apr 2009) | 7 lines
+ If the first column is empty, output a delimiter anyway. (closes
+ issue #14848) Reported by: john8675309 Patches:
+ 20090408__bug14848.diff.txt uploaded by tilghman (license 14)
+ Tested by: john8675309 ........
+
+2009-04-08 16:54 +0000 [r186988-187049] Mark Michelson <mmichelson@digium.com>
+
+ * 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 12:39 +0000 [r186929] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 186928 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186928 |
+ russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines
+ Update some comments and resolve potential memory corruption in
+ chan_sip. While browsing chan_sip the other day, I noticed this
+ dangerous code in dialog_needdestroy(). This function is an
+ ao2_callback. It is absolutely _not_ okay to unlock the container
+ from within this function. It's also not clear why it was useful.
+ Given that it could cause memory corruption, I have removed it.
+ There was also a TODO comment left describing a potential
+ implementation of an improvement to the needdestroy handling. I'm
+ not convinced that what was described is the best choice here, so
+ I have briefly described the way that this function is used today
+ that could be improved. ........
+
+2009-04-08 05:08 +0000 [r186901] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 [r186836-186845] Mark Michelson <mmichelson@digium.com>
+
+ * formats/format_wav_gsm.c, /, formats/format_wav.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 [r186807] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 20:59 +0000 [r186723] Mark Michelson <mmichelson@digium.com>
+
+ * 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 [r186469] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 20:05 +0000 [r186449] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 15:56 +0000 [r186324] Joshua Colp <jcolp@digium.com>
+
+ * 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 15:19 +0000 [r186302] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c, /: Merged revisions 186297 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r186297 | tilghman | 2009-04-03 10:18:28 -0500 (Fri, 03 Apr 2009)
+ | 4 lines Compatibility fix for glibc 2.4 (Closes issue #14820)
+ Reported by: phsultan ........
+
+2009-04-03 14:34 +0000 [r186289] Mark Michelson <mmichelson@digium.com>
+
+ * 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 02:11 +0000 [r186233] Russell Bryant <russell@digium.com>
+
+ * 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 22:00 +0000 [r186178] Mark Michelson <mmichelson@digium.com>
+
+ * 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 17:28 +0000 [r186111] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 [r186022-186063] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: 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 ........
+ ................ ................
+
+ * main/strings.c, /: Merged revisions 186021 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r186021 |
+ tilghman | 2009-04-02 10:14:22 -0500 (Thu, 02 Apr 2009) | 7 lines
+ Missed a common case for needing to extend the buffer. (closes
+ issue #14716) Reported by: sum Patches:
+ 20090402__bug14716.diff.txt uploaded by tilghman (license 14)
+ Tested by: sum ........
+
+2009-04-02 13:54 +0000 [r185957] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 22:44 +0000 [r185947] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/pbx.h, include/asterisk/strings.h,
+ main/taskprocessor.c, res/res_odbc.c,
+ include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c,
+ main/manager.c, /, main/tdd.c, include/asterisk/astobj2.h,
+ main/ast_expr2f.c: Merged revisions 185912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185912 |
+ tilghman | 2009-04-01 15:13:28 -0500 (Wed, 01 Apr 2009) | 4 lines
+ Merge changes from str_substitution that are unrelated to that
+ branch. Included is a small bugfix to an ast_str helper, but most
+ of these changes are simply doxygen fixes. ........
+
+2009-04-01 13:51 +0000 [r185775] Russell Bryant <russell@digium.com>
+
+ * 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 [r185667] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 22:13 +0000 [r185472-185605] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 185604 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r185604 |
+ mmichelson | 2009-03-31 17:12:52 -0500 (Tue, 31 Mar 2009) | 3
+ lines Fix trunk's compilation. ........
+
+ * 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 17:51 +0000 [r185428] David Brooks <dbrooks@digium.com>
+
+ * 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:
+ <iq from='dbrooksjab@235-22-24-10/Telepathy'
+ to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
+ <session xmlns='http://www.google.com/session'
+ initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
+ id='1837267342'> <description
+ xmlns='http://www.google.com/session/phone'> <payload-type
+ clockrate='16000' name='speex' id='96'/> <payload-type
+ clockrate='8000' name='PCMA' id='8'/> <payload-type
+ clockrate='8000' name='PCMU' id='0'/> <payload-type
+ clockrate='90000' name='MPA' id='97'/> <payload-type
+ clockrate='16000' name='SIREN' id='98'/> <payload-type
+ clockrate='8000' name='telephone-event' id='99'/> </description>
+ </session> </iq> Review: http://reviewboard.digium.com/r/181/
+ ........ ................
+
+2009-03-31 14:59 +0000 [r185264] Russell Bryant <russell@digium.com>
+
+ * 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 14:11 +0000 [r185200] Joshua Colp <jcolp@digium.com>
+
+ * 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 20:52 +0000 [r185128-185129] Richard Mudgett <rmudgett@digium.com>
+
+ * 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 16:52 +0000 [r185089] Mark Michelson <mmichelson@digium.com>
+
+ * 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 14:43 +0000 [r184951] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 [r184913] Russell Bryant <russell@digium.com>
+
+ * 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 05:56 +0000 [r184839-184846] Russell Bryant <russell@digium.com>
+
+ * 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 19:21 +0000 [r184779] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c, main/timing.c, main/channel.c, /,
+ bridges/bridge_softmix.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 18:12 +0000 [r184707-184729] Russell Bryant <russell@digium.com>
+
+ * 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. ........
+
+ * apps/app_queue.c, apps/app_voicemail.c, main/cli.c,
+ include/asterisk/app.h, /, apps/app_dumpchan.c, main/app.c:
+ Merged revisions 184693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184693 |
+ russell | 2009-03-27 11:21:10 -0500 (Fri, 27 Mar 2009) | 2 lines
+ Change global_app_buf to ast_str_thread_global_buf. ........
+
+2009-03-27 15:58 +0000 [r184650-184678] Joshua Colp <jcolp@digium.com>
+
+ * /, bridges/bridge_softmix.c: Merged revisions 184677 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184677 | file | 2009-03-27 12:57:28 -0300 (Fri, 27 Mar 2009) | 7
+ lines Fix a potential timer leak in bridge_softmix. It is
+ possible for a bridge to be created without actually being used.
+ In that scenario a timing file descriptor would be opened and not
+ closed. To fix this the timing file descriptor is now closed in
+ the destroy callback, not the thread function. ........
+
+ * /, 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) ........
+
+ * /, bridges/bridge_softmix.c: Merged revisions 184639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184639 | file | 2009-03-27 11:18:40 -0300 (Fri, 27 Mar 2009) | 2
+ lines Remove a cast that is not needed. ........
+
+2009-03-27 14:09 +0000 [r184632] Russell Bryant <russell@digium.com>
+
+ * 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 13:59 +0000 [r184612-184629] Joshua Colp <jcolp@digium.com>
+
+ * /, bridges/bridge_softmix.c: Merged revisions 184628 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r184628 | file | 2009-03-27 10:57:29 -0300 (Fri, 27 Mar 2009) | 6
+ lines Fix a potential race condition when creating a software
+ based mixing bridge. It was possible for no timer to become
+ available between creating the bridge and starting it. We now
+ open a timer when creating it and keep it open until the bridge
+ is destroyed. ........
+
+ * /, 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 02:35 +0000 [r184514-184552] Russell Bryant <russell@digium.com>
+
+ * /, 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 [r184454] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 ........
+ ................
+
+2009-03-25 22:15 +0000 [r184343-184346] Russell Bryant <russell@digium.com>
+
+ * /, 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. ........
+
+ * include/asterisk/_private.h, channels/chan_iax2.c,
+ channels/chan_dahdi.c, include/asterisk/event.h,
+ apps/app_minivm.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: 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/ ........
+
+2009-03-25 19:27 +0000 [r184266-184283] Joshua Colp <jcolp@digium.com>
+
+ * /, 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.
+ ........
+
+ * main/bridging.c, /: Merged revisions 183652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r183652 |
+ file | 2009-03-22 18:00:28 -0300 (Sun, 22 Mar 2009) | 4 lines Fix
+ a minor logic flaw with the bridge generic thread. We only want
+ to move the channel pointers that are actually present. ........
+
+2009-03-25 15:33 +0000 [r184256] Eliel C. Sardanons <eliels@gmail.com>
+
+ * 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 14:40 +0000 [r184150-184221] Russell Bryant <russell@digium.com>
+
+ * main/timing.c, /: Merged revisions 184219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184219 |
+ russell | 2009-03-25 09:33:32 -0500 (Wed, 25 Mar 2009) | 2 lines
+ Include poll-compat.h ........
+
+ * main/timing.c, /: Merged revisions 184151 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184151 |
+ russell | 2009-03-24 21:03:13 -0500 (Tue, 24 Mar 2009) | 2 lines
+ Change poll() to ast_poll(). ........
+
+ * 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 [r184082] Mark Michelson <mmichelson@digium.com>
+
+ * 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 22:02 +0000 [r184041-184044] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: Merged revisions 184043 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r184043 |
+ russell | 2009-03-24 17:00:58 -0500 (Tue, 24 Mar 2009) | 2 lines
+ Put siren7 and siren14 in ast_best_codec() just so they're in
+ there somewhere. ........
+
+ * 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 15:29 +0000 [r183868-183917] Tilghman Lesher <tlesher@digium.com>
+
+ * /, 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 19:00 +0000 [r183769] Mark Michelson <mmichelson@digium.com>
+
+ * 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 [r183704] Leif Madsen <lmadsen@digium.com>
+
+ * 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 17:09 +0000 [r183564] Russell Bryant <russell@digium.com>
+
+ * 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. ........ ................
+
+2009-03-20 12:19 +0000 [r183519] Eliel C. Sardanons <eliels@gmail.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 183511 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) |
+ 2 lines Remove duplicate <description> inside the xml
+ documentation. ........
+
+2009-03-19 19:20 +0000 [r183337] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 18:20 +0000 [r183263] Russell Bryant <russell@digium.com>
+
+ * 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 18:12 +0000 [r183247] Mark Michelson <mmichelson@digium.com>
+
+ * 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 Leif Madsen <lmadsen@digium.com>
+
+ * Release Asterisk 1.6.2.0-beta1
+
+2009-03-19 16:11 +0000 [r183122] Mark Michelson <mmichelson@digium.com>
+
+ * /, 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 15:45 +0000 [r183068-183111] Joshua Colp <jcolp@digium.com>
+
+ * /, 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 [r183031] Jeff Peeler <jpeeler@digium.com>
+
+ * /, 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 14:39 +0000 [r182947] Russell Bryant <russell@digium.com>
+
+ * 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 20:53 +0000 [r182725] Jeff Peeler <jpeeler@digium.com>
+
+ * 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 16:46 +0000 [r182592] Russell Bryant <russell@digium.com>
+
+ * 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 15:01 +0000 [r182528-182534] Kevin P. Fleming <kpfleming@digium.com>
+
+ * 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 [r182453] Tilghman Lesher <tlesher@digium.com>
+
+ * 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 ........ ................
+
+2009-03-17 02:02 +0000 [r182409] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 182408 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009)
+ | 8 lines OPENR2 uses an incorrect string value if the extension
+ delimiter is not present. * Fixed OPENR2 using an incorrect
+ string value if the extension delimiter is not present in the
+ Dial() function. This was fixed for SS7 and PRI in trunk
+ -r172400. * Made OPENR2 stripmsd behavior the same as the SS7,
+ PRI, and others. * Removed trailing whitespace that appeared with
+ OPENR2. ........
+
+2009-03-16 20:51 +0000 [r182360-182361] Russell Bryant <russell@digium.com>
+
+ * /: svnmerge init
+
+ * / (added): Create a branch for 1.6.2
+
+2009-03-16 20:35 +0000 [r182355] Russell Bryant <russell@digium.com>
+
+ * CREDITS, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ CHANGES, makeopts.in: Add MFC/R2 support for chan_dahdi. This
+ commit introduces official support for R2 signaling in
+ chan_dahdi. The modifications to chan_dahdi, and the supporting
+ library, LibOpenR2, were both written by Moises Silva. Many users
+ are using this code, or a variant of it, in Asterisk 1.2, 1.4 and
+ 1.6 in Brazil, México and Argentina. An unknown number of users
+ (but at least 1) are using it in each of the following countries:
+ Colombia, Nepal, Thailand, Venezuela, Perú, and probably others.
+ To use this code, LibOpenR2 must be installed from
+ http://www.libopenr2.org/. Information about configuration can be
+ found in configs/chan_dahdi.conf.sample. The code committed is
+ the most up to date version, which was being maintained in
+ svn/asterisk/team/moy/mfcr2/. I would also like to include a
+ Thank You to the many others that tested this code beyond those
+ listed in this commit message. These are the names that I could
+ find in the mantis issue. (closes issue #12509) Reported by: moy
+ Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested
+ by: moy, korihor, viniciusfontes, Skarmeth, loloski,
+ asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare,
+ ecarruda, rtorresduque, PTorres, ychen Review:
+ http://reviewboard.digium.com/r/40/
+
+2009-03-16 17:49 +0000 [r182282] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: 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 17:33 +0000 [r182211-182278] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c: 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
+
+ * channels/chan_local.c, /: 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 ........
+
+2009-03-16 13:58 +0000 [r182171] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: 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 21:26 +0000 [r182029-182121] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Change faulty comparison used when announcing
+ average hold minutes and seconds (closes issue #14227) Reported
+ by: caspy
+
+ * main/features.c: Remove ast_ prefix from functions which are not
+ public.
+
+ * /, main/features.c: 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 17:25 +0000 [r182022] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: 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 16:55 +0000 [r181985] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: improve a bit of suboptimal code
+
+2009-03-13 01:26 +0000 [r181899] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Merged revisions 181898 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Just
+ recording the v1.4 change in trunk since it originally came from
+ here. ........ r181898 | rmudgett | 2009-03-12 20:19:29 -0500
+ (Thu, 12 Mar 2009) | 4 lines Use the correct branch integrated
+ property when generating the version string. Copied the
+ make_version file from Asterisk trunk. ........
+
+2009-03-12 21:43 +0000 [r181769-181846] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Run the macro on the queue member's channel
+ when he answers, not the caller's channel.
+
+ * /, channels/chan_sip.c: 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 17:32 +0000 [r181731] Tilghman Lesher <tlesher@digium.com>
+
+ * main/translate.c: 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 16:56 +0000 [r181612-181665] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_musiconhold.c: 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 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 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: Fix crash when sleep and retries argument was
+ not given to RetryDial application. (closes issue #14647)
+ Reported by: sherpya
+
+2009-03-12 01:33 +0000 [r181542-181577] Richard Mudgett <rmudgett@digium.com>
+
+ * build_tools/make_version: Whitespace chages.
+
+ * build_tools/make_version: Use the correct branch integrated
+ property when generating the version string
+
+2009-03-11 23:14 +0000 [r181499] Michiel van Baak <michiel@vanbaak.info>
+
+ * configs/sip.conf.sample: Provide correct hint to debug SIP
+ trouble in the default config (closes issue #14646) Reported by:
+ strk
+
+2009-03-11 22:25 +0000 [r181465] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Make handling of the BRIDGE_PLAY_SOUND variable
+ thread-safe.
+
+2009-03-11 22:20 +0000 [r181444] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac: 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 22:14 +0000 [r181424-181428] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Make handling of the BRIDGEPVTCALLID variable
+ thread-safe.
+
+ * main/channel.c, /: 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 17:34 +0000 [r181371] David Vossel <dvossel@digium.com>
+
+ * channels/iax2-parser.h, /, channels/chan_iax2.c: 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 17:26 +0000 [r181345] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: 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) ........
+
+2009-03-11 17:04 +0000 [r181301] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/astobj2.h: Turn off malloc debugging of astobj2,
+ since it apparently doesn't work too well during startup.
+
+2009-03-11 16:40 +0000 [r181296] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: 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 16:19 +0000 [r181292] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: Replace contents of this doc with a
+ pointer to its new home
+
+2009-03-11 14:28 +0000 [r181244] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix segfault when dialing a typo'd queue If
+ trying to dial a non-existent queue, there would be a segfault
+ when attempting to access q->weight, even though q was NULL. This
+ problem was introduced during the queue-reset merge and thus only
+ affects trunk. (closes issue #14643) Reported by: alecdavis
+
+2009-03-11 13:44 +0000 [r181210] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_confbridge.c: Don't play the "you are about to be placed
+ into the conference" and "the leader has left the conference"
+ sounds if the quiet option is enabled. (reported by Vadim Lebedev
+ on the asterisk-dev list)
+
+2009-03-11 04:06 +0000 [r181135] Jeff Peeler <jpeeler@digium.com>
+
+ * 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,
+ pbx/pbx_config.c: 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 02:25 +0000 [r181099] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: tabs to spaces
+
+2009-03-11 00:49 +0000 [r181032-181033] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add missing comment that quotes RFC 3891
+
+ * /, channels/chan_sip.c: 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. ........
+
+2009-03-11 00:29 +0000 [r181027-181028] Tilghman Lesher <tlesher@digium.com>
+
+ * main/strings.c, main/hashtab.c, include/asterisk/astobj2.h,
+ main/heap.c, include/asterisk/strings.h,
+ include/asterisk/hashtab.h, main/astobj2.c,
+ include/asterisk/heap.h: Add MALLOC_DEBUG to various utility
+ APIs, so that memory leaks can be tracked back to their source.
+ (related to issue #14636)
+
+ * main/pbx.c: Spacing changes only
+
+2009-03-10 22:03 +0000 [r180944] Jason Parker <jparker@digium.com>
+
+ * /, configure, configure.ac, autoconf/ast_prog_sed.m4,
+ autoconf/ast_check_gnu_make.m4: 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 22:03 +0000 [r180935-180942] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: Add some notes on getting in
+ contact with the dev community
+
+ * doc/google-soc2009-ideas.txt: Remove difficulty and language
+ specifiers
+
+ * doc/google-soc2009-ideas.txt: Expand upon documentation of
+ manager event project
+
+2009-03-10 21:15 +0000 [r180898] Michiel van Baak <michiel@vanbaak.info>
+
+ * CHANGES: list the move of the astvarrundir from /var/run to
+ /var/run/asterisk (actually its $(localstatedir)/run/asterisk
+ Makes setups with asterisk as non-root easier to manage because
+ you can setup permissions on this dir instead of touching a file
+ and setting permissions on that. Files that come to mind are
+ asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell
+
+2009-03-10 19:36 +0000 [r180859-180862] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: add more projects
+
+ * doc/google-soc2009-ideas.txt: add more project ideas
+
+2009-03-10 14:40 +0000 [r180800] Joshua Colp <jcolp@digium.com>
+
+ * main/manager.c: Reset the thread local string buffer when
+ handling the UserEvent action. (closes issue #14593) Reported by:
+ JimDickenson
+
+2009-03-09 22:00 +0000 [r180750] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt: Add current mentors list, and first
+ pass on a project list broken out of "PineMango" I will work on
+ adding projects that have been sent to be via email tomorrow.
+
+2009-03-09 20:58 +0000 [r180719] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/rtp.h, include/asterisk/extconf.h,
+ main/devicestate.c, include/asterisk/tcptls.h, main/enum.c,
+ include/asterisk/callerid.h, include/asterisk/doxyref.h,
+ include/asterisk/event.h, include/asterisk/audiohook.h,
+ include/asterisk/dsp.h, include/asterisk/timing.h,
+ include/asterisk/udptl.h, include/asterisk/dlinkedlists.h,
+ include/asterisk/utils.h, include/asterisk/devicestate.h,
+ include/asterisk/taskprocessor.h, include/asterisk/enum.h,
+ include/asterisk/astobj2.h, include/asterisk/config.h,
+ include/asterisk/channel.h, include/asterisk/manager.h,
+ include/asterisk/heap.h, include/asterisk/logger.h,
+ include/asterisk/http.h, include/asterisk/res_odbc.h,
+ include/asterisk/app.h, main/tcptls.c,
+ include/asterisk/linkedlists.h, include/asterisk/sched.h,
+ include/asterisk/datastore.h, include/asterisk/lock.h,
+ include/asterisk/pbx.h, include/asterisk/dnsmgr.h: 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 <ver number>
+ <description of changes> 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/
+
+2009-03-09 14:14 +0000 [r180684] Russell Bryant <russell@digium.com>
+
+ * doc/google-soc2009-ideas.txt (added): Add skeleton for GSoC ideas
+ list
+
+2009-03-07 15:36 +0000 [r180641] Russell Bryant <russell@digium.com>
+
+ * contrib/asterisk-ng-doxygen: Make some minor updates to the
+ doxygen configuration - add bridges directory to be processed -
+ add some res/ subdirs - alphabetize subdirs - use consistent
+ indentation
+
+2009-03-06 18:25 +0000 [r180579] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: 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 17:26 +0000 [r180534] David Vossel <dvossel@digium.com>
+
+ * /, main/enum.c: 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 23:26 +0000 [r180383-180465] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: 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 ........
+
+ * /, configs/voicemail.conf.sample, apps/app_voicemail.c: 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 19:05 +0000 [r180382] Michiel van Baak <michiel@vanbaak.info>
+
+ * Makefile: Make sure we terminate the first s| command so we can
+ actually produce correct files.
+
+2009-03-05 18:29 +0000 [r180373] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c, /, include/asterisk/frame.h, main/rtp.c: 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/
+ ........
+
+2009-03-05 18:18 +0000 [r180369] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_bridge.c (added), main/Makefile,
+ bridges/bridge_simple.c, bridges/bridge_softmix.c,
+ include/asterisk/channel.h, bridges/bridge_multiplexed.c,
+ CHANGES, Makefile, include/asterisk/bridging_technology.h
+ (added), bridges (added), bridges/bridge_builtin_features.c,
+ include/asterisk/bridging_features.h (added),
+ include/asterisk/bridging.h (added), apps/app_confbridge.c
+ (added), main/bridging.c (added), bridges/Makefile: Merge phase 1
+ support for the new bridging architecture. This commit brings in
+ the bridging core, bridging technologies, and the ConfBridge
+ application. For usage information on the ConfBridge application
+ please see the output of "core show application ConfBridge" from
+ the CLI. For API documentation please see the doxygen page
+ describing the architecture and the documentation for each API
+ call. Review: http://reviewboard.digium.com/r/93/
+
+2009-03-05 06:21 +0000 [r180304-180334] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/editors/asterisk.vim: Also highlight the preamble and
+ postamble
+
+ * contrib/editors/ael.vim (added), contrib/editors/asterisk.vim
+ (added), contrib/editors (added), contrib/editors/asteriskvm.vim
+ (added): Add syntax coloring files for Vim, including a new one
+ for AEL
+
+2009-03-04 21:01 +0000 [r180261] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: 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 20:48 +0000 [r180259] Tilghman Lesher <tlesher@digium.com>
+
+ * main/aescrypt.c, main/abstract_jb.c, main/acl.c, main/app.c,
+ main/alaw.c: Spacing changes only
+
+2009-03-04 19:24 +0000 [r180195] Joshua Colp <jcolp@digium.com>
+
+ * /, main/callerid.c: 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) ........
+
+2009-03-04 17:03 +0000 [r180155] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Allow for "magic"
+ pickups to work when we wish to ignore the context When the
+ subscription context for a call pickup subscription differs from
+ the context of the call pickup target, there's not an easy way to
+ divine what context should be used for the pickup. The way to
+ work around this is to use PICKUPMARK as the context for the
+ pickup. This has been documented in the sip.conf.sample file
+ (ABE-1708) closes issue #14567 submitted by: alecdavis
+
+2009-03-04 14:39 +0000 [r180120] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: 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 23:35 +0000 [r180079] Steve Murphy <murf@digium.com>
+
+ * utils/Makefile: My bad! left check_expr2 in the ALL_UTILS list by
+ mistake. Already done to 1.6.x
+
+2009-03-03 23:21 +0000 [r180032] David Vossel <dvossel@digium.com>
+
+ * main/channel.c, include/asterisk/app.h, apps/app_read.c,
+ main/app.c: 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 22:49 +0000 [r180007] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/queues.conf.sample, apps/app_queue.c: 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 ........
+
+2009-03-03 22:12 +0000 [r179973] Steve Murphy <murf@digium.com>
+
+ * utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h,
+ main/ast_expr2.y, main/ast_expr2f.c, main/ast_expr2.fl,
+ main/ast_expr2.c: 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 22:01 +0000 [r179972] David Vossel <dvossel@digium.com>
+
+ * apps/app_meetme.c: app_meetme not setting filename and fileformat
+ correctly for realtime When app_meetme finds a realtime
+ conference, it doesn't get the filename and fileformat correctly
+ when 'r' is set. Now app_meetme first checks to see if fileformat
+ and filename are declared in the db, if they're not it checks the
+ .conf file, if its not declared there either it then uses
+ defaults. (closes issue #14545) Reported by: dalbaech Patches:
+ app_meetme-realtime5.patch uploaded by dvossel (license 671)
+ Realtime_Conference_Record_workaround.txt uploaded by dalbaech
+ (license 705) Tested by: dvossel, dalbaech Review:
+ http://reviewboard.digium.com/r/180/
+
+2009-03-03 20:59 +0000 [r179937] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_timing_dahdi.c, doc/timing.txt (added): 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 20:02 +0000 [r179903] Brian Degenhardt <bmd@digium.com>
+
+ * apps/app_directed_pickup.c: fix a leaked channel lock (and future
+ deadlock) when we try to pick up our own channel
+
+2009-03-03 18:28 +0000 [r179841] Joshua Colp <jcolp@digium.com>
+
+ * /, main/features.c: 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 17:03 +0000 [r179745] Mark Michelson <mmichelson@digium.com>
+
+ * pbx/pbx_spool.c: Convert pbx_spool to use string fields instead
+ of statically-sized buffers. In tests run after making this
+ conversion, I noticed an approximate 85% reduction in memory
+ usage for call file processing. Review:
+ http://reviewboard.digium.com/r/168/
+
+2009-03-03 16:47 +0000 [r179742] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: 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 15:13 +0000 [r179675] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Please prefix default values with DEFAULT
+
+2009-03-03 14:40 +0000 [r179672] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: 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 13:54 +0000 [r179609] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, /: 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 00:01 +0000 [r179537] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /: 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 23:36 +0000 [r179533] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_meetme.c: 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 23:10 +0000 [r179469] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/app.c: 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 23:06 +0000 [r179462-179465] Russell Bryant <russell@digium.com>
+
+ * res/res_timing_timerfd.c: Fix a reference leak in
+ timerfd_set_rate(). (found during a debugging session with
+ dvossel and mmichelson.)
+
+ * main/channel.c, /: 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 20:16 +0000 [r179396] Jason Parker <jparker@digium.com>
+
+ * /, main/editline/configure, main/editline/np/unvis.c,
+ main/editline/sys.h, main/editline/configure.in: 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 17:18 +0000 [r179361] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: Backport 1.6.0 fix to trunk (failsafe
+ if db is not loaded)
+
+2009-03-02 14:28 +0000 [r179291-179323] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Do not try to remove a registration
+ scheduled item if the scheduler context has already been
+ destroyed. (closes issue #14580) Reported by: alecdavis
+
+ * main/audiohook.c: 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 23:25 +0000 [r179219-179254] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_speech_utils.c: Swap reversed timevals. This was pointed
+ out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
+
+ * channels/chan_sip.c: 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 21:47 +0000 [r179164] Russell Bryant <russell@digium.com>
+
+ * res/res_ais.c, doc/distributed_devstate.txt,
+ configs/ais.conf.sample: Mark res_ais as experimental, as the
+ binary event format is subject to change.
+
+2009-02-27 21:32 +0000 [r179161] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_sqlite3_custom.c: If config file is blank, don't load
+ module. (Closes issue #14563)
+
+2009-02-27 21:23 +0000 [r179154] Russell Bryant <russell@digium.com>
+
+ * UPGRADE.txt: Add a note about the ordering of entries in sip.conf
+ in 1.6.1.
+
+2009-02-27 20:34 +0000 [r179122] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Add reload support to chan_skinny.
+ Special thanks goes to DEA who had to redo this patch twice
+ because we first put unload/load support in and later redid the
+ way we configure devices and lines. (closes issue #10297)
+ Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by
+ wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA
+ (license 3) With mods by me based on feedback from wedhorn and
+ Russell and seanbright Tested by: DEA, mvanbaak, pj Review:
+ http://reviewboard.digium.com/r/130/
+
+2009-02-27 19:04 +0000 [r179057] Jason Parker <jparker@digium.com>
+
+ * doc/tex/channelvariables.tex: 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 15:51 +0000 [r179021] Russell Bryant <russell@digium.com>
+
+ * sounds/Makefile: Fix downloading SIREN7 and SIREN14 sound
+ packages. In passing, also fix downloading SLIN16 extra sound
+ packages. (closes issue #14565) Reported by: jtodd
+
+2009-02-27 03:45 +0000 [r178986] Steve Murphy <murf@digium.com>
+
+ * /, main/features.c, configs/features.conf.sample: 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 18:41 +0000 [r178919] Tilghman Lesher <tlesher@digium.com>
+
+ * main/features.c, CHANGES, configs/features.conf.sample: Sound
+ confirmation of call pickup success. (closes issue #13826)
+ Reported by: azielke Patches: pickupsound2-trunk.patch uploaded
+ by azielke (license 548) __20081124_bug_13826_updated.patch
+ uploaded by lmadsen (license 10) Tested by: lmadsen
+
+2009-02-26 17:46 +0000 [r178871] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: IAX2 prune realtime, minor tweak to last
+ fix A return statement was missing which caused unexpected cli
+ output. issue #14479
+
+2009-02-26 17:45 +0000 [r178828-178870] Steve Murphy <murf@digium.com>
+
+ * apps/app_osplookup.c, apps/app_rpt.c: These small fixes prevent
+ compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to
+ break my dev-mode build. Not a problem in 1.6.x.
+
+ * /, main/features.c: 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 16:42 +0000 [r178801] Joshua Colp <jcolp@digium.com>
+
+ * main/file.c: 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 15:50 +0000 [r178767] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: 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 <friend> 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/
+
+2009-02-26 15:40 +0000 [r178764] Joshua Colp <jcolp@digium.com>
+
+ * main/indications.c: Ensure there is a valid tone part before
+ trying to play tones. (closes issue #14558) Reported by:
+ alecdavis
+
+2009-02-26 15:02 +0000 [r178733] Olle Johansson <oej@edvina.net>
+
+ * configs/res_snmp.conf.sample: Clarifications on the different
+ models and reference to further docs.
+
+2009-02-26 13:39 +0000 [r178703-178704] Kevin P. Fleming <kpfleming@digium.com>
+
+ * README: another minor commit to test post-commit script changes
+ (now testing post-revprop-change as well, third try)
+
+ * README: minor commit to test post-commit script changes
+
+2009-02-25 19:49 +0000 [r178573-178607] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c: Picky, picky buildbots
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/stdtime/localtime.c: Use notification when timezone files
+ change and re-scan then. (closes issue #14300) Reported by:
+ jamessan Patches: 20090127__bug14300.diff.txt uploaded by
+ tilghman (license 14) 20090224__bug14300.diff uploaded by
+ jamessan (license 246) Tested by: jamessan Review:
+ http://reviewboard.digium.com/r/136/
+
+ * res/res_odbc.c: Oops, wrong direction of command
+
+2009-02-25 12:45 +0000 [r178509] Russell Bryant <russell@digium.com>
+
+ * /, main/asterisk.c: 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 23:27 +0000 [r178375-178446] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/extensions.conf.sample: 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: Apparently, a void cast doesn't override
+ warn_unused_result.
+
+ * main/asterisk.c: The 3 possible errors with pipe(2) are all
+ impossible in this situation.
+
+2009-02-24 20:39 +0000 [r178374] Russell Bryant <russell@digium.com>
+
+ * /, main/rtp.c: 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 20:06 +0000 [r178303-178342] Tilghman Lesher <tlesher@digium.com>
+
+ * utils/astcanary.c, main/asterisk.c: Use a SIGPIPE to kill the
+ process, instead of depending upon the astcanary process being
+ inherited by init.
+
+ * utils/astcanary.c: 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 17:42 +0000 [r178300] David Vossel <dvossel@digium.com>
+
+ * doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Allows
+ manager command to see if IAX link is trunked and encrypted.
+ Displays what kind of encryption is enabled as well. Manager
+ command "iaxpeers" now shows if a link is trunked and encrypted.
+ Instead of encryption saying simply "yes" or "no", it now
+ displays what type of encryption is enabled and if keyrotation is
+ on or not. (closes issue #14427) Reported by: snuffy Patches:
+ iax_show_trunks.diff uploaded by snuffy (license 35)
+ 2009022200_iax2_show_trunkencryption.diff.txt uploaded by
+ mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review:
+ http://reviewboard.digium.com/r/173/
+
+2009-02-24 15:18 +0000 [r178213] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: 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 23:11 +0000 [r178142] Russell Bryant <russell@digium.com>
+
+ * /, main/rtp.c: 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 ........
+
+2009-02-23 21:02 +0000 [r178107] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
+ Permit emailsubject and emailbody to be set per mailbox. (closes
+ issue #14372) Reported by: fhackenberger Patches:
+ voicemail_individual_subject_and_body_1.6.1 uploaded by
+ fhackenberger (license 592) with additional fixes by Corydon76
+ (license 14)
+
+2009-02-23 18:23 +0000 [r178061] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: update the new manager commands in
+ chan_skinny to match chan_sip's headers. requested by oej.
+
+2009-02-23 17:59 +0000 [r178030] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: Changes the way keyrotation is enabled by
+ default Key rotation was enabled by default by setting the global
+ encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this
+ is that if encryption is not enabled, and the encryption method
+ is set to anything except 0, the peer appears to have encryption
+ enabled when issuing a "iax2 show peers". Rather than have the
+ key rotation bit always set by default, it is now only set when
+ an encryption method is enabled. (closes issue #14523) Reported
+ by: mvanbaak
+
+2009-02-23 17:48 +0000 [r178027] Michiel van Baak <michiel@vanbaak.info>
+
+ * CHANGES: list the addition of the SKINNY manager actions in the
+ CHANGES file.
+
+2009-02-23 17:29 +0000 [r178022] Russell Bryant <russell@digium.com>
+
+ * tests/test_sched.c, main/sched.c: Fix a regression in scheduler
+ entry ordering, and add a regression test for it. (closes issue
+ #14522) Reported by: pj Tested by: russell
+
+2009-02-22 23:04 +0000 [r177988] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/manager_1_1.txt, channels/chan_skinny.c: Add a couple of
+ manager commands to chan_skinny Added: SKINNYdevices
+ SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521)
+ Reported by: mvanbaak Review:
+ http://reviewboard.digium.com/r/170/
+
+2009-02-21 15:59 +0000 [r177944] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: On update, test against the existence of
+ sipregs.
+
+2009-02-21 14:37 +0000 [r177913] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: add extra check for sysinfo/sysctl (closes issue
+ #14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff
+ uploaded by snuffy (license 35)
+
+2009-02-21 14:16 +0000 [r177884] Sean Bright <sean.bright@gmail.com>
+
+ * main/hashtab.c, include/asterisk/hashtab.h: Trailing whitespace,
+ minor coding guideline fixes, and start beefing up the hashtab
+ documentation a bit.
+
+2009-02-21 13:17 +0000 [r177855] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/indications.h: Fix build issues on Solaris and
+ OpenBSD. (closes issue #14512) Reported by: snuffy
+
+2009-02-21 13:13 +0000 [r177849-177852] Michiel van Baak <michiel@vanbaak.info>
+
+ * Makefile, contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.archlinux.asterisk,
+ contrib/scripts/safe_asterisk: set
+ ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path When
+ running asterisk as non-root and without this patch the pidfile
+ wants to go into /var/run/asterisk.pid. This directory is not
+ writable for the non-root user and changing permissions is not an
+ option. Putting it in /var/run/asterisk/asterisk.pid makes it
+ possible to set permissions on the /var/run/asterisk dir so
+ everything works as it should be. Patched committed is based on
+ pabelanger's patch. (closes issue #13153) Reported by: pabelanger
+ Patches: 2009012900_bug13153-nonrootscripts.diff.txt uploaded by
+ mvanbaak (license 7) Review: http://reviewboard.digium.com/r/139/
+
+ * channels/chan_sip.c: make chan_sip.c compile on OpenBSD again.
+
+2009-02-20 23:02 +0000 [r177732-177787] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, /: 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 ........
+
+ * /, include/asterisk/threadstorage.h: 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 20:29 +0000 [r177699] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * apps/app_fax.c: 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 17:29 +0000 [r177661-177664] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/app.h, main/app.c, apps/app_system.c: 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
+
+ * apps/app_voicemail.c: Oops, merge broke trunk
+
+2009-02-20 00:35 +0000 [r177624] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: Set sip_request ast_str data to NULL so
+ ast_str_copy allocates space properly in copy_request (issue
+ #14478) Reported by: erik_dedecker
+
+2009-02-19 23:56 +0000 [r177595] Steve Murphy <murf@digium.com>
+
+ * /, main/Makefile, main/ast_expr2f.c: 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 22:33 +0000 [r177506-177537] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: 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 ........
+
+ * doc/database_transactions.txt (added): Document how to use
+ database transactions
+
+2009-02-19 16:45 +0000 [r177387] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/channel.h: Fix another merge error from 176708
+
+2009-02-19 16:38 +0000 [r177384] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_speech_utils.c: 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 15:56 +0000 [r177356] Jeff Peeler <jpeeler@digium.com>
+
+ * main/features.c: Fix mismerge from revision 176708 pointed out by
+ Kaloyan Kovachev on the asterisk-dev mailing list. Thanks!
+
+2009-02-19 00:26 +0000 [r177320] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h, funcs/func_odbc.c, CHANGES,
+ res/res_odbc.c, configs/res_odbc.conf.sample: ODBC transaction
+ support
+
+2009-02-19 00:08 +0000 [r177291] Joshua Colp <jcolp@digium.com>
+
+ * CHANGES: Update CHANGES file to include MWI subscription support
+ that was added some time ago.
+
+2009-02-18 23:51 +0000 [r177287] Tilghman Lesher <tlesher@digium.com>
+
+ * main/strings.c: Handle negative length and eliminate a condition
+ that is always true.
+
+2009-02-18 23:50 +0000 [r177286] Steve Murphy <murf@digium.com>
+
+ * /, res/ael/ael.tab.c, res/ael/ael.y: 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 23:09 +0000 [r177229] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/frame.c: 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 22:51 +0000 [r177226] David Vossel <dvossel@digium.com>
+
+ * main/features.c: 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 20:11 +0000 [r177162] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4,
+ channels/h323/cisco-h225.h, channels/h323/caps_h323.cxx,
+ channels/h323/ast_h323.cxx, channels/h323/ast_ptlib.h (added),
+ configure, channels/h323/compat_h323.h, configure.ac,
+ channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
+ channels/h323/ast_h323.h, channels/h323/chan_h323.h,
+ channels/h323/cisco-h225.cxx: 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 19:12 +0000 [r177101] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: 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 19:05 +0000 [r177098] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/config.h: 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 17:24 +0000 [r177035] Doug Bailey <dbailey@digium.com>
+
+ * main/utils.c: Fixed error where a check for an zero length,
+ terminated string was needed.
+
+2009-02-18 17:11 +0000 [r177005] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: 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 16:09 +0000 [r176948] Doug Bailey <dbailey@digium.com>
+
+ * main/utils.c: Need to take into account the \0 terminator of the
+ old string to determine the amount available.
+
+2009-02-18 15:35 +0000 [r176943] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: 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 06:14 +0000 [r176901-176904] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/heap.h: Add example code for a heap traversal.
+
+ * main/pbx.c: 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 02:55 +0000 [r176869] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * channels/chan_sip.c: T38 faxdetect should jump to the 'fax'
+ extension for incoming calls only The previous implementation of
+ T38 faxdetect resulted in both sides of the call jumping to a fax
+ extension when both sides had 't38pt_udptl=yes' and
+ 'faxdetect=yes' in sip.conf and a 'fax' extension in the current
+ context. This revision will jump to a 'fax' extension on incoming
+ calls only.
+
+2009-02-18 02:02 +0000 [r176841] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/rtp.c: suppress smoothers for Siren codecs as well as Speex
+ and G.723.1
+
+2009-02-17 22:52 +0000 [r176771] Russell Bryant <russell@digium.com>
+
+ * apps/app_milliwatt.c: Remove a dependency that no longer exists.
+
+2009-02-17 22:28 +0000 [r176760] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: 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 22:08 +0000 [r176708] Jeff Peeler <jpeeler@digium.com>
+
+ * main/channel.c, /, main/features.c, include/asterisk/channel.h:
+ 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 22:02 +0000 [r176706] Mark Michelson <mmichelson@digium.com>
+
+ * tests/test_sched.c: Use constants from inttypes.h to clear up
+ 32-bit compilation errors
+
+2009-02-17 21:59 +0000 [r176705] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * channels/chan_sip.c: 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 21:40 +0000 [r176697] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/frame.h: Clear up documentation of
+ AST_FRIENDLY_OFFSET in frame.h
+
+2009-02-17 21:23 +0000 [r176669] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Recorded merge of revisions 176661 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009)
+ | 9 lines Backport change to 1.4: 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 21:22 +0000 [r176666] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
+ res/res_timing_timerfd.c, include/asterisk/timing.h,
+ main/timing.c: 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 21:14 +0000 [r176642] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: 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 21:04 +0000 [r176632-176639] Russell Bryant <russell@digium.com>
+
+ * tests/test_sched.c (added), main/sched.c: Significantly improve
+ scheduler performance under high load. This patch changes the
+ scheduler to use a max-heap to store pending scheduler entries
+ instead of a fully sorted doubly linked list. When the number of
+ entries in the scheduler gets large, this will perform much
+ better. For much more detailed information on this change, see
+ the review request. Review: http://reviewboard.digium.com/r/160/
+
+ * tests/test_heap.c (added): Add a test module for the heap
+ implementation. Review: http://reviewboard.digium.com/r/160/
+
+ * main/Makefile, main/heap.c (added), include/asterisk/heap.h
+ (added): 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/
+
+2009-02-17 20:50 +0000 [r176631] Olle Johansson <oej@edvina.net>
+
+ * include/asterisk/config.h: Typo
+
+2009-02-17 20:41 +0000 [r176627] Russell Bryant <russell@digium.com>
+
+ * channels/chan_unistim.c, main/pbx.c, apps/app_read.c,
+ configs/indications.conf.sample, apps/app_playtones.c (added),
+ include/asterisk/indications.h, apps/app_readexten.c,
+ apps/app_disa.c, UPGRADE.txt, include/asterisk/channel.h,
+ include/asterisk/_private.h, main/indications.c, main/loader.c,
+ main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
+ funcs/func_channel.c, res/snmp/agent.c, main/app.c,
+ res/res_indications.c (removed), main/asterisk.c: Merge a large
+ set of updates to the Asterisk indications API. This patch
+ includes a number of changes to the indications API. The primary
+ motivation for this work was to improve stability. The object
+ management in this API was significantly flawed, and a number of
+ trivial situations could cause crashes. The changes included are:
+ 1) Remove the module res_indications. This included the critical
+ functionality that actually loaded the indications configuration.
+ I have seen many people have Asterisk problems because they
+ accidentally did not have an indications.conf present and loaded.
+ Now, this code is in the core, and Asterisk will fail to start
+ without indications configuration. There was one part of
+ res_indications, the dialplan applications, which did belong in a
+ module, and have been moved to a new module, app_playtones. 2)
+ Object management has been significantly changed. Tone zones are
+ now managed using astobj2, and it is no longer possible to crash
+ Asterisk by issuing a reload that destroys tone zones while they
+ are in use. 3) The API documentation has been filled out. 4) The
+ API has been updated to follow our naming conventions. 5) Various
+ bits of code throughout the tree have been updated to account for
+ the API update. 6) Configuration parsing has been mostly
+ re-written. 7) "Code cleanup" The code is from
+ svn/asterisk/team/russell/indications/. Review:
+ http://reviewboard.digium.com/r/149/
+
+2009-02-17 18:49 +0000 [r176592] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c, res/res_odbc.c: Add assertions in the quest to
+ track down a refcount leak. (closes issue #14485) Reported by:
+ davevg
+
+2009-02-17 17:33 +0000 [r176557] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c, apps/app_queue.c: 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 17:28 +0000 [r176513-176556] Olle Johansson <oej@edvina.net>
+
+ * configs/extconfig.conf.sample: Typo
+
+ * main/config.c: If there are no realtime engines, there's no
+ reason to check for realtime families
+
+2009-02-17 14:39 +0000 [r176360-176501] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: In this version, we can combine the queries,
+ because we support dropping nonexistent columns.
+
+ * /, channels/chan_sip.c: 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 ........
+
+ * main/strings.c: Might want to update the buffer pointer after a
+ realloc (or we crash) (closes issue #14485) Reported by: davevg
+
+2009-02-16 23:37 +0000 [r176356] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/sounds.xml: add support for Siren7 and Siren14 flavors of
+ prompts and music on hold
+
+2009-02-16 23:33 +0000 [r176355] David Vossel <dvossel@digium.com>
+
+ * /, channels/chan_iax2.c: 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 23:14 +0000 [r176320] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_skinny.c: 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 21:45 +0000 [r176255] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, main/utils.c, include/asterisk/stringfields.h: 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 21:40 +0000 [r176253] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_meetme.c: 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 21:30 +0000 [r176248] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: 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 18:25 +0000 [r176174] Mark Michelson <mmichelson@digium.com>
+
+ * main/logger.c: 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 17:44 +0000 [r176138] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Can't set debug level 2 (intense
+ debugging) unless the syntax matches
+
+2009-02-16 17:09 +0000 [r176100] Russell Bryant <russell@digium.com>
+
+ * channels/chan_features.c (removed): Remove chan_features. Review:
+ http://reviewboard.digium.com/r/161/
+
+2009-02-16 15:36 +0000 [r176030] Joshua Colp <jcolp@digium.com>
+
+ * /, channels/chan_sip.c: 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 02:54 +0000 [r175983] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Make the causes array static, and remove the type
+ name as it is not needed.
+
+2009-02-16 00:26 +0000 [r175952] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_unistim.c, /, channels/chan_sip.c,
+ include/asterisk/manager.h, doc/unistim.txt: 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 21:27 +0000 [r175829-175882] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/sched.h, main/sched.c: Make ast_sched_report()
+ and ast_sched_dump() thread safe.
+
+ * channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: 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.
+
+2009-02-15 20:39 +0000 [r175783-175827] Olle Johansson <oej@edvina.net>
+
+ * formats/format_ilbc.c, /: Merged revisions 175825 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb
+ 2009) | 2 lines format_ilbc does not depend on codec libraries
+ and can therefore always be made. My mistake. Ursäkta! ........
+
+ * formats/format_ilbc.c, /: Merged revisions 175792 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb
+ 2009) | 2 lines Disable format_ilbc.so by default, like
+ codec_ilbc.so ........
+
+ * /, channels/chan_sip.c: Merged revisions 175777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2
+ lines Make sure that the debug line is not printed on debug level
+ 0 ........
+
+2009-02-13 20:57 +0000 [r175655-175663] Mark Michelson <mmichelson@digium.com>
+
+ * doc/manager_1_1.txt, CHANGES, apps/app_queue.c: Merge queue-reset
+ branch to Asterisk From a user point-of-view, this adds new CLI
+ commands and Manager Actions to better facilitate the reloading
+ of queues and the resetting of their statistics. The new CLI
+ commands are the "queue reload" and "queue reset stats" commands.
+ The new manager actions are the QueueReload and QueueReset
+ commands. Review: http://reviewboard.digium.com/r/115
+
+ * doc/manager_1_1.txt, apps/app_chanspy.c: Add manager events for
+ chanspy starting or stopping (closes issue #14469) Reported by:
+ caio1982 Patches: chanspy_events2.diff uploaded by caio1982
+ (license 22)
+
+2009-02-13 20:26 +0000 [r175623-175636] Russell Bryant <russell@digium.com>
+
+ * res/res_jabber.c: fix a few more XML documentation problems
+
+ * main/pbx.c: add missing </para>
+
+2009-02-13 20:11 +0000 [r175597] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, channels/iax2.h, channels/chan_iax2.c:
+ 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. Review:
+ http://reviewboard.digium.com/r/159/
+
+2009-02-13 19:49 +0000 [r175591] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_voicemail.c: 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 16:41 +0000 [r175549] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_record.c: Add an option to keep the recorded file upon
+ hangup. (closes issue #14341) Reported by: fnordian
+
+2009-02-13 13:41 +0000 [r175508-175512] Kevin P. Fleming <kpfleming@digium.com>
+
+ * CHANGES: document G.722.1/.1C support
+
+ * main/frame.c, channels/chan_sip.c, include/asterisk/rtp.h,
+ channels/chan_h323.c, include/asterisk/frame.h,
+ formats/format_siren14.c (added), main/rtp.c,
+ formats/format_siren7.c (added): Add basic (passthrough,
+ playback, record) support for ITU G.722.1 and G.722.1C (also
+ known as Siren7 and Siren14) This patch adds passthrough, file
+ recording and file playback support for the codecs listed above,
+ with negotiation over SIP/SDP supported. Due to Asterisk's
+ current limitation of treating a codec/bitrate combination as a
+ unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are
+ supported. Along the way, some related work was done: 1) The
+ rtpPayloadType structure definition, used as a return result for
+ an API call in rtp.h, was moved from rtp.c to rtp.h so that the
+ API call was actually usable. The only previous used of the API
+ all was chan_h323.c, which had a duplicate of the structure
+ definition instead of doing it the right way. 2) The hardcoded
+ SDP sample rates for various codecs in chan_sip.c were removed,
+ in favor of storing these sample rates in rtp.c along with the
+ codec definitions there. A new API call was added to allow
+ retrieval of the sample rate for a given codec. 3) Some basic
+ 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip
+ *must* decline any media streams offered for these codecs that
+ are not at the bitrates that we support (otherwise Bad Things
+ (TM) would result). Review: http://reviewboard.digium.com/r/158/
+
+2009-02-13 04:22 +0000 [r175411-175475] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * CHANGES: add 'faxbuffers' configuration option information to
+ CHANGES
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
+ dynamic fax buffer configuration option to chan_dahdi.conf When
+ the 'faxdetect' configuration option is used, one may also want
+ to use the 'faxbuffers' configuration option in chan_dahdi.conf.
+ This option will dynamically use the configured 'faxbuffers'
+ buffer policy on a channel for the life of the call following the
+ detection of fax tones. The faxbuffers buffer policy will be
+ reverted during call teardown. An example use of 'faxbuffers' is
+ below. This example would switch to using 6 buffers with a full
+ buffer policy. faxbuffers=>6,full
+
+2009-02-12 21:41 +0000 [r175368] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Remove useless string copy, and make sscanf
+ safe again
+
+2009-02-12 21:27 +0000 [r175344] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
+ force encryption option to iax.conf This patch adds
+ forceencryption=yes as an iax.conf option. When force encryption
+ is enabled, no unencrypted connections are allowed. This insures
+ all connections are encrypted. This is a new feature, so CHANGES
+ and iax.conf.sample are updated as well. (closes issue #13285)
+ Reported by: sgofferj Tested by: russell Review:
+ http://reviewboard.digium.com/r/150/
+
+2009-02-12 21:25 +0000 [r175334] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c, /: 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 20:48 +0000 [r175298] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: 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 20:45 +0000 [r175255-175295] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Avoid using ast_strdupa() in a loop.
+
+ * build_tools/cflags.xml: 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 18:48 +0000 [r175250] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: correct warning message to not refer
+ specifically to DAHDI
+
+2009-02-12 18:00 +0000 [r175188] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c: 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 17:07 +0000 [r175127] David Vossel <dvossel@digium.com>
+
+ * channels/chan_iax2.c: 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 16:57 +0000 [r175125] Russell Bryant <russell@digium.com>
+
+ * /, main/rtp.c: 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 16:28 +0000 [r175121] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/astobj2.h, main/astobj2.c: Make lock information
+ for ao2_trylock be more useful and gnarly Core show locks
+ information involving an ao2_trylock did not show the function
+ that called ao2_trylock, but would instead show ao2_trylock as
+ the source of the lock. This is not useful when trying to debug
+ locking issues. One bizarre note is that this logic is already in
+ 1.4 but somehow did not get merged to trunk or the 1.6.X
+ branches.
+
+2009-02-12 14:25 +0000 [r175058-175089] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Issue a warning message if our candidate's
+ IP is the loopback address. (closes issue #13985) Reported by:
+ jcovert Tested by: phsultan
+
+ * /, channels/chan_gtalk.c: 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 ........
+
+2009-02-11 23:12 +0000 [r174945-174951] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix a bit of odd logic for announcing position.
+ Sync with 1.6.0's logic
+
+ * apps/app_queue.c: Fix odd "thank you" sound playing behavior in
+ app_queue.c If someone has configured the queue to play an
+ position or holdtime announcement, then it is odd and potentially
+ unexpected to hear a "Thank you for your patience" sound when no
+ position or holdtime was actually announced. This fixes the
+ announcement so that the "thanks" sound is only played in the
+ case that a position or holdtime was actually announced. There is
+ a way that the "thank you" sound can be played without a position
+ or holdtime, and that is to set announce-frequency to a value but
+ keep announce-position and announce-holdtime both turned off.
+ (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
+ uploaded by putnopvut (license 60) Tested by: caspy
+
+ * apps/app_dial.c, main/channel.c, main/pbx.c, apps/app_dictate.c,
+ apps/app_waitforsilence.c, include/asterisk/channel.h: Fix 'd'
+ option for app_dial and add new option to Answer application The
+ 'd' option would not work for channel types which use RTP to
+ transport DTMF digits. The only way to allow for this to work was
+ to answer the channel if we saw that this option was enabled. I
+ realized that this may cause issues with CDRs, specifically with
+ giving false dispositions and answer times. I therefore modified
+ ast_answer to take another parameter which would tell if the CDR
+ should be marked answered. I also extended this to the Answer
+ application so that the channel may be answered but not CDRified
+ if desired. I also modified app_dictate and app_waitforsilence to
+ only answer the channel if it is not already up, to help not
+ allow for faulty CDR answer times. All of these changes are going
+ into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the
+ changes except for the change to the Answer application will go
+ in since we do not introduce new features into stable branches
+ (closes issue #14164) Reported by: DennisD Patches: 14164.patch
+ uploaded by putnopvut (license 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/145
+
+2009-02-11 14:44 +0000 [r174844] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Tell the device state core a change happened when
+ a channel is freed but not a specific state. We need to do this
+ because while we know that the freeing of the channel may cause
+ something to become not in use we do not know this for sure.
+ There may be another channel that is still up which would cause
+ it to be in use. (closes issue #13238) Reported by: kowalma
+ Patches: 20090121__bug13238.diff.txt uploaded by Corydon76
+ (license 14) Tested by: alecdavis
+
+2009-02-10 23:17 +0000 [r174764-174805] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Fix potential for stack overflows in
+ app_chanspy.c When using the 'g' or 'e' options, the stack
+ allocations that were used could cause a stack overflow if a
+ spyer stayed on the line long enough without actually
+ successfully spying on anyone. The problem has been corrected by
+ using static buffers and copying the contents of the appropriate
+ strings into them instead of using functions like alloca or
+ ast_strdupa
+
+ * main/manager.c: Fix an fd leak that would occur in HTTP AMI
+ sessions The explanation behind this fix is a bit complicated,
+ and I've already typed it up in the code as a huge comment inside
+ of manager.c, so I'll give the abridged version here. We needed a
+ way to separate action-specific data from session-specific data.
+ Unfortunately, the only way to maintain API compatibility and to
+ not have to change every single manager action was to rename the
+ current mansession structure and wrap it inside a new mansession
+ structure which actually contains action- specific data. (closes
+ issue #14364) Reported by: awk Patches: 14364_better.patch
+ uploaded by putnopvut (license 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/148/
+
+2009-02-10 20:15 +0000 [r174710] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only decrease inringing count if above zero.
+ (issue #13238) Reported by: kowalma
+
+2009-02-10 19:38 +0000 [r174705] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/slinfactory.c, include/asterisk/slinfactory.h: improve
+ slinfactory API to remove implicit sample rate and require
+ explicit sample rate selection by creator of the slinfactory
+
+2009-02-10 18:16 +0000 [r174584] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, main/jitterbuf.c: Merged revisions 174583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb
+ 2009) | 18 lines Improve behavior of jitterbuffer when
+ maxjitterbuffer is set. This change improves the way the
+ jitterbuffer handles maxjitterbuffer and dramatically reduces the
+ number of frames dropped when maxjitterbuffer is exceeded. In the
+ previous jitterbuffer, when maxjitterbuffer was exceeded, all new
+ frames were dropped until the jitterbuffer is empty. This change
+ modifies the code to only drop frames until maxjitterbuffer is no
+ longer exceeded. Also, previously when maxjitterbuffer was
+ exceeded, dropped frames were not tracked causing stats for
+ dropped frames to be incorrect, this change also addresses that
+ problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
+ by mnicholson (license 96) Tested by: mnicholson Review:
+ http://reviewboard.digium.com/r/144/ ........
+
+2009-02-10 17:48 +0000 [r174543-174580] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Set the type for the peer structure to be a
+ peer as the default. (closes issue #14447) Reported by: triccyx
+
+ * channels/chan_sip.c: Make the logic for inuse and inringing
+ manipluation match that of 1.4. The old broken logic would reset
+ the values back to 0 during certain scenarios causing the wrong
+ state to be reported. (closes issue #14399) Reported by: caspy
+ (issue #13238) Reported by: kowalma
+
+2009-02-10 07:06 +0000 [r174470-174503] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_stack.c, apps/app_voicemail.c: Fix0ring build
+
+ * apps/app_stack.c: Remove the usage of the KeepAlive app, as it no
+ longer exists.
+
+2009-02-10 04:49 +0000 [r174370-174435] Steve Murphy <murf@digium.com>
+
+ * apps/app_rpt.c: This patch removes the use of AST_PBX_KEEPALIVE
+ from app_rpt.c. (closes issue #14435) Reported by: D_McNaul
+
+ * apps/app_rpt.c: More intptr_t work.
+
+ * /, apps/app_rpt.c: Merged revisions 174369 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5
+ lines This patch solves some compiler complaints in both 32 and
+ 64-bit environments. ........
+
+2009-02-09 17:27 +0000 [r174327] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix something I messed up in the merge I
+ just did
+
+2009-02-09 17:26 +0000 [r174325] David Vossel <dvossel@digium.com>
+
+ * apps/app_externalivr.c: Fixes issue with hangups not being sent
+ and external process never terminating. The ignore_hangup,
+ run_dead, and noanswer flags were never initilized to zero
+ causing hangups to never be issued. If the external script
+ expects to be notified of a hangup and never receives one, it
+ runs indefinitely. (closes issue #14251) Reported by: chris-mac
+ Tested by: dvossel
+
+2009-02-09 17:20 +0000 [r174301] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
+ 2009) | 12 lines Don't do an SRV lookup if a port is specified
+ RFC 3263 says to do A record lookups on a hostname if a port has
+ been specified, so that's what we're going to do. See section
+ 4.2. (closes issue #14419) Reported by: klaus3000 Patches:
+ patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000
+ (license 65) ........
+
+2009-02-09 14:49 +0000 [r174219] Joshua Colp <jcolp@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 174218 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb
+ 2009) | 4 lines Don't overwrite our pointer to the music class
+ when music on hold stops. We will use this if it starts again to
+ see if we can resume the music where it left off. (closes issue
+ #14407) Reported by: mostyn ........
+
+2009-02-07 16:16 +0000 [r174149] Russell Bryant <russell@digium.com>
+
+ * /, res/snmp/agent.c: Merged revisions 174148 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
+ | 2 lines Fix a race condition that could cause a crash. ........
+
+2009-02-06 23:51 +0000 [r174084] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
+ | 5 lines check ast_strlen_zero() before calling ast_strdupa() in
+ sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter
+ didn't actually upload a properly-formed patch, instead a
+ modified chan_sip.c file was uploaded. I created a patch to
+ determine the changes, then modified the suggested changes to
+ create a proper fix. The summary above is a complete description
+ of the changes. (closes issue #13547) Reported by: tecnoxarxa
+ Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258)
+ Tested by: tecnoxarxa ........
+
+2009-02-06 20:12 +0000 [r174046] David Vossel <dvossel@digium.com>
+
+ * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
+ immediate yes/no option to iax.conf This is very similar to the
+ DAHDI immediate=yes option. When the phone is picked up, instead
+ of giving a dialtone it connects directly to the "s" extension.
+ Changes where implemented in chan_iax2.c to directly connect to
+ the "s" extension in the appropriate context when this option is
+ enabled. Examples explaining its use are added to
+ iax2.conf.sample. CHANGES has been updated as well. (closes issue
+ #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk
+ uploaded by jcovert (license 551) iax.conf.sample.patch uploaded
+ by jcovert (license 551) Tested by: jcovert, dvossel Review:
+ http://reviewboard.digium.com/r/143/
+
+2009-02-06 19:28 +0000 [r173974-174041] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_dahdi.c: Don't subscribe to a mailbox on pseudo
+ channels. It is futile. This solves an issue where duplicated
+ pseudo channels would cause a crash because the first one would
+ unsubscribe and the next one would also try to unsubscribe the
+ same subscription. (closes issue #14322) Reported by: amessina
+
+ * /, channels/chan_sip.c: Merged revisions 173967-173968 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
+ lines Some clients do not put the call-id for replaces at the
+ beginning, so support it being anywhere in the string. (closes
+ issue #14350) Reported by: fhackenberger ........ r173968 | file
+ | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
+ debug message I put in by accident. ........
+
+2009-02-06 16:28 +0000 [r173952] Matthew Nicholson <mnicholson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 173917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
+ 2009) | 7 lines Limit the addition of the Contact header in SIP
+ responses according to various SIP RFCs. (closes issue #13602)
+ Reported by: hjourdain Tested by: mnicholson ........
+
+2009-02-06 15:59 +0000 [r173902] Joshua Colp <jcolp@digium.com>
+
+ * main/audiohook.c, apps/app_chanspy.c: Always detach and destroy
+ the whisper and barge audiohooks. Additionally also allow an
+ audiohook to be detached if it has not been attached. (closes
+ issue #14414) Reported by: bluecrow76
+
+2009-02-06 10:55 +0000 [r173848-173858] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c: Add
+ a common implementation of a scheduler context with a dedicated
+ thread. This commit expands the Asterisk scheduler API to include
+ a common implementation of a scheduler context being processed by
+ a dedicated thread. chan_iax2 has been updated to use this new
+ code. Also, as a result, this resolves some race conditions
+ related to the previous chan_iax2 scheduler handling. Related to
+ rev 171452 which resolved the same issues in 1.4. Code from
+ team/russell/sched_thread2 Review:
+ http://reviewboard.digium.com/r/129/
+
+ * main/manager.c: Resolve a memory leak that would occur on an
+ invalid channel given to Action: Status
+
+2009-02-05 23:48 +0000 [r173773-173776] Mark Michelson <mmichelson@digium.com>
+
+ * configs/extensions.conf.sample: Update extensions.conf.sample to
+ be correct. In trunk, the only necessary change pointed out was
+ that the call to ChanIsAvail uses an option that has been
+ removed. For the 1.6.1 branch, however, it appears that the
+ sample file is badly in need of updating since there are |'s used
+ all over the place there. My tentative plan is just to copy
+ trunk's sample config file to those branches since the info there
+ is most up-to-date and should be correct for use in 1.6.1 Thanks
+ to macli in #asterisk-dev for bringing this up
+
+ * apps/app_voicemail.c: Properly set "seen" and "unseen" flags when
+ moving messages from the new to the old folder when using IMAP
+ for voicemail storage (closes issue #13905) Reported by: jaroth
+ Patches: foldermove_v2.patch uploaded by jaroth (license 50)
+
+2009-02-05 21:00 +0000 [r173697] Jeff Peeler <jpeeler@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 173696 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05
+ Feb 2009) | 12 lines Add new configuration option to make shared
+ IMAP mailboxes function as expected. The new option is
+ "imapvmshareid" which is an ID to tag multiple mailboxes using
+ the same IMAP storage location to function as one mailbox. This
+ allows all messages to be retrieved for any user in the group.
+ The patch alters the 'X-Asterisk-VM-Extension' header that is
+ responsible for matching voicemails for a given user. (closes
+ issue #13673) Reported by: howardwilkinson ........
+
+2009-02-05 20:30 +0000 [r173693] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 173692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb
+ 2009) | 12 lines Fix situations where queue members could be
+ autopaused unexpectedly Specifically, this patch prevents us from
+ autopausing members when we receive a busy or congestion frame
+ from them. (closes issue #14376) Reported by: fiddur Patches:
+ 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
+ ........
+
+2009-02-05 19:36 +0000 [r173657] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_sqlite.c: Change the first field, or we don't get
+ the necessary field separation.
+
+2009-02-05 18:48 +0000 [r173507-173593] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_mixmonitor.c: Merged revisions 173592 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu,
+ 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor
+ ........
+
+ * /, apps/app_mixmonitor.c: Merged revisions 173559 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu,
+ 05 Feb 2009) | 25 lines Fix a problem where a channel pointer
+ becomes invalid due to masquerading or hanging up. app_mixmonitor
+ runs its own thread to monitor the channel's activity and write
+ the mixed audio to a file. Since this thread runs independently
+ of the channel, it is possible that the mixmonitor thread's
+ channel pointer will point to freed memory when the channel
+ either is masqueraded or hangs up (technically, both cases are
+ hangups, but we need to handle the cases slightly differently).
+ The solution for this is to employ a datastore, which has the
+ nice benefit of allowing us to hook into channel masquerades and
+ hangups and update our pointer as necessary. If this looks
+ familiar, this same technique is employed in app_chanspy.
+ app_chanspy is a bit more involved since it does a lot more
+ operations on the channel that is being spied upon.
+ app_mixmonitor does have an extra touch that app_chanspy doesn't
+ have, though. Since there is a thread race between the channel's
+ thread and the mixmonitor thread on a hangup, we em- ploy a
+ condition-and-boolean combination to ensure that the channel
+ thread finishes with our structure before the mixmonitor thread
+ attempts to free it. No crashes! (closes issue #14374) Reported
+ by: aragon Patches: 14374.patch uploaded by putnopvut (license
+ 60) Tested by: aragon, putnopvut ........
+
+ * apps/app_queue.c: Fix some areas where the incorrect interface
+ was passed to ast_device_state I swear it feels like I already
+ did this once... (closes issue #14359) Reported by: francesco_r
+
+2009-02-04 21:26 +0000 [r173503] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_jabber.c: Add XML documentation for the applications and
+ functions in res_jabber (closes issue #14405) Reported by: snuffy
+ Patches: xml_jabber.diff uploaded by snuffy (license 35)
+
+2009-02-04 21:25 +0000 [r173502] David Vossel <dvossel@digium.com>
+
+ * channels/iax2-parser.h, channels/chan_iax2.c: Fixes issue with
+ IAX2 transfer not handing off calls. Reverts changes in 116884
+ Fixes issue with IAX2 transfers not taking place. As it was, a
+ call that was being transfered would never be handed off
+ correctly to the call ends because of how call numbers were
+ stored in a hash table. The hash table, "iax_peercallno_pvt",
+ storing all the current call numbers did not take into account
+ the complications associated with transferring a call, so a
+ separate hash table was required. This second hash table
+ "iax_transfercallno_pvt" handles calls being transfered, once the
+ call transfer is complete the call is removed from the transfer
+ hash table and added to the peer hash table resuming normal
+ operations. Addition functions were created to handle storing,
+ removing, and comparing items in the iax_transfercallno_pvt
+ table. The changes reverted in 116884 caused backwards
+ compatibility issues involving iax2 transfer with 1.6.0, 1.4, and
+ 1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel
+
+2009-02-04 21:17 +0000 [r173500] Jeff Peeler <jpeeler@digium.com>
+
+ * /, main/features.c, include/asterisk/features.h: Merged revisions
+ 173211 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009)
+ | 17 lines Parking attempts made to one end of a bridge no longer
+ will hang up due to a parking failure. Parking attempts made
+ using either one-touch, or doing either a blind or assisted
+ transfer to the parking extension now keep up the bridge instead
+ of hanging up the attempted parked party. Normal causes for the
+ parking attempt to fail includes the specific specified extension
+ (via PARKINGEXTEN) not being available or if all the parking
+ spaces are currently in use. To avoid having to reverse a
+ masquerade park_space_reserve was made to provide foresight if a
+ parking attempt will succeed and if so reserve the parking space.
+ (closes issue #13494) Reported by: mdu113 Reviewed by Russell:
+ http://reviewboard.digium.com/r/133/ ........
+
+2009-02-04 18:48 +0000 [r173458] Tilghman Lesher <tlesher@digium.com>
+
+ * main/tcptls.c: When using a socket as a FILE *, the stdio
+ functions will sometimes try to do an fseek() on the stream,
+ which is an invalid operation for a socket. Turning off buffering
+ explicitly lets the stdio functions know they cannot do this,
+ thus avoiding a potential error. (closes issue #14400) Reported
+ by: fnordian Patches: tcptls.patch uploaded by fnordian (license
+ 110)
+
+2009-02-04 17:45 +0000 [r173354-173397] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 173396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb
+ 2009) | 3 lines Revert my previous change because it was stupid
+ ........
+
+ * /, apps/app_chanspy.c: Merged revisions 173392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb
+ 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever
+ matter, but it's needed. ........
+
+ * main/file.c: Fix a problem where file playback would cause fds to
+ remain open forever The problem came from the fact that a frame
+ read from a format interpreter was not freed. Adding a call to
+ ast_frfree fixed this. The explanation for why this caused the
+ problem is a bit complex, but here goes: There was a problem in
+ all versions of Asterisk where the embedded frame of a filestream
+ structure was referenced after the filestream was freed. This was
+ fixed by adding reference counting to the filestream structure.
+ The refcount would increase every time that a filestream's frame
+ pointer was pointing to an actual frame of data. When the frame
+ was freed, the refcount would decrease. Once the refcount reached
+ 0, the filestream was freed, and as part of the operation, the
+ open files were closed as well. Thus it becomes more clear why a
+ missing ast_frfree would cause a reference leak and cause the
+ files to not be closed. You may ask then if there was a frame
+ leak before this patch. The answer to that is actually no! The
+ filestream code was "smart" enough to know that since the frame
+ we received came from a format interpreter, the frame had no
+ malloced data and thus didn't need to be freed. Now, however,
+ there is cleanup that needs to be done when we finish with the
+ frame, so we do need to call ast_frfree on the frame to be sure
+ that the refcount for the filestream is decremented
+ appropriately. (closes issue #14384) Reported by: fiddur Patches:
+ 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur,
+ putnopvut
+
+2009-02-04 00:43 +0000 [r173311] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c, pbx/pbx_config.c: Ensure that commas placed in the
+ middle of extension character classes do not interfere with
+ correct parsing of the extension. Also, if an unterminated
+ character class DOES make its way into the pbx core (through some
+ other method), ensure that it does not crash Asterisk. (closes
+ issue #14362) Reported by: Nick_Lewis Patches:
+ 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76
+
+2009-02-03 17:35 +0000 [r173169] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Broke up the large conditional blocks so
+ it is easy to see if a function is compiled.
+
+2009-02-03 00:29 +0000 [r173104-173130] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/xml.c, include/asterisk/compiler.h, apps/app_stack.c,
+ include/asterisk/optional_api.h: 1. Make OS X compile cleanly
+ with app_stack. 2. Use curl to download sound files, as curl is
+ installed natively on OS X, whereas wget and fetch are not.
+ (closes issue #14332) Reported by: oej Tested by: Corydon76
+
+ * /, configs/extensions.conf.sample: Merged revisions 173070 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009)
+ | 5 lines Add warning to standard config, that globals may be
+ overridden by other dialplan configuration files. (closes issue
+ #14388) Reported by: macli ........
+
+2009-02-02 23:57 +0000 [r173067] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 173066 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009)
+ | 2 lines Fix a feature inheritance bug I added after code review
+ ........
+
+2009-02-02 23:21 +0000 [r173028-173047] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, CHANGES: Reverting commit number 173028 as there
+ are some potential issues
+
+ * main/manager.c, CHANGES: Add a CLI command to log out a manager
+ user (closes issue #13877) Reported by: eliel Patches:
+ cli_manager_logout.patch.txt uploaded by eliel (license 64)
+ Tested by: eliel, putnopvut
+
+2009-02-02 20:40 +0000 [r172963] Richard Mudgett <rmudgett@digium.com>
+
+ * /: Recorded merge of revisions 172962 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009)
+ | 11 lines channels/chan_dahdi.c * Added doxygen comments to the
+ major dahdi structures. * Fixed PRI using an incorrect string
+ value if the extension delimiter is not present in the Dial()
+ function. * Fixed some uninitialized string variables on FXS
+ ports. configs/chan_dahdi.conf.sample * Updated some
+ documentation. These changes are already in trunk -r172400
+ ........
+
+2009-02-02 19:02 +0000 [r172929] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/features.c, CHANGES,
+ include/asterisk/features.h: This reverts the changes I made for
+ 11583; will reviewboard this before committing again... reopened
+ 11583 until all Russell's issues are resolved.
+
+2009-02-02 18:13 +0000 [r172894] Leif Madsen <lmadsen@digium.com>
+
+ * configs/res_ldap.conf.sample: Update the res_ldap.conf file with
+ a better working example. (closes issue #13861) Reported by:
+ scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by
+ blitzrage (license 10) Tested by: jcovert
+
+2009-02-02 17:37 +0000 [r172890] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/features.c, CHANGES,
+ include/asterisk/features.h: This change allows the disconnect
+ feature (as in "one-touch" in features.c) to be used within the
+ dial app, before a call is bridged. Many thanks to sobomax for
+ submitting this patch. Quoting from bug 11582: "So the goal of
+ the patch was to use the user configured feature code during the
+ call setup phase. The original ast_feature_interpret() function
+ is not well suited for this purpose as it uses much call bridge
+ specific data and doesn't separate a detection of feature from a
+ feature handler call. So a new function ast_feature_detect() has
+ been extracted off the ast_feature_interpret() function but
+ keeping the original logic intact except some insignificant
+ changes to locking. "Having created the ast_feature_detect()
+ function the possibility to use feature detection in almost any
+ place of the asterisk code. So a call to this function has been
+ added to wait_for_answer() function of app_dial.so module. This
+ code doesn't call the feature handler however and uses old call
+ leg disconnect logic to make the changes as small and simple as
+ possible to prevent unexpected problems. A disconnect feature
+ currently is the only one supported during call setup as other
+ features as call parking and call transfer don't make much sense
+ during call setup. However if need in some of the features would
+ arise it is much easier to implement as the infrastructure
+ changes are already in place with this patch." I have cleaned up
+ the patch somewhat, and verified that the existing functionality
+ is not harmed, and that the new functionality works. Terry has
+ committed his stuff, and there were no conflicts (see 14274).
+ (closes issue #11583) Reported by: sobomax Patches:
+ patch-apps__app_dial.c uploaded by sobomax (license 359)
+ patch-include__asterisk__features.h uploaded by sobomax (license
+ 359) patch-res__res_features.c uploaded by sobomax (license 359)
+ enable-features-during-call-setup.diff uploaded by sobomax
+ (license 359) 11583.newdiff uploaded by murf (license 17)
+ enable-features-during-call-setup-1.diff uploaded by sobomax
+ (license 359) 11583.latest-patch uploaded by murf (license 17)
+ Tested by: sobomax, murf
+
+2009-02-02 16:42 +0000 [r172855] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix a spelling mistake.
+
+2009-02-02 10:46 +0000 [r172816-172818] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Add a todo. I do need to really check what's
+ going on with this kill-the-user business ;-) Why do we suddenly
+ have two flags to set peer type?
+
+ * channels/chan_sip.c: Small formatting change
+
+ * channels/chan_sip.c: Add some well-needed improvements to the
+ wishlist in the code, so that we can close some bug reports.
+
+2009-02-02 01:41 +0000 [r172778] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_sip.c: The CID lookup feature wasn't actually
+ working properly with dialog-info+xml supporting devices. The
+ devices (snoms, specifically) need to receive a SIP URI instead
+ of just an extension. This adds that functionality.
+
+2009-02-01 02:44 +0000 [r172706-172741] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Blank argument crashes Asterisk (closes
+ issue #14377) Reported by: amorsen
+
+ * funcs/func_strings.c: Don't increment the loop, now that
+ incrementing is taken care of by the decoder function. (closes
+ issue #14363) Reported by: andrew53 Patches:
+ func_strings_filter.patch uploaded by andrew53 (license 519)
+
+2009-01-30 22:22 +0000 [r172598] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/channel.h: Fix redefinition of flag in channel.h
+
+2009-01-30 21:50 +0000 [r172580-172581] Terry Wilson <twilson@digium.com>
+
+ * configs/features.conf.sample: Remove incorrect line from sample
+ config
+
+ * apps/app_dial.c, main/global_datastores.c, main/features.c,
+ include/asterisk/global_datastores.h, CHANGES,
+ configs/features.conf.sample: Merged revisions 172517 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009)
+ | 37 lines Fix feature inheritance with builtin features When
+ using builtin features like parking and transfers, the
+ AST_FEATURE_* flags would not be set correctly for all instances
+ when either performing a builtin attended transfer, or parking a
+ call and getting the timeout callback. Also, there was no way on
+ a per-call basis to specify what features someone should have on
+ picking up a parked call (since that doesn't involve the Dial()
+ command). There was a global option for setting whether or not
+ all users who pickup a parked call should have
+ AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
+ PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
+ variable which can be set either in the dialplan or with setvar
+ in channels that support it. This variable can be set to any
+ combination of 't', 'k', 'w', and 'h' (case insensitive matching
+ of the equivalent dial options), to set what features should be
+ activated on this channel. The patch moves the setting of the
+ features datastores into the bridging code instead of app_dial to
+ help facilitate this. 2) adds global options parkedcallparking,
+ parkedcallhangup, and parkedcallrecording to be similar to the
+ parkedcalltransfers option for globally setting features. 3) has
+ builtin_atxfer call builtin_parkcall if being transfered to the
+ parking extension since tracking everything through multiple
+ masquerades, etc. is difficult and error-prone 4) attempts to fix
+ all cases of return calls from parking and completed builtin
+ transfers not having the correct permissions (closes issue
+ #14274) Reported by: aragon Patches:
+ fix_feature_inheritence.diff.txt uploaded by otherwiseguy
+ (license 396) Tested by: aragon, otherwiseguy Review
+ http://reviewboard.digium.com/r/138/ ........
+
+2009-01-30 18:36 +0000 [r172441-172548] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_aes.c: Parameter position reversed in documentation
+
+ * /, autoconf/ast_func_fork.m4, configure, main/app.c,
+ apps/app_rpt.c, main/asterisk.c: Merged revisions 172438 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009)
+ | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at
+ startup. Otherwise, if Asterisk runs as a non-root user and the
+ administrator does a 'restart now', Asterisk loses the ability to
+ set QOS on packets. (closes issue #14004) Reported by: nemo
+ Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
+ (license 14) Tested by: Corydon76 ........
+
+2009-01-29 23:15 +0000 [r172370-172440] Richard Mudgett <rmudgett@digium.com>
+
+ * main/cli.c: Remove tabs from comment
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
+ channels/chan_dahdi.c * Added doxygen comments to the major dahdi
+ structures. * Fixed PRI and SS7 using an incorrect string value
+ if the extension delimiter is not present in the Dial() function.
+ * Fixed SS7 not checking if the dialed extension is at least as
+ long as the stripmsd option. * Fixed PRI not handling unknown
+ TON/NPI prefix letters correctly. * Fixed some uninitialized
+ string variables on FXS ports. configs/chan_dahdi.conf.sample *
+ Updated some documentation.
+
+ * include/asterisk/say.h: Fixed some doxygen comments
+
+2009-01-29 17:10 +0000 [r172318-172319] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_local.c: Revert two lines that was extra, but only
+ on fridays.
+
+ * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
+ include/asterisk/causes.h, apps/app_queue.c: Fix "cancel answered
+ elsewhere" through app_queue with members in chan_local. Also,
+ implement a private cause code (as suggested by Tilghman). This
+ works with chan_sip, but doesn't propagate through chan_local.
+
+2009-01-29 16:48 +0000 [r172315] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/func_odbc.conf.sample: Better document mode=multirow,
+ based upon a conversation with Jared.
+
+2009-01-29 13:47 +0000 [r172271] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/realtime_pgsql.sql: The realtime_pgsql.sql script
+ is missing a couple of fields. closes issue #14339) Reported by:
+ fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur
+ (license 678)
+
+2009-01-29 13:24 +0000 [r172173-172270] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample, CHANGES: Update documentation
+
+ * include/asterisk/app.h, channels/chan_sip.c, main/app.c: - Make
+ sure we set setvar= variables on outbound calls too, not only
+ inbound calls. - Also, change a function in app.c to return a
+ userful value instead of always returning 0. Patch by fnordian,
+ changed by Corydon76 and myself. This does not close the bug
+ report, as fnordian had an additional change we're still
+ discussing. (related to issue #14059) Reported by: fnordian
+ Patches: chan_sip_hfield.patch uploaded by fnordian (license 110)
+ 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: fnordian, Corydon76, oej
+
+ * channels/chan_sip.c: Make sure register= line supports both port
+ and expiry at the same time. (closes issue #14185) Reported by:
+ Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by
+ Nick (license 657) Tested by: Nick_Lewis
+
+ * /, channels/chan_sip.c: Merged revisions 172169 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16
+ lines Make sure that we always add the hangupcause headers. In
+ some cases, the owner was disconnected before we checked for the
+ cause. This patch implements a temporary storage in the pvt and
+ use that instead. The code is based on ideas from code from
+ Adomjan in issue #13385 (Add support for Reason: header) Thanks
+ to Klaus Darillion for testing! (closes issue #14294) related to
+ issue #13385 Reported by: klaus3000 and adomjan Patches:
+ bug14294b.diff uploaded by oej (license 306) Based on
+ 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan
+ (license 487) Tested by: oej, klaus3000 ........
+
+2009-01-28 22:52 +0000 [r172132] Steve Murphy <murf@digium.com>
+
+ * channels/chan_misdn.c: A further correction: cast the sizeof to
+ an int.
+
+2009-01-28 22:48 +0000 [r172131] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_odbc.c: Fix how we skip fields (to avoid fields
+ which don't exist) when doing an UPDATE. (closes issue #14205)
+ Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: blitzrage
+
+2009-01-28 21:48 +0000 [r172063-172099] Steve Murphy <murf@digium.com>
+
+ * channels/chan_misdn.c: my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2
+ didn't like the \%ld and issued a warning, breaking my dev-mode
+ build. This fixes it.
+
+ * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /,
+ main/features.c, include/asterisk/channel.h: Merged revisions
+ 172030 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) |
+ 46 lines This patch fixes h-exten running misbehavior in
+ manager-redirected situations. What it does: 1. A new Flag value
+ is defined in include/asterisk/channel.h,
+ AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
+ bridge hangup exten code not to run the h-exten there (nor
+ publish the bridge cdr there). It will done at the pbx-loop level
+ instead. 2. In the manager Redirect code, I set this flag on the
+ channel if the channel has a non-null pbx pointer. I did the same
+ for the second (chan2) channel, which gets run if name2 is set...
+ and the first succeeds. 3. I restored the ending of the cdr for
+ the pbx loop h-exten running code. Don't know why it was removed
+ in the first place. 4. The first attempt at the fix for this bug
+ was to place code directly in the async_goto routine, which was
+ called from a large number of places, and could affect a large
+ number of cases, so I tested that fix against a fair number of
+ transfer scenarios, both with and without the patch. In the
+ process, I saw that putting the fix in async_goto seemed not to
+ affect any of the blind or attended scenarios, but still, I was
+ was highly concerned that some other scenarios I had not tested
+ might be negatively impacted, so I refined the patch to its
+ current scope, and jmls tested both. In the process, tho, I saw
+ that blind xfers in one situation, when the one-touch blind-xfer
+ feature is used by the peer, we got strange h-exten behavior. So,
+ I inserted code to swap CDRs and to set the HANGUP_DONT field, to
+ get uniform behavior. 5. I added code to the bridge to obey the
+ HANGUP_DONT flag, skipping both publishing the bridge CDR, and
+ running the h-exten; they will be done at the pbx-loop (higher)
+ level instead. 6. I removed all the debug logs from the patch
+ before committing. 7. I moved the AUTOLOOP set/reset in the
+ h-exten code in res_features so it's only done if the h-exten is
+ going to be run. A very minor performance improvement, but
+ technically correct. (closes issue #14241) Reported by: jmls
+ Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer
+ uploaded by murf (license 17) Tested by: murf, jmls ........
+
+2009-01-28 17:27 +0000 [r171964] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 171963 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28
+ Jan 2009) | 2 lines Clarify log message (suggested by manxpower
+ on #asterisk-dev) ........
+
+2009-01-28 14:39 +0000 [r171838-171925] Olle Johansson <oej@edvina.net>
+
+ * CHANGES: Yep. Documentation is important.
+
+ * apps/app_queue.c: Add final part of previously committed work for
+ answered elsewhere in queue - the missing piece that started with
+ app_dial() earlier on. This is to avoid having the list and
+ counter of missed calls being touched by queue calls. Add the C
+ option to queue() and nothing will be logged on phones that
+ support the Reason: header on SIP cancel, like the SNOM phones.
+
+ * configs/sip.conf.sample: Add some more notes about device
+ matching.
+
+ * /, configs/sip.conf.sample: Merged revisions 171837 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan
+ 2009) | 2 lines Add a better explanation of the difference
+ between the device namespace and the dialplan for newbies.
+ ........
+
+2009-01-28 00:17 +0000 [r171797] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_aes.c: Fix some signedness problems in func_aes.c
+
+2009-01-27 23:28 +0000 [r171793] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: Don't complain about lack of D-channels on
+ PTMP connections
+
+2009-01-27 22:43 +0000 [r171757] David Vossel <dvossel@digium.com>
+
+ * funcs/func_aes.c (added), CHANGES: Adding AES_ENCRYPT and
+ AES_DECRYPT dialplan functions. (closes issue #14301) Reported
+ by: amorsen review: http://reviewboard.digium.com/r/128/
+
+2009-01-27 21:58 +0000 [r171618-171691] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Merged revisions 171689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan
+ 2009) | 39 lines Fix devicestate problems for "always-on" agent
+ channels A revision to chan_agent attempted to "inherit" the
+ device state of the underlying channel in order to report the
+ device state of an agent channel more accurately. The problem
+ with the logic here is that it makes no sense to use this for
+ always-on agents. If the agent is logged in, then to the
+ underlying channel, the agent will always appear to be "in use,"
+ no matter if the agent is on a call or not. The reason is that to
+ the underlying channel, the channel is currently in use on a call
+ to the AgentLogin application. The most common cause that I found
+ for this issue to occur was for a SIP channel to be the
+ underlying channel type for an Agent channel. If the SIP phone
+ re-registers, then the registration will cause the device state
+ core to query the device state of the SIP channel. Since the SIP
+ channel is in use, the Agent channel would also inherit this
+ status. Once the agent channel was set to "in use" there was no
+ way that the device state could change on that channel unless the
+ agent logged out. The solution for this problem is a bit
+ different in 1.4 than it is in the other branches. In 1.4, there
+ will be a one-line fix to make sure that only callback agents
+ will inherit device state from their underlying channel type. For
+ the other branches of Asterisk, since callback support has been
+ removed, there is also no need for device state inheritance in
+ chan_agent, so I will simply be removing it from the code. In
+ addition, the 1.4 source is getting a new comment to help the
+ next person who edits chan_agent.c. I'm adding a comment that a
+ agent_pvt's loginchan field may be used to determine if the agent
+ is a callback agent or not. (closes issue #14173) Reported by:
+ nathan Patches: 14173.patch uploaded by putnopvut (license 60)
+ Tested by: nathan, aramirez ........
+
+ * /, main/slinfactory.c: Merged revisions 171621 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan
+ 2009) | 18 lines Prevent a crash from occurring when a jitter
+ buffer interpolated frame is removed from a slinfactory
+ slinfactory used the "samples" field of an ast_frame in order to
+ determine the amount of data contained within the frame. In
+ certain cases, such as jitter buffer interpolated frames, the
+ frame would have a non-zero value for "samples" but have NULL
+ "data" This caused a problem when a memcpy call in
+ ast_slinfactory_read would attempt to access invalid memory. The
+ solution in use here is to never feed frames into the slinfactory
+ if they have NULL "data" (closes issue #13116) Reported by:
+ aragon Patches: 13116.diff uploaded by putnopvut (license 60)
+ ........
+
+ * apps/app_queue.c: Fix queue crashes that would occur after the
+ calling channel was masqueraded. The data passed to the
+ end_bridge_callback was assumed to be data which was still
+ stack'd. The problem was that with some call features, attended
+ transfers in particular, a new bridge thread is started once the
+ feature completes, meaning that when the end_bridge_callback is
+ called, the end_bridge_callback_data was invalid. To fix this
+ problem, there are two measures taken 1. Instead of pointing to
+ stacked data, we now used heap-allocated data for passing to the
+ end_bridge_callback in app_queue 2. Since bridges can end
+ multiple times on a single logical call, we wait until the final
+ bridge is broken to actually set any queue variables. This is
+ accomplished through reference-counting and the use of an
+ end_bridge_callback_data_fixup function in app_queue.c (closes
+ issue #14260) Reported by: ccesario Patches: 14260.patch uploaded
+ by putnopvut (license 60) Tested by: ccesario
+
+2009-01-27 15:23 +0000 [r171558] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: Handle new VMWI ioctl structure (Now there
+ are two VMWI ioctl calls.) (issue #14104) Reported by: alecdavis
+ Tested by: dbailey
+
+2009-01-27 15:00 +0000 [r171263-171528] Olle Johansson <oej@edvina.net>
+
+ * /, channels/chan_sip.c: Solving the same issue, but a bit
+ different in trunk... Merged revisions 171527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13
+ lines Use the same branch tag in CANCEL as in INVITE Originally
+ putnopvut implemented some changes in revision 142079 that
+ according to the bug report seemed to have worked then, but
+ somehow fails now. I guess code, as humans, get old and forget
+ stuff. Anyway, this bug caused CANCEL not to work with picky
+ systems. Thanks Fredrik for pointing out where the bug in the SIP
+ messaging was. (closes issue #14346) Reported by: oej Patches:
+ bug14346.diff uploaded by oej (license 306) Tested by: oej
+ ........
+
+ * channels/chan_sip.c: Moving generic setting to friends
+
+ * channels/chan_sip.c: Continue to move variables into the sip_cfg
+ structure to make them easier to handle in the future as a group
+ of settings for a group of devices. At some point, I want one
+ sip_cfg per domain handled, so we can have "group" settings.
+
+ * channels/chan_sip.c: Just moving around variable declarations so
+ that we have all globals in the same place. Default setting is
+ set before we activate the channel or at reloads, not where we
+ declare the variable.
+
+ * /, channels/chan_sip.c: Merged revisions 171264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9
+ lines Don't retransmit 401 on REGISTER requests when
+ alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000
+ Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000 ........
+
+ * main/channel.c: Add extensions and context on manager event when
+ new channel is created.
+
+2009-01-25 23:58 +0000 [r171188] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_oss.c: Merged revisions 171187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009)
+ | 6 lines Correctly track the hookstate (closes issue #13686)
+ Reported by: itiliti Patches: 20081013__bug13686.diff.txt
+ uploaded by Corydon76 (license 14) ........
+
+2009-01-25 16:50 +0000 [r171043-171081] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: dont segfault when a MWI event occurs on
+ a line without a registered device
+
+ * configs/skinny.conf.sample: Make the sample skinny.conf work
+ (closes issue #14325) Reported by: DEA Patches:
+ skinny.conf.sample-trunk.txt uploaded by DEA (license 3)
+
+2009-01-25 13:35 +0000 [r170980] Sean Bright <sean.bright@gmail.com>
+
+ * /, apps/app_page.c: Merged revisions 170979 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan
+ 2009) | 9 lines Resolve a logic error that was causing Page() to
+ crash when more than one channel was specified. (closes issue
+ #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
+ uploaded by seanbright (license 71) Tested by: kc0bvu ........
+
+2009-01-25 02:49 +0000 [r170902-170943] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/utils.h: Change ARRAY_LEN() to be more C++ safe.
+ When the second part of this macro is written as 0[a] instead of
+ a[0], it will force a failure if the macro is used on a C++
+ object that overloads the [] operator.
+
+ * res/res_agi.c: Add a todo to finish the XML docs in this module
+
+2009-01-24 13:55 +0000 [r170837] Tilghman Lesher <tlesher@digium.com>
+
+ * /, configs/res_odbc.conf.sample: Merged revisions 170836 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009)
+ | 2 lines Remove superfluous implementation note (closes issue
+ #14319) ........
+
+2009-01-23 23:10 +0000 [r170794] Richard Mudgett <rmudgett@digium.com>
+
+ * doc/tex/Makefile: Fix asterisk.pdf generation if branch name has
+ an underscore in it.
+
+2009-01-23 22:58 +0000 [r170790] Russell Bryant <russell@digium.com>
+
+ * doc/tex/Makefile: Don't blow up if a branch name has an
+ underscore in it
+
+2009-01-23 20:56 +0000 [r170677-170720] Mark Michelson <mmichelson@digium.com>
+
+ * /, configs/res_odbc.conf.sample: Merged revisions 170719 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan
+ 2009) | 8 lines Add notes to the idlecheck explanation in
+ res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000
+ Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by
+ klaus3000 (license 65) ........
+
+ * /, contrib/i18n.testsuite.conf: Merged revisions 170671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan
+ 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use
+ deprecated syntax * Convert Wait,1 to Wait(1) * Convert
+ SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
+ priorities beyond the first Also added test for Chinese numbers,
+ too. (closes issue #14320) Reported by: dant Patches:
+ i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
+ 670) ........
+
+2009-01-23 20:18 +0000 [r170652] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c, /: Merged revisions 170648 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4
+ lines When a channel is answered make sure any indications
+ currently playing stop. Usually the phone would do this but if
+ the channel was already answered then they are being generated by
+ Asterisk and we darn well need to stop them. (closes issue
+ #14249) Reported by: RadicAlish ........
+
+2009-01-23 19:25 +0000 [r170608] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 170588 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23
+ Jan 2009) | 2 lines Additions to AST-2009-001 ........
+
+2009-01-23 19:09 +0000 [r170505-170569] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 170568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4
+ lines When a call is forwarded stop any active indications. The
+ new channel will provide an indication, if need be, itself.
+ (closes issue #14310) Reported by: RadicAlish ........
+
+ * /, channels/chan_sip.c: Merged revisions 170504 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4
+ lines Use the on hold flag to see if the call is on hold or not.
+ It is possible that our address for them will still be valid even
+ though they are on hold. (closes issue #14295) Reported by:
+ klaus3000 ........
+
+2009-01-23 17:46 +0000 [r170501] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_h323.c: let's use SENTINEL where needed
+
+2009-01-23 17:32 +0000 [r170498] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Reset the ast_str used for escape
+ substitution. We need to do this since it is a thread local
+ variable that may contain the value of a previous substitution.
+ (closes issue #14312) Reported by: pj
+
+2009-01-23 17:03 +0000 [r170463] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: We should not do restart messages if we're
+ in PTMP mode
+
+2009-01-23 16:57 +0000 [r170460] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Dont clear the display of skinny phones
+ when not needed. (closes issue #13182) Reported by: pj Patches:
+ 2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak
+ (license 7) Tested by: mvanbaak, pj
+
+2009-01-23 16:35 +0000 [r170457] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: MWI messages included in CID spill was not
+ being properly handled and prevented the call from being
+ processed (issue #14313) Reported by: seandarcy Tested by:
+ dbailey
+
+2009-01-23 15:44 +0000 [r170393] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 170392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan
+ 2009) | 28 lines Fix broken call pickup There was a subtle change
+ in ast_do_masquerade which resulted in failed attempts to pickup
+ calls. The problem was that the value of the AST_FLAG_OUTGOING
+ flag was copied from the clone to the original channel. In the
+ case of call pickup, this meant that the AST_FLAG_OUTGOING flag
+ ended up being cleared on the channel that was attempting to
+ execute the pickup. Because this flag was not set, when ast_read
+ came across an answer frame, it ignored it. The result of this
+ was that the calling channel was never properly answered. This
+ fix changes the behavior in ast_do_masquerade to set the flags on
+ the original channel to the union of the flags on the clone
+ channel. This way, if the AST_FLAG_OUTGOING flag is set on either
+ of the two channels involved in the masquerade, the resulting
+ channel will have the flag set as well. (closes issue #14206)
+ Reported by: francesco_r Patches: 14206.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut
+ ........
+
+2009-01-22 23:23 +0000 [r170351] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c: Make sure we don't set the channel to be
+ inalarm for a D-channel drop on PTMP connections
+
+2009-01-22 21:25 +0000 [r170307] Tilghman Lesher <tlesher@digium.com>
+
+ * main/abstract_jb.c: Create logfile safely. (closes issue #14160)
+ Reported by: tzafrir Patches: 20090104__bug14160.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2009-01-22 20:04 +0000 [r170240] Joshua Colp <jcolp@digium.com>
+
+ * /, main/rtp.c: Merged revisions 170239 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7
+ lines Don't crash if RTCP is not enabled on an RTP structure but
+ statistics are output. (closes issue #14234) Reported by: jcovert
+ Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
+ rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........
+
+2009-01-22 17:19 +0000 [r170165] Tilghman Lesher <tlesher@digium.com>
+
+ * /, pbx/pbx_config.c: Merged revisions 170158 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009)
+ | 6 lines Allow global variables after substitution to be as long
+ as other variables. (closes issue #14263) Reported by: markd
+ Patches: 20090120__bug14263.diff.txt uploaded by Corydon76
+ (license 14) ........
+
+2009-01-22 16:52 +0000 [r170148] Joshua Colp <jcolp@digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 170147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4
+ lines If we are unable to request a DAHDI pseudo channel and we
+ are using the user introduction without review option make sure
+ it gets unset so other code does not blindly assume a DAHDI
+ pseudo channel exists. (closes issue #14282) Reported by:
+ cheesegrits ........
+
+2009-01-22 15:49 +0000 [r170112] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: change VMWI to
+ use new DAHDI_VMWI ioctl call. Change configure script to detect
+ the new ioctl call data structure. (issue #14104) Reported by:
+ alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded
+ by dbailey (license ) Tested by: alecdavis, dbailey
+
+2009-01-22 15:14 +0000 [r170047-170051] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, /: Merged revisions 170050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6
+ lines Do a string comparison instead of pointer comparison since
+ some people specify the context they are actually in as an
+ argument to get around some funkiness. (closes issue #14011)
+ Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga
+ (license 665) ........
+
+ * apps/app_parkandannounce.c: Clear the autoloop flag when parsing
+ and setting the context/extension/priority to go back to. When
+ the channel executes a PBX again we want it to start out at the
+ point we explicitly say and at that point it will not yet be
+ doing autoloop. (closes issue #14304) Reported by: jcovert
+
+2009-01-22 02:10 +0000 [r170007] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: * Adjust some conditionals to balance
+ curly braces. * Other minor changes.
+
+2009-01-22 00:44 +0000 [r169944] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 169943 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009)
+ | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really
+ wanted to ask is whether autoconf detected a static initializer
+ value. This fixes rwlocks on all such platforms (mainly, Mac OS
+ X). (closes issue #13767) Reported by: jcovert Patches:
+ 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: jcovert, Corydon76 ........
+
+2009-01-22 00:23 +0000 [r169910] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_dahdi.c: Whitespace changes only
+
+2009-01-21 23:25 +0000 [r169869] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, /: Merged revisions 169867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4
+ lines Read lock the contexts to maintain the locking order when
+ we are notified that the state of a device has changed. (closes
+ issue #13839) Reported by: mcallist ........
+
+2009-01-21 23:20 +0000 [r169794-169866] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c: Test commit for test issue #14303
+
+ * main/say.c: Fix a crash when saying certain numbers in Chinese
+ This commit fixes a crash that was occurring when attempting to
+ say a number between 10000 and 100000 due to dividing by 0. This
+ also removes some places where a "zero" is spoken when it should
+ not be. (closes issue #14291) Reported by: dant Patches:
+ say.c-14291.diff uploaded by dant (license 670) Tested by: dant
+
+2009-01-21 22:04 +0000 [r169793] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/tex/extensions.tex: remove duplicated sentence.
+
+2009-01-21 21:53 +0000 [r169791] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Further fix some oddities in sip show users
+ and sip show peers logic ccesario on IRC pointed out that his sip
+ peers were not displayed properly when he would issue the command
+ "sip show peers." The problem was that the onlymatchonip field
+ was used to determine if the endpoint was a "peer" or "user." The
+ tricky part is that a "friend" is supposed to be treated as both
+ a "user" and a "peer" but the logic would not allow "friends" to
+ show up as "peers" since onlymatchonip was set to FALSE for
+ friends. I have modified the sip_peer structure to more
+ explicitly keep track of what type endpoint it is so that the
+ various manager and CLI commands will display the expected
+ information Reported by ccesario via IRC Tested by ccesario
+
+2009-01-21 21:03 +0000 [r169723] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/asterisk.c: Merged revisions 169722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009)
+ | 8 lines Extra NULLs in the output cause some terminal types to
+ abort in the middle of a color code, causing terminal weirdness.
+ (closes issue #14130) Reported by: coolmig Patches:
+ 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, coolmig ........
+
+2009-01-21 17:21 +0000 [r169673] Steve Murphy <murf@digium.com>
+
+ * utils/refcounter.c: This patch corrects a segfault reported in
+ 14289, due to a null ptr being refd. Yes, seanbright is right in
+ the bug comments, that is the fix. Sorry for this oversight; I
+ guess my personal usage didn't have this happen! murf (closes
+ issue #14289) Reported by: jamesgolovich
+
+2009-01-21 10:49 +0000 [r169620-169625] Russell Bryant <russell@digium.com>
+
+ * /: Remove properties that erroneously got merged into trunk
+
+ * main/tcptls.c: Fix a regression in TCP support. This patch fixes
+ a problem that caused chan_sip to think that every open TCP
+ session was to a remote address of 0.0.0.0:0. (closes issue
+ #14287) Reported by: jamesgolovich Patches: bug-14287.diff.txt
+ uploaded by jamesgolovich (license 176)
+
+2009-01-21 00:33 +0000 [r169557-169611] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix device state parsing issues for channel
+ names with multiple slashes The fix being applied is a bit
+ different for trunk and the 1.6.X branches. For trunk, we only
+ wish to strip off the characters beyond the second slash if the
+ channel is a Local channel (i.e. we are removing the /n from the
+ device name). Other channel technologies with multiple slashes
+ (e.g. DAHDI) need the information after the second slash in order
+ to get the proper device state information. In addition to this
+ fix, the 1.6.X branches are receiving a much more important fix
+ as well. The problem in 1.6.X is that the member's device name
+ was being directly changed instead of having a copy changed. This
+ meant that we would strip off the second slash and trailing
+ characters and then leave the member's device name like that
+ permanently thereafter. (closes issue #14014) Reported by:
+ kebl0155 Patches: 14014_number2.patch uploaded by putnopvut
+ (license 60) Tested by: kebl0155
+
+ * apps/app_queue.c: Use the default timeout for a queue instead of
+ -1 (closes issue #14272) Reported by: timking
+
+ * /, channels/chan_sip.c: Convert the character pointers in a
+ sip_request to be pointer offsets When an ast_str expands to hold
+ more data, any pointers that were pointing to the data prior to
+ the expansion will be pointing at invalid memory. This change
+ makes such pointers used in chan_sip.c instead be offsets from
+ the beginning of the string so that the same math may be applied
+ no matter where in memory the string resides. To help ease this
+ transition, a macro called REQ_OFFSET_TO_STR has been added to
+ chan_sip.c so that given a sip_request and an offset, the string
+ at that offset is returned. (closes issue #14220) Reported by:
+ riksta Tested by: putnopvut Review
+ http://reviewboard.digium.com/r/126/
+
+2009-01-20 19:22 +0000 [r169486-169510] Terry Wilson <twilson@digium.com>
+
+ * main/features.c: Make a proper builtin attended transfer to
+ parking work This is an ugly hack from 1.4 that allows the
+ timeout callback from a parked call to use the right channel name
+ for the callback when the park is done with a builtin attended
+ transfer (that isn't completed early). This hasn't ever worked in
+ trunk and no one has complained yet, so eh.
+
+ * /, main/features.c: Merged revisions 169485 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009)
+ | 6 lines Don't play audio to the channel if we've masqueraded
+ (closes issue #14066) Reported by: bluefox Tested by:
+ otherwiseguy, bluefox ........
+
+2009-01-19 21:42 +0000 [r169438] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/res_odbc.h, funcs/func_odbc.c,
+ include/asterisk/strings.h, res/res_odbc.c: ast_str_SQLGetData is
+ *not* part of the ast_str API, it's part of the ast_odbc API and
+ just happens to use an ast_str as the buffer; move all of it to
+ res_odbc.c and res_odbc.h, renaming appropriately along the way
+ fix some minor coding style issues in strings.h and add some
+ attribute_pure annotations to functions in the ast_str API
+
+2009-01-19 20:14 +0000 [r169367-169369] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: fix assignment in swapmode plug. Spotted and fix
+ provided by ys (closes issue #14129) Reported by: ys Tested by:
+ ys
+
+ * channels/chan_skinny.c: Redo the event-based MWI in chan_skinny.
+ Dan saw regular segfaults with the old implementation and rewrote
+ it to make it really eventbased. I altered it to be trunk
+ compatible and wedhorn gave some feedback and ideas how to make
+ it even better. (closes issue #13821) Reported by: DEA Patches:
+ chan_skinny-mwi-events.txt uploaded by DEA (license 3) Tested by:
+ mvanbaak, DEA "no probs by me" from wedhorn
+
+2009-01-19 20:05 +0000 [r169365] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, /, apps/app_userevent.c: Merged revisions 169364
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009)
+ | 4 lines Truncate userevents at the end of a line, when the
+ command exceeds the buffer. (closes issue #14278) Reported by:
+ fnordian ........
+
+2009-01-19 18:36 +0000 [r169327] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: Make asterisk compile on non-amd64 versions of
+ OpenBSD. The HW_PHYSMEM64 is only available in latest OpenBSD
+ and/or amd64 versions of OpenBSD. Use HW_PHYSMEM when
+ HW_PHYSMEM64 is not available. (closes issue #14129) Reported by:
+ ys Patches: 2009011600_physmem64.diff.txt uploaded by mvanbaak
+ (license 7) Tested by: mvanbaak, jtodd
+
+2009-01-19 18:22 +0000 [r169277-169325] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: Get rid of magic number and replace with
+ DAHDI_VMWI_NUMBER_MASK when determining the number of messages
+ pending for MWI call
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
+ enhanced MWI generation to take advantage of new dahdi line
+ reversal MWI ability. (closes issue #14104) Reported by:
+ alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey
+ (license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis, dbailey
+
+2009-01-19 15:54 +0000 [r169211] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 169210 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon,
+ 19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a
+ potential NULL pointer dereference Move the check for if both
+ channels on a local_pvt have generators to below where p->chan is
+ checked for NULLity (NULLness?). This prevents a crash from
+ occurring if p->chan is NULL. (closes issue #14189) Reported by:
+ sascha Patches: 14189.patch uploaded by putnopvut (license 60)
+ Tested by: sascha ........
+
+2009-01-17 18:26 +0000 [r169153] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
+ discriminator for when ring pulse alert signal is used to preface
+ MWI spills This prevents the situation when MWI messages are
+ added to caller ID spills causing the channel to be hung up
+
+2009-01-17 02:52 +0000 [r169116] Sean Bright <sean.bright@gmail.com>
+
+ * pbx/pbx_dundi.c: Change intializer types. Found while working on
+ asterisk-cpp. I have a new favorite error message from g++:
+ pbx_dundi.c:4580: sorry, unimplemented: non-trivial designated
+ initializers not supported I like it when compilers are
+ apologetic.
+
+2009-01-17 01:56 +0000 [r169044-169080] Terry Wilson <twilson@digium.com>
+
+ * main/tcptls.c, main/http.c, include/asterisk/tcptls.h: Fix
+ qualify for TCP peer (closes issue #14192) Reported by:
+ pabelanger Patches: asterisk-bug14192.diff.txt uploaded by
+ jamesgolovich (license 176) Tested by: jamesgolovich
+
+ * channels/chan_sip.c: Fix port :0 added to SIP INVITE URI when
+ outboundproxy used (closes issue #14233) Reported by: chris-mac
+ Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich
+ (license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy
+
+2009-01-16 22:43 +0000 [r168976] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168975 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan
+ 2009) | 18 lines Account for possible NULL pointer when we
+ receive a 408 in response to a REGISTER It may be that by the
+ time we receive a reply to a REGISTER request, the attempt has
+ timed out and thus the registry structure pointed to by the
+ corresponding sip_pvt has gone away. This situation was handled
+ properly for a 200 OK response, but the 408 case assumed that the
+ sip_registry struct was non-NULL, thus potentially causing a
+ crash This commit fixes this assumption and prints out a message
+ to the console if we should receive a late 408 response to a
+ REGISTER (closes issue #14211) Reported by: aborghi Patches:
+ 14211.diff uploaded by putnopvut (license 60) Tested by: aborghi
+ ........
+
+2009-01-16 22:16 +0000 [r168941] Terry Wilson <twilson@digium.com>
+
+ * /, main/features.c: Merged revisions 168716 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009)
+ | 12 lines Convert call to park_call_full to
+ masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
+ return value, we need to use masqueraded parking, otherwise we
+ will try to call ast_hangup() in __pbx_run() and in
+ do_parking_thread() and then promptly crash. (closes issue
+ #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
+ issue #14228) Reported by: kobaz Tested by: otherwiseguy ........
+
+2009-01-16 19:54 +0000 [r168898] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_timing_timerfd.c: Fix a logic error that occur when using
+ the timerfd interface This sequence of events posed a problem
+ timerfd_timer_open timerfd_timer_enable_continuous
+ timerfd_timer_set_rate timerfd_timer_disable_continuous The
+ reason was that the timing module was written under the
+ assumption that timerfd_timer_set_rate would not be called
+ between enabling and disabling continuous mode. What happened in
+ this situation was that timerfd_timer_enable_continuous saved off
+ our previously set timer (in this situation a 0 timer, meaning it
+ never runs out). Then timerfd_timer_disable_continuous would
+ restore this 0 timer, even though it logically should set the
+ timer to be whatever was set in timerfd_timer_set_rate. Now the
+ behavior in timerfd_timer_set_rate is to overwrite the saved
+ timer that may or may not have been set in
+ timerfd_timer_enable_continuous. Even if
+ timerfd_timer_enable_continuous has not been previously called,
+ this will not harm the operation. Thanks to Terry Wilson for
+ discovering the problem and giving me a really great debug
+ capture that pointed out the problem clearly
+
+2009-01-16 18:49 +0000 [r168832] Tilghman Lesher <tlesher@digium.com>
+
+ * /, main/say.c, include/asterisk/say.h, apps/app_voicemail.c:
+ Merged revisions 168828 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009)
+ | 6 lines Fix the conjugation of Russian and Ukrainian languages.
+ (related to issue #12475) Reported by: chappell Patches:
+ vm_multilang.patch uploaded by chappell (license 8) ........
+
+2009-01-16 17:09 +0000 [r168759-168760] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Fix a spelling mistake.
+
+ * channels/chan_misdn.c: build in dev mode
+
+2009-01-16 00:34 +0000 [r168737-168746] Steve Murphy <murf@digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 168745 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) |
+ 14 lines This patch fixes a problem where a goto (or jump, in
+ this case) fails a consistency check because it can't find a
+ matching extension. The problem was a missing instruction to end
+ the range notation in the code where it converts the pattern into
+ a regex and uses the regex code to determine the match. I tested
+ using the AEL code the user supplied, and now, the consistency
+ check passes. (closes issue #14141) Reported by: dimas ........
+
+ * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: This patch
+ allows null args in ast_expr2 func calls, and fixes commas being
+ converted to pipes, which was 1.4 type stuff. If the user says
+ count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); then it
+ won't complain about the empty arg (c,,...) and fabled's patch
+ won't let it swap the commas for pipes. Ran it thru my dialplan
+ and no complaints. (closes issue #14169) Reported by: fabled
+ Patches: function-argument-separator-fix.diff uploaded by fabled
+ (license 448)
+
+2009-01-15 20:18 +0000 [r168734] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_config_odbc.c, build_tools/menuselect-deps.in, configure,
+ funcs/func_odbc.c, configure.ac, cdr/cdr_adaptive_odbc.c,
+ cdr/cdr_odbc.c, makeopts.in, res/res_odbc.c,
+ apps/app_voicemail.c: remove the PBX_ODBC logic from the
+ configure script, and add GENERIC_ODCB logic that includes
+ copying the relevant LIB and INCLUDE data from either UnixODBC or
+ iODBC, based on which was found; if both were found, prefer
+ UnixODBC this stops modules from being linked against both sets
+ of libraries on systems that have both installed
+
+2009-01-15 20:00 +0000 [r168725-168732] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add missing brace
+
+ * channels/chan_sip.c: Fix the compactheaders option in sip.conf
+
+ * channels/chan_sip.c: Remove an unneeded condition for line
+ addition to a SIP request/response In Asterisk 1.4 and 1.6.0, the
+ sip_request structure had a statically allocated buffer to hold
+ the text of the request. There was a check in the add_line
+ function to not attempt to write the line into the buffer if we
+ did not have room for it. In trunk and Asterisk versions starting
+ with 1.6.1, an expandable ast_str structure is used to hold the
+ text. Since it may grow to fit an arbitrarily sized string, this
+ check in add_line is no longer valid. I found this oddity while
+ attempting to fix issue #14220; however, I do not believe that
+ this is the fix for that issue since the output supplied by the
+ reporter did not contain the warning message that would be
+ printed had this condition been satisfied.
+
+2009-01-15 18:47 +0000 [r168722] Olle Johansson <oej@edvina.net>
+
+ * /, configs/extconfig.conf.sample: Merged revisions 168721 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2
+ lines Meetme actually has realtime but wasn't documented ........
+
+2009-01-15 18:39 +0000 [r168719] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/strings.h: Resolve issue with negative vs
+ non-negative length parameters. (closes issue #14245) Reported
+ by: dveiga
+
+2009-01-15 18:08 +0000 [r168711-168712] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Make sure that we have the same terminology
+ in sip.conf.sample and the source code warning. Thanks Nick Lewis
+ for pointing this out in the bug tracker.
+
+ * configs/sip.conf.sample: Clarify some misunderstandings and make
+ it even more clear that you can refer to a peer in the register=
+ line.
+
+2009-01-15 15:33 +0000 [r168705] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_meetme.c: Add a missing unlock and properly handle the
+ 'maxusers' setting on MeetMe conferences. We were using the 'user
+ number' field to compare against the maximum allowed users, which
+ works assuming users with lower user numbers didn't leave the
+ conference. (closes issue #14117) Reported by: sergedevorop
+ Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright
+ (license 71) Tested by: sergedevorop
+
+2009-01-15 13:37 +0000 [r168636-168639] Olle Johansson <oej@edvina.net>
+
+ * CREDITS, CHANGES: Related to issue #14246 Update changes for
+ SIPRemoveHeader()
+
+ * channels/chan_sip.c: Add capability to remove added SIP headers
+ *before* INVITE is generated. (closes issue #14246) Reported by:
+ klaus3000 Patches: 2patch_chan_sip_SIPRemoveHeader_trunk.txt
+ uploaded by klaus3000 (license 65)
+
+ * apps/app_queue.c: Add support for setting the Reason header when
+ cancelling a call in the queue because someone else answered.
+ Previously, only dial() was supported. EDV-102
+
+2009-01-15 00:14 +0000 [r168629] Mark Michelson <mmichelson@digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 168628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan
+ 2009) | 16 lines Fix some crashes from bad datastore handling in
+ app_queue.c * The queue_transfer_fixup function was searching for
+ and removing the datastore from the incorrect channel, so this
+ was fixed. * Most datastore operations regarding the
+ queue_transfer datastore were being done without the channel
+ locked, so proper channel locking was added, too. (closes issue
+ #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
+ putnopvut (license 60) Tested by: ZX81, festr ........
+
+2009-01-14 23:10 +0000 [r168626] Sean Bright <sean.bright@gmail.com>
+
+ * main/cli.c: Don't crash when typing 'core set verbose' or 'core
+ set debug' by themselves. (closes issue #14219) Reported by:
+ jamesgolovich Patches: asterisk-setverbosecrash.diff.txt uploaded
+ by jamesgolovich (license 176)
+
+2009-01-14 21:51 +0000 [r168623] Richard Mudgett <rmudgett@digium.com>
+
+ * /, channels/misdn/isdn_lib.c: Merged revisions 168622 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009)
+ | 4 lines * Fixed create_process() allocation of process ID
+ values. The allocated process IDs could overflow their respective
+ NT and TE fields. Affects outgoing calls. ........
+
+2009-01-14 21:19 +0000 [r168619] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: This fixes a problem where MWI FSK spills
+ were being injected onto off hook fxs lines. (closes issue
+ #14143) Reported by: alecdavis Patches:
+ chan_dahdi-14143.patch.txt uploaded by dbailey (license ) Tested
+ by: alecdavis
+
+2009-01-14 20:58 +0000 [r168615] Sean Bright <sean.bright@gmail.com>
+
+ * /, contrib/scripts/autosupport: Merged revisions 168614 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan
+ 2009) | 9 lines Update autosupport script to supply info for both
+ Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x
+ and trunk instead of zttest. (closes issue #14132) Reported by:
+ dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by
+ dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded
+ by dsedivec (license 638) ........
+
+2009-01-14 20:51 +0000 [r168613] Steve Murphy <murf@digium.com>
+
+ * /, apps/app_page.c: Merged revisions 168608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1
+ line app_page was failing to compile in dev-mode on my gcc-4.2.4
+ system. This change gets rid of the warning. ........
+
+2009-01-14 20:13 +0000 [r168610] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Restore the "sip show users" and "sip show
+ user" CLI commands (closes issue #14180) Reported by: amorsen
+ Patches: sip_show_users_161v3.diff uploaded by putnopvut (license
+ 60) Tested by: blitzrage, amorsen
+
+2009-01-14 19:36 +0000 [r168609] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/asterisk.c: Fix compilation on FreeBSD and OSX This started
+ as work to fix the 'core show sysinfo' CLI command but while
+ working on it oej pointed out that read_credentials did not
+ compile neither. So while being there, fix that as well. Thanks
+ for all the testing oej! (closes issue #14129) Reported by: ys
+ Tested by: oej, mvanbaak
+
+2009-01-14 19:11 +0000 [r168601-168604] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c, /: Merged revisions 168603 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009)
+ | 7 lines Don't read into a buffer without first checking if a
+ value is beyond the end. (closes issue #13600) Reported by: atis
+ Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
+ (license 14) Tested by: atis ........
+
+ * channels/chan_misdn.c: Mostly spacing changes; no functionality
+ change at all.
+
+2009-01-14 02:00 +0000 [r168594] Terry Wilson <twilson@digium.com>
+
+ * /, apps/app_page.c: Merged revisions 168593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009)
+ | 20 lines Don't overflow when paging more than 128 extensions
+ The number of available slots for calls in app_page was hardcoded
+ to 128. Proper bounds checking was not in place to enforce this
+ limit, so if more than 128 extensions were passed to the Page()
+ app, Asterisk would crash. This patch instead dynamically
+ allocates memory for the ast_dial structures and removes the
+ (non-functional) arbitrary limit. This issue would have special
+ importance to anyone who is dynamically creating the argument
+ passed to the Page application and allowing more than 128
+ extensions to be added by an outside user via some external
+ interface. The patch posted by a_villacis was slightly modified
+ for some coding guidelines and other cleanups. Thanks,
+ a_villacis! (closes issue #14217) Reported by: a_villacis
+ Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
+ uploaded by a (license 660) Tested by: otherwiseguy ........
+
+2009-01-13 23:57 +0000 [r168591] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_misdn.c: Janitor patch for chan_misdn (make channel
+ variable access safe) (closes issue #12887) Reported by: pputman
+ Patches: chan_misdn_threadsafe.patch uploaded by pputman (license
+ 81)
+
+2009-01-13 23:05 +0000 [r168585-168588] Terry Wilson <twilson@digium.com>
+
+ * res/res_http_post.c: Fully overwrite a same-named file when
+ uploading (closes issue #14190) Reported by: timking
+
+ * Makefile, include/asterisk/options.h, main/asterisk.c: Add option
+ to hide console connect messages (closes issue #14222) Reported
+ by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded
+ by jamesgolovich (license 176) Tested by: otherwiseguy
+
+2009-01-13 22:30 +0000 [r168579] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Clarify a message that app_queue prints and
+ change to a debug-level message The "No one is answering..."
+ verbose message contained 3 numbers that were not explained in
+ any way to whoever was viewing the message. It is more helpful
+ now since the message explains what the numbers mean. Also, the
+ message has been downgraded to "DEBUG" level. (closes issue
+ #14172) Reported by: caio1982 Patches: queue_answering_debug.diff
+ uploaded by caio1982 (license 22)
+
+2009-01-13 22:22 +0000 [r168578] Terry Wilson <twilson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168551 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009)
+ | 7 lines Don't pass a value with a side effect to a macro
+ (closes issue #14176) Reported by: paraeco Patches:
+ chan_sip.c.diff uploaded by paraeco (license 658) ........
+
+2009-01-13 21:18 +0000 [r168575] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Allow
+ specifying a port number in the user portion of a register =>
+ line in sip.conf With this commit, a register => line in sip.conf
+ may contain a port number in the "user" section of the line.
+ Please see CHANGES and sip.conf.sample for more details regarding
+ this. (closes issue #14198) Reported by: Nick_Lewis Patches:
+ chan_sip.c-domainport2.patch uploaded by Nick (license 657)
+ Tested by: Nick_Lewis
+
+2009-01-13 19:22 +0000 [r168562] Russell Bryant <russell@digium.com>
+
+ * channels/chan_unistim.c, main/pbx.c, apps/app_read.c, /,
+ include/asterisk/indications.h, apps/app_readexten.c,
+ apps/app_disa.c, include/asterisk/channel.h, main/indications.c,
+ main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
+ funcs/func_channel.c, main/app.c, res/snmp/agent.c,
+ res/res_indications.c: Merged revisions 168561 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009)
+ | 2 lines Revert unnecessary indications API change from rev
+ 122314 ........
+
+2009-01-13 17:51 +0000 [r168547] Tilghman Lesher <tlesher@digium.com>
+
+ * /, funcs/func_logic.c: Merged revisions 168546 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009)
+ | 6 lines If either conditional is NULL, don't try copying it.
+ (closes issue #14226) Reported by: caspy Patches:
+ 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
+ ........
+
+2009-01-13 16:02 +0000 [r168539] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
+
+ * main/taskprocessor.c: correct a CLI description
+
+2009-01-12 23:45 +0000 [r168526] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_alsa.c: Merged revisions 167095 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31
+ Dec 2008) | 5 lines Repeat attempts to write when we receive
+ -EAGAIN from the driver, as detailed in the ALSA sample code (see
+ http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
+ Reported by: Jerry Geis (via the -users list) Fixed by: me
+ (license 14) ........
+
+2009-01-12 23:12 +0000 [r168523] Mark Michelson <mmichelson@digium.com>
+
+ * main/srv.c: bump the verbosity of a message in srv.c up by one.
+ It used to be at this level prior to a large patch merge which
+ converted ast_verbose calls to ast_verb (closes issue #14221)
+ Reported by: jcovert Patches: srv.c.patch uploaded by jcovert
+ (license 551)
+
+2009-01-12 23:06 +0000 [r168522] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/app.c: Some platforms (notably, the BSDs) have a more
+ efficient implementation called closefrom(3).
+
+2009-01-12 21:51 +0000 [r168508-168517] Jeff Peeler <jpeeler@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 168516 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009)
+ | 5 lines (closes issue #13881) Reported by: hoowa Update the app
+ CDR field for AGI commands that are not executing an application
+ via "exec". ........
+
+ * /, channels/chan_agent.c: Merged revisions 168507 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12
+ Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG
+ Tested by: denisgalvao This gits rid of the notion of an
+ owning_app allowing the request and hangup to be initiated by
+ different threads. Originating from an active agent channel
+ requires this. The implementation primarily changes __login_exec
+ to wait on a condition variable rather than a lock. Review:
+ http://reviewboard.digium.com/r/35/ ........
+
+2009-01-12 16:31 +0000 [r168497] Olle Johansson <oej@edvina.net>
+
+ * apps/app_minivm.c: Better to use the proper app name
+
+2009-01-12 15:00 +0000 [r168485] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Merged revisions 168482 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan
+ 2009) | 5 lines I am reverting the fix made in revision 168128
+ (and its upward merges) after being contacted by Olle Johansson
+ and being shown how this fix is incorrect. Thanks to Olle for
+ clearing this up for me. ........
+
+2009-01-12 14:57 +0000 [r168481] Russell Bryant <russell@digium.com>
+
+ * /, configs/indications.conf.sample: Merged revisions 168480 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009)
+ | 2 lines s/ringdance/ringcadence/ for Bulgaria ........
+
+2009-01-12 14:35 +0000 [r168479] Olle Johansson <oej@edvina.net>
+
+ * main/asterisk.c: Don't include swap.h unless we have swapctl
+
+2009-01-10 01:42 +0000 [r168334] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: sizeof for a stringfield is 4. Kinda low for
+ reconstructing a field value.
+
+2009-01-09 23:16 +0000 [r168270] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, sounds/Makefile: Merged revisions 168267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan
+ 2009) | 1 line update to use new sound file packages that include
+ license files ........
+
+2009-01-09 23:15 +0000 [r168269] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Spacing change
+
+2009-01-09 23:04 +0000 [r168265] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/sip_nat_settings (added), CHANGES: Add a script
+ to find out the correct settings for Asterisk behind NAT (closes
+ issue #13065) Reported by: tzafrir Patches: sip_nat_settings
+ uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by
+ mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and
+ moi
+
+2009-01-09 22:21 +0000 [r168200] Russell Bryant <russell@digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 168198 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09
+ Jan 2009) | 2 lines Make this compile for mvanbaak ........
+
+2009-01-09 21:53 +0000 [r168193] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 168128 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan
+ 2009) | 13 lines Add check_via calls to more request handlers
+ INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
+ checking the topmost Via to determine where to send the response.
+ Adding check_via calls to those request handlers solves this.
+ (closes issue #13071) Reported by: baron Patches: check_via.patch
+ uploaded by baron (license 531) Tested by: baron ........
+
+2009-01-09 21:43 +0000 [r168192] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 168191 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09
+ Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 *
+ Miscellaneous doxygen comments added. ........
+
+2009-01-09 20:25 +0000 [r168142] Terry Wilson <twilson@digium.com>
+
+ * res/res_phoneprov.c: Don't leak memory if phoneprov.conf does not
+ exist (closes issue #14203) Reported by: jamesgolovich Patches:
+ asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich
+ (license 176)
+
+2009-01-09 18:30 +0000 [r168090] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c, include/asterisk/strings.h: When using ast_str
+ with a non-ast_str-enabled API, we need to update the buffer or
+ otherwise, we cannot use ast_str_strlen().
+
+2009-01-09 18:01 +0000 [r168014-168054] Matthew Nicholson <mnicholson@digium.com>
+
+ * main/logger.c: Added a comment to logger.c about where to put
+ includes
+
+ * main/logger.c: Use ast_safe_system() in logger.c instead of
+ system() (closes issue #14194) Reported by: pabelanger
+
+2009-01-09 01:15 +0000 [r167935-167973] Terry Wilson <twilson@digium.com>
+
+ * apps/app_originate.c: Set ORIGINATE_STATUS instead of
+ OUTGOING_STATUS to match the documentation
+
+ * apps/app_dial.c: Set peer context and exten values so MACRO_EXTEN
+ and MACRO_CONTEXT will be set
+
+2009-01-08 22:37 +0000 [r167894] Tilghman Lesher <tlesher@digium.com>
+
+ * /, res/res_agi.c: Merged revisions 167840 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009)
+ | 6 lines Don't truncate database results at 255 chars. (closes
+ issue #14069) Reported by: evandro Patches:
+ 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
+ ........
+
+2009-01-08 22:34 +0000 [r167888] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Revert chan_sip changes which were
+ accidentally committed in revision 167792
+
+2009-01-08 21:40 +0000 [r167835-167837] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_minivm.c: Fix variables to comply with documentation
+ changes
+
+ * apps/app_minivm.c: Textual changes, consistency in status
+ variable naming, and other minor bugs. (closes issue #13943)
+ Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded
+ by Marquis (license 32)
+
+2009-01-08 19:48 +0000 [r167792] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c, CHANGES, apps/app_queue.c: Add the average
+ talk time for a queue This patch adds the functionality to
+ app_queue of calculating the average amount of time that channels
+ are bridged for a queue. The algorithm used to calculate the
+ average is the same exponential average currently used to
+ calculate the average holdtime. See the CHANGES file to see the
+ methods you may use to view this information. (closes issue
+ #13960) Reported by: coolmig Patches:
+ app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)
+
+2009-01-08 19:44 +0000 [r167791] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c, CHANGES: Convert dialplan application
+ DAHDISendCallreroutingFacility to use commas. (closes issue
+ #13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded
+ by eliel (license 64)
+
+2009-01-08 17:26 +0000 [r167700-167720] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 167714 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan
+ 2009) | 1 line remove an unnecessary argument to queue_request()
+ ........
+
+ * channels/chan_sip.c: Merged revisions 167620 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan
+ 2009) | 5 lines When a SIP request or response arrives for a
+ dialog with an associated Asterisk channel, and the lock on that
+ channel cannot be obtained because it is held by another thread,
+ instead of dropping the request/response, queue it for later
+ processing when the channel lock becomes available.
+ http://reviewboard.digium.com/r/123/ ........
+
+2009-01-08 14:27 +0000 [r167662] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/sip-friends.sql: Oops... fix the fieldname I
+ changed yesterday to be right.
+
+2009-01-07 22:36 +0000 [r167542-167569] Russell Bryant <russell@digium.com>
+
+ * /, main/file.c: Merged revisions 167566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009)
+ | 2 lines Fix the last couple of places where free() was
+ improperly used directly. ........
+
+ * /, main/file.c: Merged revisions 167554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009)
+ | 2 lines Don't fclose() the file early, the filestream
+ destructor will handle it. ........
+
+ * /, main/file.c: Merged revisions 167545 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009)
+ | 2 lines Only try to close the file if one was actually opened
+ ........
+
+ * /, main/file.c: Merged revisions 167541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009)
+ | 4 lines Don't use free() directly. This caused a crash since
+ ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
+ #asterisk-dev ........
+
+2009-01-07 18:20 +0000 [r167478] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_followme.c: Answer the channel if it has not already
+ been answered and we've already found a valid profile for
+ followme. (closes issue #14140) Reported by: dimas Patches:
+ 14140.patch uploaded by dimas
+
+2009-01-07 18:18 +0000 [r167477] Leif Madsen <lmadsen@digium.com>
+
+ * configs/queues.conf.sample: Update queues.conf.sample
+ documentation. Update the queues.conf.sample documentation to
+ mention that you need to preload chan_local.so as well if you
+ plan on using Local channels for queue members, and you're
+ preloading pbx_config.so. (closes issue #14179) Reported by:
+ CrashHD Tested by: CrashHD
+
+2009-01-07 17:35 +0000 [r167442] Russell Bryant <russell@digium.com>
+
+ * /, main/indications.c: Merged revisions 167432 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009)
+ | 4 lines Treat an empty string the same way as a NULL country
+ argument. In passing, simplify the handling of returning a
+ default tone zone. ........
+
+2009-01-07 17:05 +0000 [r167416] Doug Bailey <dbailey@digium.com>
+
+ * channels/chan_dahdi.c: Cleanup fsk spill if off hook is detected
+ during mwi spill. Correct logic error in handling events when
+ sending mwi spill (closes issue #14143) Reported by: alecdavis
+ Patches: chan_dahdi.handle_init_event2.diff.txt uploaded by
+ dbailey
+
+2009-01-07 14:26 +0000 [r167373] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/sip-friends.sql: Update the sip-friends.sql file
+ to use the non-deprecated 'defaultname' instead of 'username' and
+ remove an extra comma that would cause the script to fail as-is
+
+2009-01-06 21:36 +0000 [r167301] Mark Michelson <mmichelson@digium.com>
+
+ * /, main/db.c: Merged revisions 167299 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan
+ 2009) | 8 lines Use the correct variable when creating the format
+ string (closes issue #14177) Reported by: nic_bellamy Patches:
+ asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
+ (license 299) ........
+
+2009-01-06 21:02 +0000 [r167265] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 167260 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r167260 | tilghman | 2009-01-06 14:48:05 -0600
+ (Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
+ Jan 2009) | 2 lines Security fix AST-2009-001. ........
+ ................
+
+2009-01-05 16:59 +0000 [r167180] Mark Michelson <mmichelson@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 167179 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan
+ 2009) | 41 lines A couple of changes to T.38 SDP attribute
+ handling There are some boolean attributes for T.38 such as
+ T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
+ T38FaxTranscodingJBIG. By simply being present, we should treat
+ these as a "true" value. The current code, however, was requiring
+ a 1 or 0 as the value of the attribute in order to parse it. This
+ is due to the fact that there are some T.38 endpoints and
+ gateways that also transmit this information incorrectly. This
+ patch follows the "be liberal in what you accept and strict in
+ what you send" philosophy by accepting both the correctly- and
+ incorrectly-formatted attributes, but only sending information as
+ it is supposed to be sent. It was also discovered that a
+ particular type of T.38 gateway sends some non-standard T.38 SDP
+ attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
+ it used T38MaxDatagram and T38FaxMaxRate respectively. We now
+ will properly accept these attributes as well. Note that there
+ are a lot of patches cited in the below commit message template.
+ This is because the person who submitted these patches is an
+ awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
+ issue #13976) Reported by: linulin Patches:
+ chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
+ by arcivanov (license 648)
+ chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) Tested by: arcivanov ........
+
+2009-01-05 16:44 +0000 [r167176] Tilghman Lesher <tlesher@digium.com>
+
+ * UPGRADE-1.6.txt: More clearly explain that quote marks are no
+ longer necessary. (closes issue #13718) Reported by: davidw
+ Patches: 20081020__bug13718.diff.txt uploaded by Corydon76
+ (license 14) Tested by: blitzrage
+
+2009-01-03 20:29 +0000 [r167125] Jeff Peeler <jpeeler@digium.com>
+
+ * main/asterisk.c: When parsing environment variable
+ ASTERISK_PROMPT, make sure to proceed to the next character when
+ a non format specifier is used (no %). Otherwise, the while loop
+ looking for the null byte will never exit.
+
+2008-12-31 23:07 +0000 [r167061] Sean Bright <sean.bright@gmail.com>
+
+ * doc/CODING-GUIDELINES, include/asterisk.h, channels/h323/README:
+ Mostly just whitespace, but also convert 'CVS' to 'SVN' in a
+ couple places and fix a few typos I found in the
+ CODING_GUIDELINES.
+
+2008-12-31 22:53 +0000 [r167057] Terry Wilson <twilson@digium.com>
+
+ * main/xmldoc.c: Don't forget to free typename
+
+2008-12-31 21:52 +0000 [r167021] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c: Change some incorrect syntax for pri set
+ debug and correct an off-by-one error in ss7 set debug command
+
+2008-12-31 19:39 +0000 [r166954-166958] Tilghman Lesher <tlesher@digium.com>
+
+ * main/ast_expr2.h, main/ast_expr2.c: That was weird...
+
+ * channels/chan_local.c, /, main/ast_expr2.h, main/ast_expr2.c:
+ Merged revisions 166953 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008)
+ | 5 lines Also inherit the musiconhold class. (Closes #14153)
+ Reported by: Jerry Geis, via the users list. Patch by: me
+ (license 14) ........
+
+2008-12-30 20:50 +0000 [r166908] Terry Wilson <twilson@digium.com>
+
+ * res/res_phoneprov.c, doc/sip-retransmit.txt,
+ doc/tex/phoneprov.tex, res/res_http_post.c,
+ phoneprov/polycom_line.xml, doc/realtimetext.txt: Fix some
+ svn:keywords
+
+2008-12-29 18:04 +0000 [r166861] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, apps/app_queue.c: Update app_queue to deal with
+ the removal of AST_PBX_KEEPALIVE When placing a call to a queue
+ which ran a gosub on the member's channel, Asterisk would crash
+ every time, stemming from the fact that the member's channel was
+ being hung up unexpectedly when the Gosub completed. The
+ necessary change was pretty much copied and pasted from
+ app_dial's similar changes made last week. I also took the
+ opportunity to change a LOG_DEBUG message in app_dial to use
+ ast_debug. I am guessing this was due to a direct merge from 1.4
+ that was not corrected to use trunk's preferred syntax.
+
+2008-12-28 15:36 +0000 [r166823] Eliel C. Sardanons <eliels@gmail.com>
+
+ * funcs/func_audiohookinherit.c: Fix a typo in the XML
+ documentation of the AUDIOHOOK_INHERIT dialplan function.
+
+2008-12-28 15:15 +0000 [r166773] Russell Bryant <russell@digium.com>
+
+ * /, channels/misdn_config.c: Merged revisions 166772 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28
+ Dec 2008) | 4 lines Use strncat() instead of an sprintf() in
+ which source and target buffers overlap
+ http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
+ ........
+
+2008-12-24 15:10 +0000 [r166731] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: There is no section 22.2.2 in rfc 3261. I
+ believe 26.2.2 is what was meant: Note that in the SIPS URI
+ scheme, transport is independent of TLS, and thus
+ "sips:alice@atlanta.com;transport=tcp" and
+ "sips:alice@atlanta.com;transport=sctp" are both valid (although
+ note that UDP is not a valid transport for SIPS). The use of
+ "transport=tls" has consequently been deprecated, partly because
+ it was specific to a single hop of the request. This is a change
+ since RFC 2543.
+
+2008-12-23 20:47 +0000 [r166696] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Allow semicolons and extended characters in
+ user-specified SIP headers. (closes issue #14110) Reported by:
+ gork Patches: 20081222__bug14110__2.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: gork, putnopvut
+
+2008-12-23 18:13 +0000 [r166665] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, /, main/features.c,
+ apps/app_macro.c, include/asterisk/pbx.h, apps/app_queue.c,
+ include/asterisk/features.h: Merged revisions 166093 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4 In
+ order to merge this 1.4 patch into trunk, I had to resolve some
+ conflicts and wait for Russell to make some changes to res_agi. I
+ re-ran all the tests; 39 calls in all, and made fairly careful
+ notes and comparisons: I don't want this to blow up some aspect
+ of asterisk; I completely removed the KEEPALIVE from the pbx.h
+ decls. The first 3 scenarios involving feature park; feature xfer
+ to 700; hookflash park to Park() app call all behave the same,
+ don't appear to leave hung channels, and no crashes. ........
+ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) |
+ 131 lines This merges the masqpark branch into 1.4 These changes
+ eliminate the need for (and use of) the KEEPALIVE return code in
+ res_features.c; There are other places that use this result code
+ for similar purposes at a higher level, these appear to be left
+ alone in 1.4, but attacked in trunk. The reason these changes are
+ being made in 1.4, is that parking ends a channel's life, in some
+ situations, and the code in the bridge (and some other places),
+ was not checking the result code properly, and dereferencing the
+ channel pointer, which could lead to memory corruption and
+ crashes. Calling the masq_park function eliminates this danger in
+ higher levels. A series of previous commits have replaced some
+ parking calls with masq_park, but this patch puts them ALL to
+ rest, (except one, purposely left alone because a masquerade is
+ done anyway), and gets rid of the code that tests the KEEPALIVE
+ result, and the NOHANGUP_PEER result codes. While bug 13820
+ inspired this work, this patch does not solve all the problems
+ mentioned there. I have tested this patch (again) to make sure I
+ have not introduced regressions. Crashes that occurred when a
+ parked party hung up while the parking party was listening to the
+ numbers of the parking stall being assigned, is eliminated. These
+ are the cases where parking code may be activated: 1. Feature one
+ touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3.
+ Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi
+ hookflash xfer to 700) 4. Run Park via manager. The interesting
+ testing cases for parking are: I. A calls B, A parks B a. B hangs
+ up while A is getting the numbers announced. b. B hangs up after
+ A gets the announcement, but before the parking time expires c. B
+ waits, time expires, A is redialed, A answers, B and A are
+ connected, after which, B hangs up. d. C picks up B while still
+ in parking lot. II. A calls B, B parks A a. A hangs up while B is
+ getting the numbers announced. b. A hangs up after B gets the
+ announcement, but before the parking time expires c. A waits,
+ time expires, B is redialed, B answers, A and B are connected,
+ after which, A hangs up. d. C picks up A while still in parking
+ lot. Testing this throroughly involves acting all the
+ permutations of I and II, in situations 1,2,3, and 4. Since I
+ added a few more changes (ALL references to KEEPALIVE in the
+ bridge code eliimated (I missed one earlier), I retested most of
+ the above cases, and no crashes. H-extension weirdness. Current
+ h-extension execution is not completely correct for several of
+ the cases. For the case where A calls B, and A parks B, the 'h'
+ exten is run on A's channel as soon as the park is accomplished.
+ This is expected behavior. But when A calls B, and B parks A,
+ this will be current behavior: After B parks A, B is hung up by
+ the system, and the 'h' (hangup) exten gets run, but the channel
+ mentioned will be a derivative of A's... Thus, if A is DAHDI/1,
+ and B is DAHDI/2, the h-extension will be run on channel
+ Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be
+ those relating to Channel A. And, in the case where A is
+ reconnected to B after the park time expires, when both parties
+ hang up after the joyful reunion, no h-exten will be run at all.
+ In the case where C picks up A from the parking lot, when either
+ A or C hang up, the h-exten will be run for the C channel. CDR's
+ are a separate issue, and not addressed here. As to WHY this
+ strange behavior occurs, the answer lies in the procedure
+ followed to accomplish handing over the channel to the parking
+ manager thread. This procedure is called masquerading. In the
+ process, a duplicate copy of the channel is created, and most of
+ the active data is given to the new copy. The original channel
+ gets its name changed to XXX<ZOMBIE> and keeps the PBX
+ information for the sake of the original thread (preserving its
+ role as a call originator, if it had this role to begin with),
+ while the new channel is without this info and becomes a call
+ target (a "peer"). In this case, the parking lot manager thread
+ is handed the new (masqueraded) channel. It will not run an
+ h-exten on the channel if it hangs up while in the parking lot.
+ The h exten will be run on the original channel instead, in the
+ original thread, after the bridge completes. See bug 13820 for
+ our intentions as to how to clean up the h exten behavior.
+ Review: http://reviewboard.digium.com/r/29/ ........
+
+2008-12-23 16:04 +0000 [r166625] Russell Bryant <russell@digium.com>
+
+ * CHANGES: Fix spelling error.
+
+2008-12-23 15:17 +0000 [r166569] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, /: Merged revisions 166568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec
+ 2008) | 12 lines Fix a crash resulting from a datastore with
+ inheritance but no duplicate callback The fix for this is to
+ simply set the newly created datastore's data pointer to NULL if
+ it is inherited but has no duplicate callback. (closes issue
+ #14113) Reported by: francesco_r Patches: 14113.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r ........
+
+2008-12-23 04:32 +0000 [r166533] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c, /: Merged revisions 166509 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008)
+ | 4 lines Use the integer form of condition for integer
+ comparisons. (closes issue #14127) Reported by: andrew ........
+
+2008-12-22 23:25 +0000 [r166470] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_agi.c: Always use the value of the AGISIGHUP when running
+ an AGI. Prior to this patch, the value of AGISIGUP was not always
+ honored when set on a channel. (closes issue #13711) Reported by:
+ fmueller Patches: 13711.patch uploaded by putnopvut (license 60)
+
+2008-12-22 21:45 +0000 [r166436] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Cosmetic change - don't mix struct
+ initializer styles.
+
+2008-12-22 21:08 +0000 [r166382] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 166380 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon,
+ 22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks
+ and autoservice It has been discovered that if a channel is
+ locked prior to a call to ast_autoservice_stop, then it is likely
+ that a deadlock will occur. The reason is that the call to
+ ast_autoservice_stop has a check built into it to be sure that
+ the thread running autoservice is not currently trying to
+ manipulate the channel we are about to pull out of autoservice.
+ The autoservice thread, however, cannot advance beyond where it
+ currently is, though, because it is trying to acquire the lock of
+ the channel for which autoservice is attempting to be stopped.
+ The gist of all this is that a channel MUST NOT be locked when
+ attempting to stop autoservice on the channel. In this particular
+ case, the channel was locked by a call to ast_read. A call to
+ ast_exists_extension led to autoservice being started and stopped
+ due to the existence of dialplan switches. It may be that there
+ are future commits which handle the same symptoms but in a
+ different location, but based on my looks through the code, it is
+ very rare to see a construct such as this one. (closes issue
+ #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
+ by putnopvut (license 60) Tested by: rtrauntvein Review:
+ http://reviewboard.digium.com/r/107/ ........
+
+2008-12-22 20:26 +0000 [r166273-166377] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Fix a bad typo.
+
+ * main/astobj2.c: Remove some error messages. This is the default
+ handler that is valid to use.
+
+ * /, main/utils.c: Merged revisions 166297 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008)
+ | 2 lines Fix up timeout handling in ast_carefulwrite(). ........
+
+ * include/asterisk/utils.h, main/manager.c, main/utils.c: Introduce
+ ast_careful_fwrite() and use in AMI to prevent partial writes.
+ This patch introduces a function to do careful writes on a file
+ stream which will handle timeouts and partial writes. It is
+ currently used in AMI to address the issue that has been
+ reported. However, there are probably a few other places where
+ this could be used. (closes issue #13546) Reported by: srt Tested
+ by: russell http://reviewboard.digium.com/r/104/
+
+ * res/res_musiconhold.c: Re-work ref count handling of MoH classes
+ using astobj2 to resolve crashes. (closes issue #13566) Reported
+ by: igorcarneiro Tested by: russell Review:
+ http://reviewboard.digium.com/r/106/
+
+2008-12-22 16:08 +0000 [r166268] Joshua Colp <jcolp@digium.com>
+
+ * main/dnsmgr.c: Record the previous port in the temporary address
+ structure so that the comparison does not treat the host as
+ having changed even if it did not. This would have been
+ uninitialized before and would have led to a baddddd port.
+ (closes issue #13628) Reported by: pananix Patches:
+ bug13628.patch uploaded by jpeeler (license 325) Tested by: file,
+ blitzrage
+
+2008-12-22 16:07 +0000 [r166267] Mark Michelson <mmichelson@digium.com>
+
+ * funcs/func_timeout.c, main/file.c: Fix a file playback crash and
+ explicitly initialize values in func_timeout.c A crash was
+ brought up on the bugtracker. The first run through valgrind was
+ full of legitimate complaints of uninitialized values in
+ func_timeout when setting a response timeout. These were fixed
+ but the crash persisted. A second run through showed the real
+ problem. The reference counting used for filestreams was
+ incorrect because there were some missing increments when a frame
+ was read from a format module. (closes issue #14118) Reported by:
+ blitzrage Patches: 14118v2.patch uploaded by putnopvut (license
+ 60) Tested by: blitzrage
+
+2008-12-22 14:16 +0000 [r166258] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: Remove AST_PBX_KEEPALIVE usage from res_agi. This
+ patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The
+ only usage was for the AGI command, "asyncagi break". This patch
+ removes this feature. Normally, a feature would not be removed
+ like this. However, this code is broken and usage of it will
+ result in a memory leak. Usage of this feature will make the AGI
+ code return a result of AST_PBX_KEEPALIVE. The PBX handler
+ assumes that another thread has assumed ownership of the channel.
+ The channel thread will exit without destroying the channel.
+ Unfortunately, _no_ thread has ownership of the channel at this
+ point. There are a couple of serious problems here: 1) The only
+ way to recover the caller is to issue a channel redirect. This
+ will work, but this will be done with a masquerade, and the old
+ ast_channel structure will be lost. 2) Until the channel redirect
+ happens, there is no code servicing the channel. That means
+ nothing is reading audio or handling events coming from the
+ channel. This is very bad. The recommended way to get this same
+ "break" functionality is to issue the redirect while the channel
+ is still being handled by the AGI code. That way, there will be
+ no memory leak, and there will be no period of time that the
+ channel is not being serviced.
+
+2008-12-20 01:37 +0000 [r166219] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/doxyref.h: Make a note about formatting the
+ review URL in commit messages
+
+2008-12-19 23:45 +0000 [r166092-166162] Mark Michelson <mmichelson@digium.com>
+
+ * main/audiohook.c: Get rid of an extra space. I don't know how
+ this crept back in when I had already fixed it earlier
+
+ * funcs/func_audiohookinherit.c: Remove the verbatim tag from the
+ author line I could have sworn I already did that before,
+ though...
+
+ * main/channel.c, funcs/func_audiohookinherit.c (added),
+ include/asterisk/audiohook.h, main/audiohook.c, CHANGES: Adding a
+ new dialplan function AUDIOHOOK_INHERIT This function is being
+ added as a method to allow for an audiohook to move to a new
+ channel during a channel masquerade. The most obvious use for
+ such a facility is for MixMonitor when a transfer is performed.
+ Prior to the addition of this functionality, if a channel running
+ MixMonitor was transferred by another party, then the recording
+ would stop once the transfer had completed. By using
+ AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the
+ call even after the transfer has completed. It has also been
+ determined that since this is seen by most as a bug fix and is
+ not an invasive change, this functionality will also be
+ backported to 1.4 and merged into the 1.6.0 branches, even though
+ they are feature-frozen. (closes issue #13538) Reported by: mbit
+ Patches: 13538.patch uploaded by putnopvut (license 60) Tested
+ by: putnopvut Review: http://reviewboard.digium.com/r/102/
+
+2008-12-19 21:44 +0000 [r166058] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Add configuration
+ support for half_full DAHDI buffer policy
+
+2008-12-19 18:20 +0000 [r165954] Eliel C. Sardanons <eliels@gmail.com>
+
+ * apps/app_record.c: Fix the XML documentation for Record().
+ <value> tags inside <variable> elements must have CDATA and no
+ another XML node.
+
+2008-12-19 15:05 +0000 [r165801-165890] Russell Bryant <russell@digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 165889 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008)
+ | 9 lines Ensure that the chanspy datastore is fully initialized.
+ This patch resolved some random crash issues observed by a user
+ on a BSD system (closes issue #14111) Reported by: ys Patches:
+ app_chanspy.c.diff uploaded by ys (license 281) ........
+
+ * include/asterisk/doxyref.h: Disable some automatic links
+ generated by doxygen.
+
+ * include/asterisk/doxyref.h: Introduce commit message formatting
+ guidelines. This documents the recommended outline to use for
+ commit message. It also covers information on special tags that
+ can be used in commit messages, as well as the template
+ functionality that is available on bugs.digium.com. Review:
+ http://reviewboard.digium.com/r/96/
+
+ * /, main/utils.c: Merged revisions 165796 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008)
+ | 11 lines Make ast_carefulwrite() be more careful. This patch
+ handles some additional cases that could result in partial writes
+ to the file description. This was done to address complaints
+ about partial writes on AMI. (issue #13546) (more changes needed
+ to address potential problems in 1.6) Reported by: srt Tested by:
+ russell Review: http://reviewboard.digium.com/r/99/ ........
+
+2008-12-18 21:43 +0000 [r165798] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: (closes issue #13993) Reported by: mika Add
+ ActionID response to ping if sent with request.
+
+2008-12-18 21:41 +0000 [r165797] Tilghman Lesher <tlesher@digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 165767 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18
+ Dec 2008) | 8 lines Add mutexes around accesses to the IMAP
+ library interface. This prevents certain crashes, especially when
+ shared mailboxes are used. (closes issue #13653) Reported by:
+ howardwilkinson Patches:
+ asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
+ howardwilkinson (license 590) Tested by: jpeeler ........
+
+2008-12-18 21:21 +0000 [r165792] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_dahdi.c, channels/chan_misdn.c,
+ channels/chan_sip.c, pbx/pbx_ael.c, apps/app_queue.c,
+ channels/chan_oss.c: Numerous documentation updates. (closes
+ issue #13970) Reported by: pkempgen Patches:
+ __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage
+ (license 10)
+
+2008-12-18 19:34 +0000 [r165724] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_odbc.c: Fix crashes in res_odbc. The variable "class" was
+ being set NULL just prior to being dereferenced in an ao2_link
+ call. I have moved the setting of the variable to NULL until
+ after the ao2_link call.
+
+2008-12-18 19:33 +0000 [r165662-165723] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: Remove the
+ need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This
+ is part of an effort to completely remove AST_PBX_KEEPALIVE and
+ other similar return codes from the source. While this usage was
+ perfectly safe, there are others that are problematic. Since we
+ know ahead of time that we do not want to PBX to destroy the
+ channel, the PBX API has been changed so that information can be
+ provided as an argument, instead, thus removing the need for the
+ KEEPALIVE return value. Further changes to get rid of KEEPALIVE
+ and related code is being done by murf. There is a patch up for
+ that on review 29. Review: http://reviewboard.digium.com/r/98/
+
+ * /, res/res_musiconhold.c: Merged revisions 165661 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18
+ Dec 2008) | 7 lines Set the process group ID on the MOH process
+ so that all children will get killed (closes issue #14099)
+ Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2
+ uploaded by caspy (license 645) ........
+
+2008-12-18 18:36 +0000 [r165658] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Fix 2 resource leaks and fix another
+ pipe-to-comma conversion
+
+2008-12-18 17:13 +0000 [r165599] Joshua Colp <jcolp@digium.com>
+
+ * /, main/rtp.c: Merged revisions 165591 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4
+ lines Only care about a compatible codec for early bridging if we
+ are actually bridging to another channel. If we are not we
+ actually want to bring the audio back to us. (closes issue
+ #13545) Reported by: davidw ........
+
+2008-12-18 16:36 +0000 [r165541] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Fix reference counts of the class and add an
+ assertion to the end.
+
+2008-12-18 15:25 +0000 [r165502] Eliel C. Sardanons <eliels@gmail.com>
+
+ * main/strings.c, include/asterisk/strings.h: Remove duplicate code
+ from the ast_str API. We now use __AST_STR_* to access 'struct
+ ast_str' members, but this must only be used inside the API
+ implementation. (closes issue #14098) Reported by: eliel Patches:
+ ast_str.patch uploaded by eliel (license 64)
+
+2008-12-18 14:23 +0000 [r165433-165469] Russell Bryant <russell@digium.com>
+
+ * apps/app_originate.c: Add a \todo note for app_originate. Jared
+ Smith suggested that we add a way to be able to set variables and
+ functions on the outbound channel. I think that it's a great
+ idea, so I have added it as a todo so that it gets done at some
+ point.
+
+ * apps/app_originate.c (added), CHANGES: Add a new application,
+ Originate. (closes issue #14075) Reported by: rcasas Patches:
+ app_originate.c uploaded by rcasas (license 641), heavily
+ modified by me Tested by: russell Review:
+ http://reviewboard.digium.com/r/95/
+
+2008-12-17 23:39 +0000 [r165397] Tilghman Lesher <tlesher@digium.com>
+
+ *