diff options
| author | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-05-06 14:39:23 +0000 |
|---|---|---|
| committer | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-05-06 14:39:23 +0000 |
| commit | deae75ccb2047e3b30dc440bba292f778dc09291 (patch) | |
| tree | cc0edc19754f8a53086e40035f0082c254b5ab1b | |
| parent | df2cc9ba334947c023f819b21c05500268fe5f63 (diff) | |
Importing files for 1.6.0.28-rc1 release.
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.28-rc1@261546 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | .lastclean | 1 | ||||
| -rw-r--r-- | .version | 1 | ||||
| -rw-r--r-- | ChangeLog | 58079 |
3 files changed, 58081 insertions, 0 deletions
diff --git a/.lastclean b/.lastclean new file mode 100644 index 000000000..7facc8993 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +36 diff --git a/.version b/.version new file mode 100644 index 000000000..10819f3bb --- /dev/null +++ b/.version @@ -0,0 +1 @@ +1.6.0.28-rc1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 000000000..40884f03b --- /dev/null +++ b/ChangeLog @@ -0,0 +1,58079 @@ +2010-05-06 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.0.28-rc1 Released + +2010-05-05 19:19 +0000 [r261326] Paul Belanger <paul.belanger@polybeacon.com> + + * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May + 2010) | 19 lines Merged revisions 261274 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May + 2010) | 12 lines Registration fix for SIP realtime. Make sure + realtime fields are not empty. (closes issue #17266) Reported by: + Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick + Lewis (license 657) Tested by: Nick_Lewis, sberney Review: + https://reviewboard.asterisk.org/r/643/ ........ ................ + +2010-05-04 23:55 +0000 [r261096] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 261095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010) + | 18 lines Merged revisions 261093-261094 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010) + | 7 lines Protect against overflow, when calculating how long to + wait for a frame. (closes issue #17128) Reported by: under + Patches: d.diff uploaded by under (license 914) ........ r261094 + | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2 + lines Add a tiny corner case to the previous commit ........ + ................ + +2010-05-04 18:53 +0000 [r260925] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500 + (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) + | 12 lines Voicemail transfer to operator should occur + immediately, not after main menu. There were two scenarios in the + advanced options that while using the operator=yes and review=yes + options, the transfer occurred only after exiting the main menu + (after sending a reply or leaving a message for an extension). + Now after the audio is processed for the reply or message the + transfer occurs immediately as expected. ABE-2107 ABE-2108 + ........ ................ + +2010-05-04 15:51 +0000 [r260744-260803] Jason Parker <jparker@digium.com> + + * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260802 | qwell | 2010-05-04 10:49:57 -0500 + (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May + 2010) | 1 line Fix fallout from removing from configure script. + Pointed out by philipp64 on #asterisk-dev ........ + ................ + + * /: Fix merge props + +2010-05-03 17:29 +0000 [r260721] Paul Belanger <paul.belanger@polybeacon.com> + + * Makefile, /: Merged revisions 260661-260662 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May + 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend + libdir when executing mkpkgconfig allowing non-root installs to + work. (closes issue #17268) Reported by: pabelanger Patches: + issue17268.patch uploaded by pabelanger (license 224) Tested by: + pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41 + -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/ + part. Thanks Qwell. ........ + +2010-05-03 14:59 +0000 [r260573] Leif Madsen <lmadsen@digium.com> + + * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500 + (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 + May 2010) | 1 line Minor typo pointed out by pabelanger on IRC. + ........ ................ + +2010-04-30 22:45 +0000 [r260439] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500 + (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) + | 11 lines Ensure channel state is not incorrectly set in the + case of a very early answer. The needringing bit was being read + in dahdi_read after answering thereby setting the state to + ringing from up. This clears needringing upon answering so that + is no longer possible. (closes issue #17067) Reported by: tzafrir + Patches: needringing.diff uploaded by tzafrir (license 46) + ........ ................ + +2010-04-30 20:14 +0000 [r260347] Mark Michelson <mmichelson@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 260346 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500 + (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr + 2010) | 18 lines Fix potential crash from race condition due to + accessing channel data without the channel locked. In + res_musiconhold.c, there are several places where a channel's + stream's existence is checked prior to calling ast_closestream on + it. The issue here is that in several cases, the channel was not + locked while checking the stream. The result was that if two + threads checked the state of the channel's stream at + approximately the same time, then there could be a situation + where both threads attempt to call ast_closestream on the + channel's stream. The result here is that the refcount for the + stream would go below 0, resulting in a crash. I have added + proper channel locking to res_musiconhold.c to ensure that we do + not try to check chan->stream without the channel locked. A + Digium customer has been using this patch for several weeks and + has not had any crashes since applying the patch. ABE-2147 + ........ ................ + +2010-04-29 22:57 +0000 [r260232] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500 + (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) + | 26 lines DTMF CallerID detection problems. The code handling + DTMF CallerID drops digits on long CallerID numbers and may + timeout waiting for the first ring with shorter numbers. The DTMF + emulation mode was not turned off when processing DTMF CallerID. + When the emulation code gets behind in processing the DTMF digits + it can skip a digit. For shorter numbers, the timeout may have + been too short. I increased it from 2 seconds to 4 seconds. Four + seconds is a typical time between rings for many countries. + (closes issue #16460) Reported by: sum Patches: issue16460.patch + uploaded by rmudgett (license 664) issue16460_v1.6.2.patch + uploaded by rmudgett (license 664) Tested by: sum, rmudgett + Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA + AST-334 JIRA SWP-901 ........ ................ + +2010-04-29 18:17 +0000 [r260152] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 260148 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29 + Apr 2010) | 2 lines Pattern match fail. ........ + +2010-04-29 15:39 +0000 [r260053] David Vossel <dvossel@digium.com> + + * /, include/asterisk/audiohook.h, main/audiohook.c: Merged + revisions 260050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010) + | 21 lines Merged revisions 260049 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) + | 14 lines Fixes crash in audiohook_write_list The middle_frame + in the audiohook_write_list function was being freed if a + audiohook manipulator returned a failure. This is incorrect + logic. This patch resolves this and adds detailed descriptions of + how this function should work and why manipulator failures must + be ignored. (closes issue #17052) Reported by: dvossel Tested by: + dvossel (closes issue #16196) Reported by: atis Review: + https://reviewboard.asterisk.org/r/623/ ........ ................ + +2010-04-28 22:37 +0000 [r259960] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 | + mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11 + lines Don't override peer context with domain context. (closes + issue #17040) Reported by: pprindeville Patches: + asterisk-1.6-bugid17040.patch uploaded by pprindeville (license + 347) Tested by: pprindeville Review: + https://reviewboard.asterisk.org/r/565/ ........ + +2010-04-28 21:35 +0000 [r259936] David Vossel <dvossel@digium.com> + + * main/channel.c, channels/chan_local.c, /: Merged revisions 259870 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500 + (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) + | 33 lines resolves deadlocks in chan_local Issue_1. In the + local_hangup() 3 locks must be held at the same time... pvt, + pvt->chan, and pvt->owner. Proper deadlock avoidance is done when + the channel to hangup is the outbound chan_local channel, but + when it is not the outbound channel we have an issue... We + attempt to do deadlock avoidance only on the tech pvt, when both + the tech pvt and the pvt->owner are locked coming into that loop. + By never giving up the pvt->owner channel deadlock avoidance is + not entirely possible. This patch resolves that by doing deadlock + avoidance on both the pvt->owner and the pvt when trying to get + the pvt->chan lock. Issue_2. ast_prod() is used in + ast_activate_generator() to queue a frame on the channel and make + the channel's read function get called. This function is used in + ast_activate_generator() while the channel is locked, which + mean's the channel will have a lock both from the generator code + and the frame_queue code by the time it gets to chan_local.c's + local_queue_frame code... local_queue_frame contains some of the + same crazy deadlock avoidance that local_hangup requires, and + this recursive lock prevents that deadlock avoidance from + happening correctly. This patch removes ast_prod() from the + channel lock so only one lock is held during the + local_queue_frame function. (closes issue #17185) Reported by: + schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel + (license 671) issue_17185_v2.diff uploaded by dvossel (license + 671) Tested by: schmoozecom, GameGamer43 Review: + https://reviewboard.asterisk.org/r/631/ ........ ................ + +2010-04-28 21:10 +0000 [r259856] Leif Madsen <lmadsen@digium.com> + + * config.guess: Merged revisions 259853 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010) + | 14 lines Merged revisions 259852 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010) + | 6 lines Update config.guess. Updating config.guess because + after installing Ubuntu Server 9.10 and running all the update + scripts, running ./configure would not continue because it was + unable to determine what kind of system I had. After updating + config.guess things started working again. ........ + ................ + +2010-04-28 20:33 +0000 [r259770-259849] Jason Parker <jparker@digium.com> + + * /, configure, configure.ac: Merged revisions 259848 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259848 | qwell | 2010-04-28 15:32:14 -0500 + (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr + 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so + systems without install can use install-sh from our source dir. + ........ ................ + + * /, makeopts.in: Merged revisions 259837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) | + 9 lines Merged revisions 259833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) | + 1 line Missed this when removing $ID ........ ................ + + * Makefile, /, configure, configure.ac: Merged revisions 259760 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259760 | qwell | 2010-04-28 14:19:54 -0500 + (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) | + 7 lines Remove usage of `id` since it isn't useful and was + causing breakge. Solaris `id` doesn't support the -u argument. + Instead of figuring out how to fix this to work on Solaris, I + decided to check why it was necessary and where else it was used. + It was only used in one place, and it hasn't been needed for a + very long time (I question whether it was ever needed). ........ + ................ + +2010-04-28 17:19 +0000 [r259676] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500 + (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010) + | 4 lines Do not play goodbye prompt after timeout of message + review. ABE-2124 ........ ................ + +2010-04-27 22:28 +0000 [r259591] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500 + (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) + | 11 lines DAHDI "WARNING" message is confusing and vague + "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed + failed: Success" Changed the warning to "Failed to decode + CallerID on channel 'name'". The message before it is likely more + specific about why the CallerID decode failed. SWP-501 AST-283 + ........ ................ + +2010-04-27 21:51 +0000 [r259530] Leif Madsen <lmadsen@digium.com> + + * sounds/Makefile: Merged revisions 259527 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010) + | 23 lines Merged revisions 259526 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010) + | 15 lines Update sounds files. * Add additional sounds prompts + for say_enumeration * Update the English conference sounds + prompts so they are better quality and all sound more consistent + * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files + to include all present sound files Both core (en, fr, es) and + extra (en, fr) sounds files have been updated. (closes issue + #16200) Reported by: murf (closes issue #17137) Reported by: + lmadsen ........ ................ + +2010-04-27 21:22 +0000 [r259354-259466] Jason Parker <jparker@digium.com> + + * /, main/editline/configure, main/editline/Makefile.in, + main/editline/configure.in: Merged revisions 259439 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) | + 5 lines Add gar to the check for AR for those silly OSes + (Solaris) that don't have ar. autoconf2.13 couldn't handle + AC_PROG_GREP, so I removed it. This is fine, since we don't need + to use anything that the configure script doesn't. ........ + + * /, configure, configure.ac: Merged revisions 259353 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r259353 | qwell | 2010-04-27 14:31:55 -0500 + (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) | + 5 lines Support the silly OSes that don't have ar and strip. + Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't + specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to + AC_CHECK_TOOLS. ........ ................ + +2010-04-27 18:41 +0000 [r259308] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged + revisions 259307 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010) + | 21 lines Merged revisions 259270 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) + | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue + #7321 implements a new chan_dahdi configuration option. However, + a change mentioned in the issue was never implemented. This is + the change that will allow the feature to work. I added a note to + chan_dahdi.conf.sample about the feature. (closes issue #17143) + Reported by: djensen99 Patches: diff.txt uploaded by djensen99 + (license NA) (One line change) Tested by: djensen99 ........ + ................ + +2010-04-26 21:47 +0000 [r259036-259106] Mark Michelson <mmichelson@digium.com> + + * main/channel.c, /: Merged revisions 259105 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr + 2010) | 9 lines Merged revisions 259104 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr + 2010) | 3 lines Let compilation succeed warning-free when + DONT_OPTIMIZE is turned off. ........ ................ + + * main/channel.c, /: Merged revisions 259023 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr + 2010) | 19 lines Merged revisions 259018 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr + 2010) | 13 lines Prevent Newchannel manager events for dummy + channels. No Newchannel manager event will be fired for channels + that are allocated to not match a registered technology type. + Thus bogus channels allocated solely for variable substitution or + CDR operations do not result in a Newchannel event. (closes issue + #16957) Reported by: atis Review: + https://reviewboard.asterisk.org/r/601 ........ ................ + +2010-04-25 18:14 +0000 [r258777] Tilghman Lesher <tlesher@digium.com> + + * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010) + | 13 lines Merged revisions 258775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) + | 6 lines When StopMonitor is called, ensure that it will not be + restarted by a channel event. (closes issue #16590) Reported by: + kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm + (license 888) ........ ................ + +2010-04-22 22:25 +0000 [r258711] Matthew Nicholson <mnicholson@digium.com> + + * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions + 258671,258675 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr + 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May + 2009) | 8 lines Set the proper disposition on originated calls. + (closes issue #14167) Reported by: jpt Patches: + call-file-missing-cdr2.diff uploaded by mnicholson (license 96) + Tested by: dlotina, rmartinez, mnicholson ........ r258670 | + mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11 + lines Fix broken CDR behavior. This change allows a CDR record + previously marked with disposition ANSWERED to be set as BUSY or + NO ANSWER. Additionally this change partially reverts r235635 and + does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated + from ast_call(). To preserve proper CDR behavior, the + AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in + ast_bridge_call(). (closes issue #16797) Reported by: + VarnishedOtter Tested by: mnicholson ........ (closes issue + #16222) Reported by: telles Tested by: mnicholson + ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500 + (Thu, 22 Apr 2010) | 2 lines Fix previous commit. + ................ + +2010-04-21 22:09 +0000 [r258434] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500 + (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) + | 8 lines Fix looping forever when no input received in certain + voicemail menu scenarios. Specifically, prompting for an + extension (when leaving or forwarding a message) or when + prompting for a digit (when saving a message or changing + folders). ABE-2122 SWP-1268 ........ ................ + +2010-04-20 17:43 +0000 [r258104] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500 + (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) + | 11 lines Play correct prompt when voicemail store failure + occurs after attempted forward. If a user's mailbox was full and + a message was attempted to be forwarded to said box, warnings on + the console would indicate failure. However, the played prompt + was that of success (vm-msgsaved). Now storage failure is taken + into account and the correct prompt (vm-mailboxfull) is played + when appropriate. ABE-2123 SWP-1262 ........ ................ + +2010-04-19 18:06 +0000 [r257811] Terry Wilson <twilson@digium.com> + + * /, main/features.c: Merged revisions 257810 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 | + twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines + Fix incomplete CDR merge from r195881 Because res/res_features.c + was removed and main/cdr.c added, these changes didn't make it to + trunk and the 1.6.x branches ........ + +2010-04-18 17:27 +0000 [r257769] Tilghman Lesher <tlesher@digium.com> + + * configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18 + Apr 2010) | 2 lines Removing unused configuration parameters + ........ + +2010-04-16 21:28 +0000 [r257738] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> + + * apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500 + (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) + | 21 lines Make the mixmonitor thread process audio frames faster + Mantis issue 17078 reports MixMonitor recordings have shorter + durations than the call duration. This was because the mixmonitor + thread was not processing frames from the audiohook fast enough. + The mixmonitor thread would slowly fall behind the most recent + audio frame and when the channel hangs up, the mixmonitor thread + would exit without processing the same number of frames as the + channel; leaving the mixmonitor recording shorter than actual + call duration. This revision fixes this issue by moving the + ast_audiohook_trigger_wait() and the subsequent audiohook.status + check into the block where the ast_audiohook_read_frame() + function returns NULL. (closes issue #17078) Reported by: + geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license + 733) Tested by: dhubbard, geoff2010 Review: + https://reviewboard.asterisk.org/r/611/ ........ ................ + +2010-04-15 21:33 +0000 [r257507-257592] Tilghman Lesher <tlesher@digium.com> + + * include/asterisk/app.h, /, main/app.c: Merged revisions 257560 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500 + (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) + | 6 lines Allow application options with arguments to contain + parentheses, through a variety of escaping techniques. Fixes + SWP-1194 (ABE-2143). Review: + https://reviewboard.asterisk.org/r/604/ ........ ................ + + * /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010) + | 20 lines Merged revisions 257467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) + | 13 lines Don't recreate peer, when responding to a repeated + deregistration attempt. When a reply to a deregistration is lost + in transmit, the client retries the deregistration. Previously, + this would cause a realtime/autocreate peer to be loaded back + into memory, after it had already been correctly purged. Instead, + we just want to resend the reply without loading the peer. + (closes issue #16908) Reported by: kkm Patches: + 20100412__issue16908.diff.txt uploaded by tilghman (license 14) + Tested by: kkm ........ ................ + +2010-04-15 19:43 +0000 [r257346-257430] Leif Madsen <lmadsen@digium.com> + + * doc/backtrace.txt: Merged revisions 257427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010) + | 21 lines Merged revisions 257426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) + | 13 lines Update backtrace.txt documentation. Update the + backtrace.txt documentation so it conforms to the same layout as + other documents we've been working on recently. Additionally, add + a bunch of new information about gathering backtraces for crashes + and deadlocks, along with ways of verifying your file before + uploading it. Create a couple of one line commands for people to + generate the files we need. (closes issue #17190) Reported by: + lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen + (license 10) Tested by: lmadsen, pabelanger ........ + ................ + + * doc/backtrace.txt: Merged revisions 257343 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010) + | 9 lines Merged revisions 257342 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) + | 1 line Update address of the bug tracker. ........ + ................ + +2010-04-14 22:59 +0000 [r257263] Tilghman Lesher <tlesher@digium.com> + + * configs/features.conf.sample, /, main/features.c: Merged + revisions 257262 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 | + tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15 + lines Yet another issue where the conversion of the application + delimiter to comma caused an issue. Application arguments within + the feature map could possibly contain a comma, which conflicts + with the syntax of the features.conf configuration file. This + patch allows the argument to be wrapped in parentheses or quoted, + to allow the application arguments to be interpreted as a single + configuration parameter. (closes issue #16646) Reported by: + pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by + tilghman (license 14) Tested by: tilghman Review: + https://reviewboard.asterisk.org/r/547/ ........ + +2010-04-13 19:44 +0000 [r257218] Matthew Nicholson <mnicholson@digium.com> + + * main/manager.c, /, configs/manager.conf.sample: Merged revisions + 257146 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr + 2010) | 16 lines Merged revisions 257070 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr + 2010) | 9 lines Add an option to restore past broken behavor of + the Events manager action Before r238915, certain values for the + EventMask parameter of the Events action would result in no + response being returned. This patch adds an option to restore + that broken behavior. Also while fixing this bug I discovered + that passing an empty EventMasks parameter would also result in + no response being returned, this has been fixed as well while + being preserved when the broken behavior is requested. (closes + issue #17023) Reported by: nblasgen Review: + https://reviewboard.asterisk.org/r/602/ ........ ................ + +2010-04-13 19:20 +0000 [r257066-257206] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 | + tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10 + lines Also unref the pvt when we delete the provisional keepalive + job. (closes issue #16774) Reported by: kowalma Patches: + 20100315__issue16774.diff.txt uploaded by tilghman (license 14) + Tested by: falves11, jamicque Review: + https://reviewboard.asterisk.org/r/591/ ........ + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010) + | 8 lines Ensure that we can have commas within cdr values. + (closes issue #17001) Reported by: snuffy Patches: + 20100412__issue17001.diff.txt uploaded by tilghman (license 14) + Tested by: snuffy ........ + +2010-04-12 17:31 +0000 [r256904] Leif Madsen <lmadsen@digium.com> + + * doc/HOWTO_collect_debug_information.txt (added): Merged revisions + 256901 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010) + | 23 lines Merged revisions 256900 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) + | 15 lines Add How-To document on collecting debugging info for + issues.asterisk.org Paul Belanger has been helping a lot with bug + tracking recently and created this document that we can now point + to when additional debugging information is required. This + document will help those filing issues to know how to get the + information required when filing their issues. This will make + things easier on the developers. Initial text and changes by + pabelanger. Tweaks and editing by myself. (closes issue #17159) + Reported by: pabelanger Patches: + HOWTO_collect_debug_information.txt.patch uploaded by lmadsen + (license 10) Tested by: tzafrir, pabelanger, lmadsen ........ + ................ + +2010-04-06 19:40 +0000 [r256371] Tilghman Lesher <tlesher@digium.com> + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + include/asterisk/lock.h: Merged revisions 256370 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010) + | 2 lines Mac OS X does not support comparing a mutex to its + initializer. Create a test for this. ........ + +2010-04-06 18:07 +0000 [r256266-256364] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c: Fix malformed if test. Regression of issue + 15883. Converted if statement to a switch statement for clarity. + + * channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500 + (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) + | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by + PRI lock. SWP-1231 ABE-2163 ........ ................ + +2010-04-05 15:16 +0000 [r256164] Leif Madsen <lmadsen@digium.com> + + * /, doc/tex/localchannel.tex: Merged revisions 256161 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010) + | 1 line Fix for localchannel.tex to allow PDFs to be generated + again. ........ + +2010-04-05 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.0.27-rc1 Released + +2010-04-05 15:16 +0000 [r256164] Leif Madsen <lmadsen@digium.com> + + * /, doc/tex/localchannel.tex: Merged revisions 256161 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010) + | 1 line Fix for localchannel.tex to allow PDFs to be generated + again. ........ + +2010-04-02 23:47 +0000 [r256011-256016] Russell Bryant <russell@digium.com> + + * channels/chan_local.c, /: Merged revisions 256015 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r256015 | russell | 2010-04-02 18:46:45 -0500 + (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) + | 9 lines Resolve a deadlock that occurs due to a pointless call + to ast_bridged_channel() (closes issue #16840) Reported by: + bzing2 Patches: patch.txt uploaded by bzing2 (license 902) + issue_16840.rev1.diff uploaded by russell (license 2) Tested by: + bzing2, russell ........ ................ + + * main/channel.c, /: Merged revisions 256010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010) + | 9 lines Merged revisions 256009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010) + | 2 lines Remove extremely verbose debug message. ........ + ................ + +2010-04-02 20:20 +0000 [r255953] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 255952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 | + tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines + Pass the PID of the Asterisk process, not the PID of the canary. + (closes issue #17065) Reported by: globalnetinc Patches: + astcanary.patch uploaded by makoto (license 38) Tested by: frawd, + globalnetinc ........ + +2010-04-01 18:21 +0000 [r255674-255813] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010) + | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue + #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt + uploaded by tilghman (license 14) ........ + + * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500 + (Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) + | 15 lines Ensure line terminators in email are consistent. Fixes + an issue with certain Mail Transport Agents, where attachments + are not interpreted correctly. (closes issue #16557) Reported by: + jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by + tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt + uploaded by tilghman (license 14) + 20100308__issue16557__trunk.diff.txt uploaded by tilghman + (license 14) Tested by: ebroad, zktech Reviewboard: + https://reviewboard.asterisk.org/r/544/ ........ ................ + +2010-03-31 17:54 +0000 [r255507] Leif Madsen <lmadsen@digium.com> + + * apps/app_dial.c, configs/sip.conf.sample: Merged revisions 255504 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31 + Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T' + can be used. (closes issue #17021) Reported by: kovzol Tested by: + lmadsen, kovzol, davidw, ebroad ........ + +2010-03-30 20:57 +0000 [r255324-255411] Russell Bryant <russell@digium.com> + + * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r255410 | russell | 2010-03-30 15:56:26 -0500 + (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 + Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does + not start. ........ ................ + + * /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010) + | 9 lines Merged revisions 255322 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010) + | 2 lines Don't make Asterisk not start if pbx_dundi fails to + initialize. ........ ................ + +2010-03-26 19:24 +0000 [r255054] Leif Madsen <lmadsen@digium.com> + + * /, configs/sip.conf.sample: Merged revisions 255021 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010) + | 8 lines Update confusing documentation for tlsbindaddr. Update + some confusing documentation for the tlsbindaddr option in + sip.conf.sample. Point at a link instead which has better + documentation. (closes issue #17054) Reported by: klaus3000 + ........ + +2010-03-25 20:42 +0000 [r254803] Jason Parker <jparker@digium.com> + + * utils/Makefile, /: Merged revisions 254802 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) | + 9 lines Merged revisions 254800 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) | + 1 line Don't remove local copies of utils in uninstall. ........ + ................ + +2010-03-25 20:09 +0000 [r254719] Russell Bryant <russell@digium.com> + + * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010) + | 2 lines chan_usbradio depends on alsa. ........ + +2010-03-25 19:59 +0000 [r254716] Jason Parker <jparker@digium.com> + + * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS + issue with out-of-tree modules. Take 2, without ABI breakage this + time. Review: https://reviewboard.asterisk.org/r/588/ + +2010-03-25 17:45 +0000 [r254549-254554] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500 + (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar + 2010) | 5 lines Add doxygen for acl.h Review: + https://reviewboard.asterisk.org/r/528 ........ ................ + + * channels/chan_sip.c: Undo unnecessary commit. Sean Bright beat me + to the punch on this one. + + * channels/chan_sip.c: Fix potential use of uninitialized value. + +2010-03-25 17:19 +0000 [r254546] Sean Bright <sean@malleable.com> + + * channels/chan_sip.c: Initialize stream to avoid a compilation + error. + +2010-03-25 17:05 +0000 [r254540] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Fix crashes resulting from reading + non-existent RTP streams. Specifically, when using the CHANNEL + dialplan function, it was possible to crash Asterisk by trying to + get the rtpdest of a stream type that is not in use by the + channel. This commit fixes that issue. + +2010-03-25 17:02 +0000 [r254539] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/safe_asterisk, /: Make safe_asterisk work on + dash/sh/bash etc. Merged from the change to trunk via issue + #13111. For some reason the changes there were only done on + trunk, and thus were available for 1.6.1 and 1.6.2 when they were + branched. Because this change is available on both 1.6.1 and + 1.6.2, it makes sense to allow it on the 1.6.0 branch as well. + (closes issue #17094) Reported by: stuarth Much thanks to + Tilghman and Sean Bright for the help on this merge. Merged + revisions 135061 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135061 | + mvanbaak | 2008-08-01 07:17:33 -0500 (Fri, 01 Aug 2008) | 8 lines + Make safe_asterisk work on dash/sh/bash etc. (closes issue + #13111) Reported by: pabelanger Patches: + 2008071901_issue13111_safe_asterisk.diff uploaded by mvanbaak + (license 7) Tested by: mvanbaak, pabelanger ........ + +2010-03-25 16:57 +0000 [r254538] Sean Bright <sean@malleable.com> + + * /: Unblock r135061 + +2010-03-25 16:19 +0000 [r254466] Terry Wilson <twilson@digium.com> + + * /, main/file.c: Merged revisions 254453 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010) + | 9 lines Merged revisions 254451 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) + | 2 lines Handle new SRCCHANGE control message here too ........ + ................ + +2010-03-25 16:11 +0000 [r254455] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500 + (Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar + 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection. + Here is a copy and paste of the details from my request on + reviewboard that dealt with these changes: Fix 1. The first + change in place is to fix Mantis issue 15811, which deals with a + situation where Asterisk will incorrectly interpret out of order + RFC2833 frames as duplicate DTMF digits. For instance, we would + receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: + DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 + seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch + when we received the frame with seqno 5, we would interpret this + as a new DTMF 1. With this patch, we will check the seqno of the + incoming digit and not process the frame if the seqno is lower + than the last recorded seqno. Note that we do not record the + seqno of the dropped DTMF frame for future processing. While the + above situation is what was designed to be fixed, the patch is + written in such a way that the following would also be fixed too: + seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) + seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno + 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In + this second situation, the beginning of the DTMF 2 arrives before + the final end frame of the DTMF 1. With the patch, seqno 12 is no + processed and thus we properly interpret the DTMF. Fix 2. The + second change in place is to fix an issue like the following: + seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet + lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end) + *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had + code in place that was supposed to properly end the previously + unended DTMF 1. The problem was that the code was essentially a + no-op. The code would set up an end frame for the DTMF 1 but + would immediately overwrite the frame with the begin for DTMF 2. + I changed process_dtmf_rfc2833() so that instead of returning a + single frame, it is given as an output parameter a list of + frames. Each frame that needs to be returned is appended to this + list. Fix 3. The final change is a minor one where an + AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco + DTMF or an RFC 3389 frame and no frame was returned, then we + would return &ast_null_frame. The problem is that earlier in the + function, we may have generated an AST_CONTROL_SRCCHANGE frame + and put it in the list of frames we wish to return. This frame + would be lost in such a case. The patch fixes this problem + ........ ................ + +2010-03-25 15:22 +0000 [r254449] Leif Madsen <lmadsen@digium.com> + + * /, res/res_agi.c: Merged revisions 254446 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 | + lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines + handle_speechset has 4 arguments. Update code to reflect that + handle_speechset has 4 arguments. (closes issue #17093) Reported + by: gpatri Patches: res_agi.patch uploaded by gpatri (license + 1014) Tested by: pabelanger, mmichelson ........ + +2010-03-24 17:17 +0000 [r254061-254278] Jeff Peeler <jpeeler@digium.com> + + * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010) + | 78 lines Merged revisions 254235 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) + | 72 lines Ensure that monitor recordings are written to the + correct location (again) This is an extension to 248860. As such + the dialplan test has been extended: ; non absolute path, not + combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test) + exten => 5040, n, dial(sip/5001) ; absolute path, not combined + exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten => + 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1, + monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ; + combined: changemonitor from non absolute to no path (leaves + tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m) + exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n, + dial(sip/5001) ; combined: changemonitor from no path to non + absolute path exten => 5044, 1, monitor(wav,monitor_test6,m) + exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this + wasn't possible before exten => 5044, n, dial(sip/5001) ; non + absolute path, combined exten => 5045, 1, + monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n, + dial(sip/5001) ; absolute path, combined exten => 5046, 1, + monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n, + dial(sip/5001) ; no path, combined exten => 5047, 1, + monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ; + combined: changemonitor from non absolute to absolute (leaves + tmp/jeff) exten => 5048, 1, + monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n, + changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n, + dial(sip/5001) ; combined: changemonitor from absolute to non + absolute (leaves /tmp/jeff) exten => 5049, 1, + monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n, + changemonitor(tmp/jeff/monitor_test14) exten => 5049, n, + dial(sip/5001) ; combined: changemonitor from no path to absolute + exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n, + changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n, + dial(sip/5001) ; combined: changemonitor from absolute to no path + (leaves /tmp/jeff) exten => 5051, 1, + monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n, + changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ; + not combined: changemonitor from non absolute to no path (leaves + tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19) + exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n, + dial(sip/5001) ; not combined: changemonitor from no path to non + absolute exten => 5053, 1, monitor(wav,monitor_test21) exten => + 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n, + dial(sip/5001) ; not combined: changemonitor from non absolute to + absolute (leaves tmp/jeff) exten => 5054, 1, + monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n, + changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n, + dial(sip/5001) ; not combined: changemonitor from absolute to non + absolute (leaves /tmp/jeff) exten => 5055, 1, + monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n, + changemonitor(tmp/jeff/monitor_test25) exten => 5055, n, + dial(sip/5001) ; not combined: changemonitor from no path to + absolute exten => 5056, 1, monitor(wav,monitor_test26) exten => + 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056, + n, dial(sip/5001) ; not combined: changemonitor from absolute to + no path (leaves /tmp/jeff) exten => 5057, 1, + monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n, + changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001) + ........ ................ + + * main/channel.c, /: Merged revisions 254050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 | + jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines + Exit native bridging early for greater timing accuracy with + warnings This changes native bridging to break one millisecond + early so that the more accurate timeval calculations done in the + generic bridge can be performed using the bridge config. + Currently the time between exiting native bridging slightly late + can sometimes cause a large enough discrepancy for warnings to be + missed. For the record, 1.4 does not attempt to native bridge at + all when warnings are enabled. (closes issue #15815) Reported by: + adomjan Review: https://reviewboard.asterisk.org/r/577/ ........ + +2010-03-23 20:52 +0000 [r254044] Tzafrir Cohen <tzafrir.cohen@xorcom.com> + + * tests/Makefile, /: Merged revisions 254001 via svnmerge from + http://svn.digium.com/svn/asterisk/trunk ........ r254001 | + tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines + Change the name of the category 'TEST' to match the name of the + subdir ........ + +2010-03-22 19:57 +0000 [r253803] Matthew Nicholson <mnicholson@digium.com> + + * /, main/features.c: Merged revisions 253800 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar + 2010) | 11 lines Merged revisions 253799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar + 2010) | 4 lines Unconditionally copy the caller's account code to + the called party. (related to issue #16331) ........ + ................ + +2010-03-20 18:29 +0000 [r253625-253630] Russell Bryant <russell@digium.com> + + * main/sched.c, main/manager.c, main/features.c, + apps/app_waituntil.c, main/logger.c: Resolve 1.6.0 compilation + issues on FreeBSD. + + * apps/app_dial.c, channels/chan_dahdi.c, main/tcptls.c, /, + main/features.c, pbx/pbx_dundi.c, cdr/cdr_pgsql.c, + main/stdtime/localtime.c, apps/app_followme.c: Merged revisions + 253536-253538,253540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r253536 | + russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010) | 4 lines + Use SHRT_MAX instead of MAXSHORT. These changes fix build issues + I had with this module on FreeBSD. ........ r253537 | russell | + 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve a + compiler warning on FreeBSD. ........ r253538 | russell | + 2010-03-20 06:43:08 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve + compiler warnings on FreeBSD. ........ r253540 | russell | + 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve + more compiler warnings on FreeBSD. ........ + + * main/utils.c: Resolve compiler warnings on FreeBSD. + +2010-03-18 17:56 +0000 [r253259-253348] Leif Madsen <lmadsen@digium.com> + + * apps/app_userevent.c: Slightly different fix for UserEvent docs + update. (issue #16961) + + * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010) + | 9 lines Update to new Local channel documentation. Add same + changes as commit to 1.4, but convert to TeX. (issue #16963) + Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz + (license 834) ........ + +2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson@digium.com> + + * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_mgcp.c, channels/chan_sip.c, + include/asterisk/rtp.h: Revert API change in release branches + This re-renames ast_rtp_update_source to ast_rtp_new_source + +2010-03-17 00:31 +0000 [r253031] Leif Madsen <lmadsen@digium.com> + + * /, configs/say.conf.sample: Merged revisions 253028 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500 + (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010) + | 6 lines Add french snipset to say.conf. Add the french snipset + to say.conf. (Closes issue #15799) ........ ................ + +2010-03-16 23:54 +0000 [r252979] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 | + tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines + Mask out previous arguments on each nested invocation of Gosub. + (closes issue #16758) Reported by: wdoekes Patches: + 20100316__issue16758.diff.txt uploaded by tilghman (license 14) + Review: https://reviewboard.asterisk.org/r/561/ ........ + +2010-03-16 19:01 +0000 [r252768] Russell Bryant <russell@digium.com> + + * utils/Makefile, /: Merged revisions 252767 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010) + | 13 lines Merged revisions 252766 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010) + | 6 lines Don't treat warnings as errors for muted. muted + supports OS X, but uses functions marked as deprecated in 10.6. + However, the functions are still supported, so just ignore the + warnings for now and allow the build to proceed. ........ + ................ + +2010-03-16 18:49 +0000 [r252765] Leif Madsen <lmadsen@digium.com> + + * /, configs/extensions.ael.sample: Merged revisions 252762 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500 + (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) + | 7 lines Additional extensions.ael global variable fixes. Fixing + up a couple more overlapping global variable namespaces shared + with extensions.conf.sample. Also noticed a few of the lines that + were commented out didn't have the closing semi-colon so I added + that as well. (issue #17035) ........ ................ + +2010-03-15 21:59 +0000 [r252624] Sean Bright <sean@malleable.com> + + * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 | + seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4 + lines Resolve a crash in SLATrunk when the specified trunk + doesn't exist. Reported by philipp64 in #asterisk-dev. ........ + +2010-03-15 21:53 +0000 [r252620] Tilghman Lesher <tlesher@digium.com> + + * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions + 252619 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010) + | 9 lines Merged revisions 252617 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010) + | 2 lines Uh, yeah. Umask. I'm stupid. ........ ................ + +2010-03-15 20:54 +0000 [r252537] Leif Madsen <lmadsen@digium.com> + + * configs/extensions.ael.sample: Merged revisions 252534 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500 + (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) + | 7 lines Update extensions.ael file to not overlap + extensions.conf. Updated the extensions.ael file so the global + variables don't overlap those that we have in extensions.conf + (sample files). This way unexpected things won't happed hopefully + if both pbx_ael and res_config are loaded. (closes issue #17035) + Reported by: pprindeville ........ ................ + +2010-03-15 01:37 +0000 [r252363] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, Makefile, + contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged + revisions 252362 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010) + | 11 lines Merged revisions 252361 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010) + | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard: + https://reviewboard.asterisk.org/r/551/ ........ ................ + +2010-03-14 17:45 +0000 [r252315] Sean Bright <sean@malleable.com> + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar + 2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They + added a sqlite3_log() function which was conflicting with our + function names. (closes issue #17017) Reported by: alephlg + ........ + +2010-03-13 00:30 +0000 [r252134-252176] Terry Wilson <twilson@digium.com> + + * main/rtp.c: Remove unused field + + * main/rtp.c, channels/chan_mgcp.c, main/channel.c, + channels/chan_sip.c, channels/chan_skinny.c, + include/asterisk/rtp.h, channels/chan_h323.c, + configs/sip.conf.sample, include/asterisk/frame.h: Merged + revisions 252089 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | + twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines + Only change the RTP ssrc when we see that it has changed This + change basically reverts the change reviewed in + https://reviewboard.asterisk.org/r/374/ and instead limits the + updating of the RTP synchronization source to only those times + when we detect that the other side of the conversation has + changed the ssrc. The problem is that SRCUPDATE control frames + are sent many times where we don't want a new ssrc, including + whenever Asterisk has to send DTMF in a normal bridge. This is + also not the first time that this mistake has been made. The + initial implementation of the ast_rtp_new_source function also + changed the ssrc--and then it was removed because of this same + issue. Then, we put it back in again to fix a different issue. + This patch attempts to only change the ssrc when we see that the + other side of the conversation has changed the ssrc. It also + renames some functions to make their purpose more clear. Review: + https://reviewboard.asterisk.org/r/540/ ........ + +2010-03-12 19:53 +0000 [r251995] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c: Forward declaring dahdi_pri was already + done. + +2010-03-12 19:49 +0000 [r251992] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010) + | 8 lines Don't override a user option with the global option. + (closes issue #16849) Reported by: ip-rob Patches: + 20100311__issue16849.diff.txt uploaded by tilghman (license 14) + Tested by: ip-rob ........ + +2010-03-12 19:42 +0000 [r251988] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 251987 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r251987 | rmudgett | 2010-03-12 13:40:16 -0600 + (Fri, 12 Mar 2010) | 9 lines Merged revisions 251986 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 + Mar 2010) | 1 line Make chan_dahdi wakeup_sub() prototype not + conditional. ........ ................ + +2010-03-11 21:08 +0000 [r251882-251885] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_exec.c: Merged revisions 251884 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 | + tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines + Because ExecIf needs to reprocess arguments, it's best if we + don't remove quotes during parsing. (closes issue #16905) + Reported by: ip-rob Patches: 20100303__issue16905.diff.txt + uploaded by tilghman (license 14) Tested by: ip-rob ........ + + * /, apps/app_system.c: Merged revisions 251877 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 | + tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines + If the argument to the system application is quoted, ensure we + remove the quotes before trying to execute. (closes issue #16842) + Reported by: ip-rob Patches: 20100310__issue16842.diff.txt + uploaded by tilghman (license 14) Tested by: ip-rob ........ + +2010-03-11 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.0.26 released + +2010-03-04 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.0.26-rc1 released + +2010-03-03 21:27 +0000 [r250612] Leif Madsen <lmadsen@digium.com> + + * doc/tex/localchannel.tex: Merged revisions 250609 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010) + | 11 lines Update existing Local channel documentation. A + complete re-write of the Local channel documentation has been + performed, with the existing information from localchannel.txt + and localchannel.tex merged in. (closes issue #16637) Reported + by: kobaz Patches: localchannel.tex uploaded by lmadsen (license + 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: + lmadsen, jsmith, mmichelson ........ + +2010-03-03 19:07 +0000 [r250482] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600 + (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) + | 15 lines Make sure to clear red alarm after polarity reversal. + From the issue: The automatic overnight line tests (or manual + ones) used on UK (BT) lines causes a red alarm on a dahdi / + TDM400P connected channel. This is because the line uses voltage + tests (battery loss) and polarity reversal. The polarity reversal + causes chan_dahdi to initiate v23 CallerID processing but during + this the event DAHDI_EVENT_NOALARM is ignored so that the alarm + is never cleared. (closes issue #14163) Reported by: jedi98 + Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license + 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ + ................ + +2010-03-03 18:06 +0000 [r250265-250398] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 250395 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600 + (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) + | 16 lines fixes problem with duplicate TXREQ packets When + Asterisk receives an IAX2 TXREQ packet, try_transfer() will call + store_by_transfercallno() to link the chan_iax2_pvt struct into + iax_transfercallno_pvts. If a duplicate TXREQ packet is received + for the same call, the pvt struct will be linked into + iax_transfercallno_pvts multiple times. This patch fixes this. + Thanks rain for debugging this and providing a patch! (closes + issue #16904) Reported by: rain Patches: + iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested + by: rain, dvossel ........ ................ + + * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 | + dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines + fixes signed to unsigned int comparision issue for FaxMaxDatagram + value. ........ + +2010-03-02 21:11 +0000 [r250040-250054] Leif Madsen <lmadsen@digium.com> + + * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010) + | 8 lines Update IMAP documentation. Update the IMAP + documentation to make it clear that storing voicemails in the + same folder as a large number of emails could potentially cause + significant slow downs when writing or retrieving voicemails. + (issue #16704) Reported by: TimeHider Tested by: lmadsen, + TimeHider ........ + + * configs/cdr.conf.sample: Merged revisions 250045 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500 + (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) + | 7 lines Update documentation to clarify purpose of unanswered + option. (closes issue #16267) Reported by: elsto Patches: + cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested + by: davidw, elsto ........ ................ + + * doc/tex/configuration.tex: Merged revisions 250037 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 Mar 2010) + | 4 lines Update documentation to not imply we support overriding + options. (closes issue #16855) Reported by: davidw ........ + +2010-03-02 19:45 +0000 [r249948] Alec L Davis <sivad.a@paradise.net.nz> + + * apps/app_echo.c: revert ability to exit echo app caused a + regression, as only supported VOICE, not VIDEO etc. (issue + #16880) + +2010-03-02 19:20 +0000 [r249907] David Vossel <dvossel@digium.com> + + * channels/chan_oss.c, channels/misdn_config.c, + include/asterisk/abstract_jb.h, configs/alsa.conf.sample, + channels/chan_jingle.c, channels/chan_usbradio.c, + channels/chan_dahdi.c, channels/chan_skinny.c, + configs/mgcp.conf.sample, main/abstract_jb.c, + channels/chan_h323.c, channels/chan_alsa.c, + configs/sip.conf.sample, channels/chan_mgcp.c, + channels/chan_unistim.c, configs/console.conf.sample, + configs/chan_dahdi.conf.sample, channels/chan_local.c, + configs/oss.conf.sample, channels/chan_sip.c, /, + configs/usbradio.conf.sample, configs/misdn.conf.sample, + channels/chan_gtalk.c, channels/chan_console.c: Merged revisions + 249893 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | + dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines + fixes adaptive jitterbuffer configuration When configuring the + adaptive jitterbuffer, the target_extra value not only could not + be set from the configuration, but was not even being set to its + proper default. This value is required in order for the adaptive + jitterbuffer to work correctly. To resolve this a config option + has been added to expose this value to the conf files, and a + default value is provided when no config specific value is + present. ........ + +2010-03-02 09:16 +0000 [r249846] Alec L Davis <sivad.a@paradise.net.nz> + + * apps/app_echo.c: fixes ability to exit echo app when called from + a ISDN channel, null frames prevent '#' exit. Now only echo back + VOICE and DTMF frames (closes issue #16880) Reported by: + alecdavis Patches: based on echo_exit_1-6-1.diff.txt uploaded by + alecdavis (license 585) Tested by: alecdavis + +2010-03-01 19:38 +0000 [r249673] Sean Bright <sean@malleable.com> + + * apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500 + (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar + 2010) | 11 lines Fix crash in app_voicemail related to message + counting. We were passing a 'struct inprocess **' and treating it + like a 'struct inprocess *' causing a segfault. (closes issue + #16921) Reported by: whardier Patches: 20100301_issue16921.patch + uploaded by seanbright (license 71) Tested by: whardier ........ + ................ + +2010-03-01 17:13 +0000 [r249539] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_local.c, /: Merged revisions 249538 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600 + (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) + | 11 lines Modify queued frames from local channels to not set + the other side to up In this case, attended transfers were broken + due to ast_feature_request_and_dial detecting the channel being + set to up before the answer frame could be read and therefore + failing to mark the channel as ready. This fix is a regression + fix for 244785, which should continue to work properly as well. + (closes issue #16816) Reported by: jamhed Tested by: jamhed, + corruptor ........ ................ + +2010-02-27 23:38 +0000 [r249364] Alec L Davis <sivad.a@paradise.net.nz> + + * channels/chan_dahdi.c: overlap receiving: automatically send CALL + PROCEEDING when dialplan starts Following Q.931 5.2.4 When the + user has determined that sufficient call information has been + received the user shall stop T302 and send CALL PROCEEDING to the + network. Previously timeouts were possible if the dialplan took a + long time to issue any response back to the network. Verified + that our local TELCO also does the same. (issue #16789) Reported + by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded + by alecdavis (license 585) Tested by: alecdavis (closes issue + #16789) + +2010-02-27 14:09 +0000 [r249236] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 249235 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500 + (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 + Feb 2010) | 1 line add a reference to the now-published IAX2 RFC + ........ ................ + +2010-02-26 17:05 +0000 [r249102] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb + 2010) | 14 lines Merged revisions 249100 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb + 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. + (closes issue #16792) Reported by: vrban Patches: t38_606.patch + uploaded by vrban (license 756) ........ ................ + +2010-02-25 23:11 +0000 [r248953] Jeff Peeler <jpeeler@digium.com> + + * res/res_monitor.c, /: Merged revisions 248952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010) + | 24 lines Merged revisions 248860 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) + | 18 lines Ensure that monitor recordings are written to the + correct location (again) This is an extension to 248757. As such + the dialplan test has been extended: exten => 5040, 1, + monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, + dial(sip/5001) exten => 5041, 1, + monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, + dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) + exten => 5042, n, dial(sip/5001) exten => 5043, 1, + monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, + changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) + exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, + changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by + design and emits a warning exten => 5044, n, dial(sip/5001) + ........ ................ + +2010-02-25 22:42 +0000 [r248947] Mark Michelson <mmichelson@digium.com> + + * /, main/acl.c: Merged revisions 248946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 | + mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5 + lines Fix incorrect ACL behavior when CIDR notation of "/0" is + used. AST-2010-003 ........ + +2010-02-25 21:24 +0000 [r248862] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 248861 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010) + | 22 lines Merged revisions 248859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) + | 15 lines Some platforms clear /var/run at boot, which makes + connecting a remote console... difficult. Previously, we only + created the default /var/run/asterisk directory at install time. + While we could create it in the init script, that would not work + for those who start asterisk manually from the command line. So + the safest thing to do is to create it as part of the Asterisk + boot process. This also changes the ownership of the directory, + because the pid and ctl files are created after we setuid/setgid. + (closes issue #16802) Reported by: Brian Patches: + 20100224__issue16802.diff.txt uploaded by tilghman (license 14) + Tested by: tzafrir ........ ................ + +2010-02-25 18:46 +0000 [r248795] Jeff Peeler <jpeeler@digium.com> + + * res/res_monitor.c, /: Merged revisions 248793 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010) + | 22 lines Merged revisions 248757 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) + | 15 lines Ensure that monitor recordings are written to the + correct location. Recordings should be placed in the monitor + directory when a non-absolute path is used. Exact dialplan used + for testing: exten => 5040, 1, + monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, + dial(sip/5001) exten => 5041, 1, + monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, + dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) + exten => 5042, n, dial(sip/5001) ABE-2101 ........ + ................ + +2010-02-24 21:23 +0000 [r248613] Tilghman Lesher <tlesher@digium.com> + + * /, main/logger.c: Merged revisions 248584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010) + | 14 lines Merged revisions 248582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) + | 7 lines Remove color code sequences from verbose messages that + go to logfiles. (closes issue #16786) Reported by: dodo Patches: + logger2.patch uploaded by dodo (license 989) Tested by: tilghman + ........ ................ + +2010-02-23 16:51 +0000 [r248400] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) + | 15 lines Merged revisions 248396 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) + | 9 lines fixes invite with replaces deadlock (closes issue + #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 + uploaded by dvossel (license 671) Tested by: pwalker, dvossel + ........ ................ + +2010-02-19 19:04 +0000 [r247933-248008] Tilghman Lesher <tlesher@digium.com> + + * main/loader.c, /, channels/chan_console.c: Merged revisions + 228798 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk (closes issue + #16470) Reported by: kjotte ........ r228798 | tilghman | + 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix + various problems detected with Valgrind. * chan_console accessed + pvts after deallocation. * The module loader did not check + usecount on shutdown, which led to chan_iax2 reading a timer that + was already unloaded. (closes issue #16062) Reported by: + alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by + tilghman (license 14) Tested by: tilghman ........ + + * main/ast_expr2f.c: Restore generated file from flex source + +2010-02-19 18:13 +0000 [r247919-247922] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600 + (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 + (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... + .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, + 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing + consistent with other channel technologies. The processing of + DTMF tones on the receiving side of an ISDN channel is + inconsistent with the way it is handled in other channels, + especially DAHDI analog. This causes DTMF tones sent from an ISDN + phone to be doubled at the connected party. We are using the + following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes + Option one is necessary because the asterisk DSP DTMF detection + is better than mISDN's internal DSP. Not as many false positives. + Option two is necessary to transmit DTMF tones end to end when + mISDN channels are connected to SIP channels with out of band + DTMF for example. The symptom is that DTMF tones sent by an ISDN + phone are doubled on the way through asterisk when two mISDN + channels are connected with a Local channel in between or if it + is bridged to an analog channel. The doubling of DTMF tones is + because DTMF is passed inband to asterisk by the mISDN channel + and passed out of band once again after the release of the DTMF + tone. Passing it inband is wrong. Neither an analog channel nor + SIP channel passes DTMF inband if configured to inband DTMF. + Analog and SIP channels filter out the DTMF tones because they + use the voice frames returned by ast_dsp_process. But chan_misdn + passes the unfiltered input voice frames instead. To overcome one + aspect of the problem, the doubling of DTMF tones when two mISDN + channels are directly bridged, someone made an 'optimization', + where in that case the DTMF tone passed out-of-band to the peer + channel is not translated to an inband tone at the transmit side. + This optimization is bad because it does not work in general. For + example, analog channels or mISDN channels when bridged through + an intermediary local channel will generate DTMF tones from + out-of-band information. Also, of course, it must not be done + when there is no inband DTMF available. This patch fixes the + issue. Now chan_misdn will filter the received inband DTMF signal + the same as other channel types. Another change included: No need + to build an extra translation path because ast_process_dsp does + it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 + ................ ................ + + * main/ast_expr2f.c: Restore fwrite() line so ast_expr2f.c can + compile. + +2010-02-18 23:15 +0000 [r247842] Tilghman Lesher <tlesher@digium.com> + + * res/res_speech.c, /: Merged revisions 247841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 | + tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines + Revert an errant part of a previous cleanup, to fix a memory + corruption issue. (closes issue #16368) Reported by: thirionjwf + Patches: res_speech.c.patch uploaded by thirionjwf (license 955) + ........ + +2010-02-18 22:45 +0000 [r247839] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c: fixes dialog ref count crash isolated to the + 1.6.0 branch (closes issue #16375) Reported by: kobaz (closes + issue #16796) Reported by: kobaz + +2010-02-18 21:47 +0000 [r247789] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 | + tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 + lines If the peer record is from realtime, it could be set to 0, + due to MySQL not representing NULL well in integer columns. NULL + means the value is not specified for the column, which normally + means the driver uses whatever is the default value. However, on + MySQL, placing a NULL in either a float or integer column results + in a retrieval of the 0 value. Hence, users get an errant error + on load. This patch suppresses that error and makes the value as + if it was not there. Note that this cannot be done in the + realtime driver, because the lack of difference between NULL and + 0 can only be intepreted correctly by the driver itself. If we + did it in the realtime driver, then it would be effectively + impossible to set any realtime field to 0, because it would act + as if the field were unspecified and possibly take on a different + value. (closes issue #16683) Reported by: wdoekes ........ + +2010-02-18 19:45 +0000 [r247655] Matthew Nicholson <mnicholson@digium.com> + + * /, main/features.c: Merged revisions 247652 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb + 2010) | 13 lines Merged revisions 247651 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb + 2010) | 6 lines Copy the calling party's account code to the + called party if they don't already have one. (closes issue + #16331) Reported by: bluefox Tested by: mnicholson ........ + ................ + +2010-02-18 16:56 +0000 [r247504-247510] Leif Madsen <lmadsen@digium.com> + + * README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500 + (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 + Feb 2010) | 1 line Add additional link to best practices document + per jsmith. ........ ................ + + * README-SERIOUSLY.bestpractices.txt (added): Merged revisions + 247503 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010) + | 18 lines Merged revisions 247502 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) + | 10 lines Add best practices documentation. (issue #16808) + Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis + Tested by: lmadsen Review: + https://reviewboard.asterisk.org/r/507/ ........ ................ + +2010-02-18 04:20 +0000 [r247424] Russell Bryant <russell@digium.com> + + * Makefile, /, sounds/Makefile: Merged revisions 247423 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r247423 | russell | 2010-02-17 22:20:11 -0600 + (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) + | 10 lines Tweak argument handling for wget in the sounds + Makefile. 1) Fix the check to see if we are using wget to not be + full of fail. The configure script populates this variable with + the absolute path to wget if it is found, so it didn't work. 2) + Allow some extra arguments to be passed in for wget. This is just + a simple change to allow our Bamboo build script to tell wget to + be quiet and not fill up our logs with download status output. + ........ ................ + +2010-02-17 21:35 +0000 [r246986-247338] Mark Michelson <mmichelson@digium.com> + + * /, main/utils.c, include/asterisk/strings.h: Merged revisions + 247335 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 | + mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20 + lines Fix two problems in ast_str functions found while writing a + unit test. 1. The documentation for ast_str_set and + ast_str_append state that the max_len parameter may be -1 in + order to limit the size of the ast_str to its current allocated + size. The problem was that the max_len parameter in all cases was + a size_t, which is unsigned. Thus a -1 was interpreted as + UINT_MAX instead of -1. Changing the max_len parameter to be + ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an + off-by-one error in the case where we attempted to write a string + larger than the current allotted size to a string when -1 was + passed as the max_len parameter. When trying to write more than + the allotted size, the ast_str's __AST_STR_USED was set to 1 + higher than it should have been. Thanks to Tilghman for quickly + spotting the offending line of code. Oh, and the unit test that I + referenced in the top line of this commit will be added to + reviewboard shortly. Sit tight... ........ + + * /, apps/app_queue.c: Merged revisions 247169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb + 2010) | 9 lines Merged revisions 247168 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb + 2010) | 3 lines Make sure that when autofill is disabled that + callers not in the front of the queue cannot place calls. + ........ ................ + + * /, include/asterisk/strings.h: Merged revisions 246985 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue, + 16 Feb 2010) | 3 lines Add some clarifying documentation to the + ast_str_set and ast_str_append functions. ........ + +2010-02-16 21:07 +0000 [r246903-246984] David Vossel <dvossel@digium.com> + + * main/tcptls.c, /: Merged revisions 246980 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 | + dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines + warning message if openssl support is missing while attempting + tls connection (closes issue #16673) Reported by: michaesc + Patches: tls_error_msg.diff uploaded by dvossel (license 671) + ........ + + * main/channel.c, /: Merged revisions 246899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 | + dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines + fixes sample rate conversion issue with Monitor application When + using ast_seekstream with the read/write streams of a monitor, + the number of samples we are seeking must be of the same rate as + the stream or the jump calculation will be incorrect. This patch + adds logic to correctly convert the number of samples to jump to + the sample rate the read/write stream is using. For example, if + the call is G722 (16khz) and the read/write stream is recording a + 8khz wav, seeking 320 samples of 16khz audio is not the same as + seeking 320 samples of 8khz audio when performing the + ast_seekstream on the stream. ABE-2044 ........ + +2010-02-15 23:44 +0000 [r246711] Tilghman Lesher <tlesher@digium.com> + + * Makefile, /: Merged revisions 246710 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010) + | 12 lines Merged revisions 246709 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) + | 5 lines Make the menuselect instructions correct by allowing + 'make menuselect' to actually solve dependency problems. + (Previously, it would fail out again with the same message about + running 'make menuselect', which was NOT at all helpful.) + ........ ................ + +2010-02-12 23:35 +0000 [r246549] David Vossel <dvossel@digium.com> + + * main/channel.c, /: Merged revisions 246546 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010) + | 21 lines Merged revisions 246545 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) + | 16 lines lock channel during datastore removal On channel + destruction the channel's datastores are removed and destroyed. + Since there are public API calls to find and remove datastores on + a channel, a lock should be held whenever datastores are removed + and destroyed. This resolves a crash caused by a race condition + in app_chanspy.c. (closes issue #16678) Reported by: + tim_ringenbach Patches: datastore_destroy_race.diff uploaded by + tim ringenbach (license 540) Tested by: dvossel ........ + ................ + +2010-02-12 18:57 +0000 [r246462] Jason Parker <jparker@digium.com> + + * main/channel.c: Fix some silly formatting that made my head hurt. + +2010-02-10 21:27 +0000 [r246201-246205] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_strings.c: Merged revisions 246204 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010) + | 2 lines Fussy compiler on another machine... ........ + + * /, funcs/func_strings.c: Merged revisions 246200 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010) + | 2 lines Fix weird issue with unit tests on optimized build - + turned out to be a signing issue. ........ + +2010-02-10 17:56 +0000 [r246122] David Vossel <dvossel@digium.com> + + * /, apps/app_queue.c: Merged revisions 246116 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010) + | 14 lines Merged revisions 246115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) + | 8 lines fixes random deadlock in app_queue with use_weight + during reload (closes issue #16677) Reported by: tim_ringenbach + Patches: app_queue_use_weight_deadlock.diff uploaded by tim + ringenbach (license 540) ........ ................ + +2010-02-10 16:53 +0000 [r246071] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_local.c, /: Merged revisions 246070 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010) + | 22 lines Change channel state on local channels for + busy,answer,ring. Previously local channels channel state never + changed. This became problematic when the state of the other side + of the local channel was lost, for example during a masquerade. + Changing the state of the local channel allows for the scenario + to be detected when the channel state is set to ringing, but the + peer isn't ringing. The specific problem scenario is described in + 164201. Although this was noted on one of the issues, here is the + tested dialplan verified to work: exten => + 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten => + *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) + exten => *9700,n,wait(3) ;3 works, 1 did not exten => + *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did + not exten => + 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes + issue #14992) Reported by: davidw ........ + +2010-02-10 15:38 +0000 [r245946-246023] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_strings.c: Merged revisions 246022 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010) + | 2 lines Enable warnings on atypical conditions for the FILTER + function (suggested by mmichelson on the -dev list). ........ + + * configs/extensions.conf.sample, /, funcs/func_strings.c: Merged + revisions 245945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010) + | 9 lines Merged revisions 245944 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) + | 2 lines Include examples of FILTER usage in extension patterns + where a "." may be a risk. ........ ................ + +2010-02-09 23:14 +0000 [r245796] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 245793 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600 + (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) + | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 = + 32768 which is the maximum allowed iax2 callnumber. Creating the + iaxs and iaxsl array of size 32768 means the maximum callnumber + is actually out of bounds. This causes a nasty crash. (closes + issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded + by dvossel (license 671) ........ ................ + +2010-02-09 18:09 +0000 [r245730] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_fax.c: Merged revisions 245729 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 | + tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines + Ensure frames are only freed once. (closes issue #16361) Reported + by: vlad Patches: 20100208__issue16361.diff.txt uploaded by + tilghman (license 14) Tested by: kenny, bloodoff, misaksen + ........ + +2010-02-09 16:25 +0000 [r245681] Kevin P. Fleming <kpfleming@digium.com> + + * /, apps/app_fax.c: Merged revisions 245680 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 | + kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8 + lines Don't offer MMR or JBIG transcoding during T.38 + negotiation. After further discussion with Steve Underwood, we + should not (yet) be offering to receive MMR or JBIG transcoded + streams from T.38 endpoints. A future spandsp release will + support those features, and then they can be enabled during + negotiation ........ + +2010-02-08 23:51 +0000 [r245627] Jason Parker <jparker@digium.com> + + * apps/app_voicemail.c: Stop playing the message number multiple + times. Also remove some accidentally duplicated code, which may + have been causing a memleak. This was caused by a bad merge. + (closes issue #16579) Reported by: kue Patches: 0016525.patch + uploaded by hokie21 (license 987) + +2010-02-08 22:46 +0000 [r245579] Tilghman Lesher <tlesher@digium.com> + + * /, main/Makefile, channels/Makefile: Merged revisions 245578 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08 + Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and + channels/ Makefiles. They were previously passed correctly, but + they simply weren't used. This caused issues with various + platforms whose builds needed to pass special linker flags via + the configure script. (closes issue #16596) Reported by: + pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by + pprindeville (license 347) Tested by: tilghman ........ + +2010-02-08 20:42 +0000 [r245498] Jason Parker <jparker@digium.com> + + * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r245497 | qwell | 2010-02-08 14:41:05 -0600 + (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) | + 4 lines Remove reference of documentation in source directory. + People don't always build Asterisk from source (distro packages, + anybody?). ........ ................ + +2010-02-05 19:26 +0000 [r245093] Jeff Peeler <jpeeler@digium.com> + + * contrib/firmware (removed), /, LICENSE: Merged revisions 245090 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600 + (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb + 2010) | 5 lines Remove contrib/firmware directory as it is empty + Remove explicit license for IAXy firmware as it is no longer + included in the tree ........ ................ + +2010-02-05 17:10 +0000 [r244928] Sean Bright <sean@malleable.com> + + * main/asterisk.c, /: Merged revisions 244927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb + 2010) | 9 lines Merged revisions 244926 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb + 2010) | 1 line Update main copyright date. ........ + ................ + +2010-02-03 18:40 +0000 [r244506] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 244505 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r244505 | tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) + | 8 lines The chanvar= setting should inherit the entire list of + variables, not just the first one. (closes issue #16359) Reported + by: raarts Patches: dahdi-setvars.diff uploaded by raarts + (license 937) Tested by: raarts ........ + +2010-02-02 22:32 +0000 [r244447] David Vossel <dvossel@digium.com> + + * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: + Merged revisions 244443 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 | + dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines + fixes crash during T.38 negotiation caused by invalid or missing + FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported + by: krn (closes issue #16724) Reported by: barthpbx (closes issue + #16517) Reported by: bklang (closes issue #16485) Reported by: + elsto ........ + +2010-02-02 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.22 + + * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can + remotely crash Asterisk by modifying the FaxMaxDatagram field of + the SDP to contain either a negative or exceptionally large value. + The same crash occurs when the FaxMaxDatagram field is omitted from + the SDP as well. + +2010-01-14 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.21 + +2010-01-08 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.21-rc1 + +2010-01-07 21:17 +0000 [r238494] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_oss.c, main/poll.c, channels/chan_usbradio.c, + include/asterisk/utils.h, /, channels/chan_sip.c, + channels/chan_alsa.c, channels/chan_console.c: Merged revisions + 209400 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | + kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 + lines Define side-effect-safe MIN and MAX macros and remove + duplicate definitions from various files. (closes issue #16251) + Reported by: asgaroth ........ + +2010-01-07 20:22 +0000 [r238364-238441] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 238412 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600 + (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) + | 10 lines fixes crash in "scheduled_destroy" in chan_iax A + signed short was used to represent a callnumber. This is makes it + possible to attempt to access the iaxs array with a negative + index. (closes issue #16565) Reported by: jensvb ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 | + dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines + Change in sip show channels display format allowing more digits + for CID (closes issue #16459) Reported by: Rzadzins Patches: + chan_sip_longer_cid.patch uploaded by Rzadzins (license 953) + ........ + + * /, apps/app_queue.c: Merged revisions 238361 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 | + dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines + cli 'queue show' formatting fix. queue name was truncated over 12 + characters (closes issue #16078) Reported by: RoadKill Patches: + quequename_limit.patch uploaded by ppyy (license 906) Tested by: + dvossel ........ + +2010-01-06 21:48 +0000 [r238232] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010) + | 11 lines Merged revisions 238230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010) + | 4 lines Revise documentation on disposition values to the + actual values used. (closes issue #16289) Reported by: wdoekes + ........ ................ + +2010-01-06 20:38 +0000 [r238135-238182] Jeff Peeler <jpeeler@digium.com> + + * /, apps/app_meetme.c: Merged revisions 238181 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 | + jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines + Fix misreverting from 177158. (closes issue #15725) Reported by: + shanermn Patches: v1-15725.patch uploaded by dimas (license 88) + Tested by: shanermn ........ + + * /, main/features.c: Merged revisions 238134 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 | + jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines + Fix channel name comparison for bridge application. The channel + name comparison was not comparing the whole string and therefore + if one channel name was a substring of the other, the bridge + would fail. (closes issue #16528) Reported by: telecos82 Patches: + res_features_r236843.diff uploaded by telecos82 (license 687) + ........ + +2010-01-06 15:20 +0000 [r238011] Russell Bryant <russell@digium.com> + + * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010) + | 14 lines Merged revisions 238009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) + | 7 lines Resolve a crash due to an ast_frame not being fully + initialized. (closes issue #16531) Reported by: john8675309 + (closes SWP-615) ........ ................ + +2010-01-06 06:51 +0000 [r237966] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_sip.c: Something clearly went wrong with a merge + somewhere, because these are all duplicates (and therefore dead + code). + +2010-01-05 23:10 +0000 [r237843-237923] David Vossel <dvossel@digium.com> + + * /, apps/app_queue.c: Merged revisions 237920 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 | + dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines + fixes holdtime playback issue in app_queue When reporting hold + time, the number of seconds should be mod 60. Otherwise audio + playback could be something like "2 minutes 123 seconds" rather + than "2 minutes 3 seconds". Also, the "minute" sound file is + missing, so for the moment until that file can be created the + "minutes" file is used instead. (closes issue #16168) Reported + by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by + nickilo (license ) Tested by: nickilo, wonderg ........ + + * main/pbx.c, /: Merged revisions 237839 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 | + dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines + fixes subscriptions being lost after 'module reload' During a + module reload if multiple extension configs are present, such as + both extensions.conf and extensions.ael, watchers for one + config's hints will be lost during the merging of the other + config. This happens because hint watchers are only preserved for + the current config being merged. The old context list is + destroyed after the merging takes place, meaning any watchers + that were not perserved will be removed. Now all hints are + preserved during merging regardless of what config file is being + merged. These hints are only restored if they are present within + the new context list. (closes issue #16093) Reported by: jlaroff + ........ + +2010-01-05 17:19 +0000 [r237712] Russell Bryant <russell@digium.com> + + * /, main/utils.c: Merged revisions 237699 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010) + | 14 lines Merged revisions 237697 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010) + | 7 lines Change a NOTICE log message to DEBUG where it belongs. + (closes issue #16479) Reported by: alexrecarey (closes SWP-577) + ........ ................ + +2010-01-04 21:51 +0000 [r237407-237575] Tilghman Lesher <tlesher@digium.com> + + * /, main/say.c: Merged revisions 237574 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010) + | 13 lines Merged revisions 237573 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010) + | 6 lines Bounds checking for input string (closes issue #16407) + Reported by: qwell Patches: 20100104__issue16407.diff.txt + uploaded by tilghman (license 14) ........ ................ + + * main/pbx.c, /: Merged revisions 237494 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010) + | 15 lines Merged revisions 237493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010) + | 8 lines Regression in issue #15421 - Pattern matching (closes + issue #16482) Reported by: wdoekes Patches: + astsvn-16482-betterfix.diff uploaded by wdoekes (license 717) + 20091223__issue16482.diff.txt uploaded by tilghman (license 14) + Tested by: wdoekes, tilghman ........ ................ + + * main/config.c, /: Merged revisions 237414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 | + tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines + Oops, didn't compile (thanks, kpfleming) ........ + + * main/config.c, /: Merged revisions 237410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 | + tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines + Further reduce the encoded blank values back to blank in the + realtime API. (closes issue #16533) Reported by: sergee Patches: + 200100104__issue16533.diff.txt uploaded by tilghman (license 14) + Tested by: sergee ........ + + * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged + revisions 237406 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) + | 23 lines Merged revisions 237405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) + | 16 lines Add a flag to disable the Background behavior, for AGI + users. This is in a section of code that relates to two other + issues, namely issue #14011 and issue #14940), one of which was + the behavior of Background when called with a context argument + that matched the current context. This fix broke FreePBX, + however, in a post-Dial situation. Needless to say, this is an + extremely difficult collision of several different issues. While + the use of an exception flag is ugly, fixing all of the issues + linked is rather difficult (although if someone would like to + propose a better solution, we're happy to entertain that + suggestion). (closes issue #16434) Reported by: rickead2000 + Patches: 20091217__issue16434.diff.txt uploaded by tilghman + (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by + tilghman (license 14) Tested by: rickead2000 ........ + ................ + +2010-01-04 16:26 +0000 [r237324] Jeff Peeler <jpeeler@digium.com> + + * /, res/res_agi.c: Merged revisions 237323 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 | + jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines + Fix timeout for AGI command speech recognize. (closes issue + #16297) Reported by: semond ........ + +2010-01-04 16:21 +0000 [r237320] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /: Merged revisions 237319 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600 + (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) + | 3 lines It's also possible for the Local channel to directly + execute an Application. Reviewboard: + https://reviewboard.asterisk.org/r/452/ ........ ................ + +2010-01-02 09:56 +0000 [r237137] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10 + lines Merged revisions 237135 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 + lines Release memory of the contact acl before unloading module + ........ ................ + +2009-12-30 22:00 +0000 [r236983] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_local.c, /: Merged revisions 236982 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600 + (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) + | 9 lines Don't queue frames to channels that have no means to + process them. (closes issue #15609) Reported by: aragon Patches: + 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by + tilghman (license 14) Tested by: aragon Review: + https://reviewboard.asterisk.org/r/452/ ........ ................ + +2009-12-30 21:12 +0000 [r236903] Jeff Peeler <jpeeler@digium.com> + + * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 | + jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines + One more LOW_MEMORY compile fix. ........ + +2009-12-30 17:55 +0000 [r236805-236849] Tilghman Lesher <tlesher@digium.com> + + * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009) + | 4 lines When the field is blank, don't warn about the field + being unable to be coerced, just skip the column. (closes + http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html) + Reported by Nic Colledge on the -dev list, fixed by me. ........ + + * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 | + tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines + Shut down the SIP session timers more gracefully, in order to + prevent a possible crash. (closes issue #16452) Reported by: + corruptor Patches: 20091221__issue16452.diff.txt uploaded by + tilghman (license 14) Tested by: corruptor ........ + +2009-12-28 22:10 +0000 [r236714] Jason Parker <jparker@digium.com> + + * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec + 2009) | 8 lines Allow "REMAINDER" to function properly in + expressions. (closes issue #16427) Reported by: wdoekes Patches: + ast16-reminder-remainder.patch uploaded by wdoekes (license 717) + Tested by: wdoekes ........ + +2009-12-28 17:39 +0000 [r236668] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009) + | 4 lines Use recommended option, not deprecated option. (closes + issue #16515) Reported by: ManChicken ........ + +2009-12-28 15:31 +0000 [r236511-236633] Sean Bright <sean@malleable.com> + + * include/asterisk/threadstorage.h, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: Merged revisions + 236613 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec + 2009) | 14 lines Merged revisions 236585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec + 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT + requires extra braces. There was conditional code (based on build + platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that + was removed since it is fixed in newer versions of + Solaris/OpenSolaris, but I am still running into it on Solaris 10 + x86 so add a configure-time check for it. ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec + 2009) | 19 lines Merged revisions 236509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec + 2009) | 12 lines Avoid a crash with large numbers of MeetMe + conferences. Similar to changes made to Queue(), when we have + large numbers of conferences in meetme.conf (1000s) and we use + alloca()/strdupa(), we can blow out the stack and crash, so + instead just use a single fixed buffer. (closes issue #16509) + Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded + by seanbright (license 71) Tested by: seanbright ........ + ................ + +2009-12-27 18:22 +0000 [r236435] Tilghman Lesher <tlesher@digium.com> + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600 + (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 + Dec 2009) | 2 lines Turn on colors in the daemon, since there's + many requests for it on Ubuntu. ........ ................ + +2009-12-26 15:29 +0000 [r236359] Kevin P. Fleming <kpfleming@digium.com> + + * /, sounds/Makefile: Merged revisions 236358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec + 2009) | 9 lines Merged revisions 236357 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec + 2009) | 1 line update to latest releases with zero uid/gid + ........ ................ + +2009-12-23 18:26 +0000 [r236187-236301] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 | + tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines + AGI may be invoked from outside the dialplan (closes issue + #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt + uploaded by tilghman (license 14) Tested by: atis ........ + + * /, res/res_agi.c: Merged revisions 236186 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009) + | 11 lines Merged revisions 236184 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009) + | 4 lines If EXEC only gets a single argument, don't crash when + the second is used. (closes issue #16504) Reported by: bklang + ........ ................ + +2009-12-22 17:10 +0000 [r236066] David Vossel <dvossel@digium.com> + + * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009) + | 18 lines Merged revisions 236062 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) + | 11 lines fixes issue with p->method incorrectly set to ACK It + is possible for a second ACK to come in for a retransmitted + message. If an ack does not match an unacked message in our + queue, restore the previous p->method as this ACK is completely + ignored. (closes issue #16295) Reported by: omolenkamp Patches: + issue16295_v2.diff uploaded by dvossel (license 671) ........ + ................ + +2009-12-21 19:54 +0000 [r235942] Jeff Peeler <jpeeler@digium.com> + + * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009) + | 20 lines Merged revisions 235940 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) + | 13 lines Change Monitor to not assume file to write to does not + contain pathing. 227944 changed the fname_base argument to always + append the configured monitor path. This change was necessary to + properly compare files for uniqueness. If a full path is given + though, nothing needs to be appended and that is handled + correctly now. (closes issue #16377) (closes issue #16376) + Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch + uploaded by dant (license 670) ........ ................ + +2009-12-21 17:11 +0000 [r235824] Tilghman Lesher <tlesher@digium.com> + + * /, main/features.c: Merged revisions 235822 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009) + | 15 lines Merged revisions 235821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009) + | 8 lines Send parking lot announcement to the channel which + parked the call, not the park-ee. (closes issue #16234) Reported + by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded + by tilghman (license 14) 20091221__issue16234__1.4.diff.txt + uploaded by tilghman (license 14) Tested by: yeshuawatso ........ + ................ + +2009-12-18 22:58 +0000 [r235662] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /, include/asterisk/cdr.h: Merged revisions + 235660 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009) + | 55 lines Merged revisions 235635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) + | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is + simple in that it reorders the disposition defines so that the + fix for issue 12946 works properly (the default CDR disposition + was changed to AST_CDR_NOANSWER). Also, the + AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all + CDR records are written. The side effects of CDR changes are + scary, so I'm documenting the test cases performed to attempt to + catch any regressions. The following tests were all performed + using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls + B (busy) Hangup C Hangup A (Both SIP and features) A calls B A + blind transfers to C Hangup C (Both SIP and features) A calls B A + attended transfers to C Hangup C A calls B A attended transfers + to C (SIP) C blind transfers to A (features) Hangup A All of the + test scenario CDRs matched. The following tests were performed + just with the patch to ensure proper operation (with + unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten + =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) + (closes issue #16180) Reported by: aatef Patches: bug16180.patch + uploaded by jpeeler (license 325) ........ ................ + +2009-12-18 22:42 +0000 [r235574-235657] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 235656 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600 + (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18 + Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion + ........ ................ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 235573 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r235573 | tilghman | 2009-12-18 15:19:43 -0600 (Fri, 18 Dec 2009) + | 9 lines Merged revisions 235572 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009) + | 2 lines Point to the typical missing package, not the cryptic + "termcap support". ........ ................ + +2009-12-17 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.20 + +2009-12-09 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.20-rc1 + +2009-12-09 20:00 +0000 [r233841-233881] Russell Bryant <russell@digium.com> + + * main/loader.c, /: Fix breakage of the "module load <module>" CLI + command. + + * main/loader.c, formats/format_ilbc.c, formats/format_vox.c, + include/asterisk/module.h, formats/format_pcm.c, + formats/format_h263.c, formats/format_g723.c, + formats/format_h264.c, formats/format_jpeg.c, + formats/format_g726.c, formats/format_gsm.c, + formats/format_g729.c, main/editline/makelist.in, + formats/format_sln.c, formats/format_wav.c, + formats/format_ogg_vorbis.c, UPGRADE.txt, UPGRADE-1.6.txt, + formats/format_wav_gsm.c, formats/format_sln16.c: Set a module + load priority for format modules. A recent change to + app_voicemail made it such that the module now assumes that all + format modules are available while processing voicemail + configuration. However, when autoloading modules, it was possible + that app_voicemail was loaded before the format modules. Since + format modules don't depend on anything, set a module load + priority on them to ensure that they get loaded first when + autoloading. This version of the patch is specific to Asterisk + 1.4 and 1.6.0. These versions did not already support module load + priority in the module API. This adds a trivial version of this + which is just a module flag to include it in a pass before + loading "everything". Thanks to mmichelson for the review! + (closes issue #16412) Reported by: jiddings Tested by: russell + Review: https://reviewboard.asterisk.org/r/445/ + +2009-12-08 18:28 +0000 [r233729] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 233718 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009) + | 8 lines Find another ref leak and change how we manage module + references. (closes issue #16388) Reported by: parisioa Patches: + 20091208__issue16388.diff.txt uploaded by tilghman (license 14) + Tested by: parisioa, tilghman Review: + https://reviewboard.asterisk.org/r/442/ ........ + +2009-12-07 23:57 +0000 [r233617] Atis Lezdins <atis@iq-labs.net> + + * contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 + lines Fix compatibility with valgrind 3.3 and older. (noticed in + issue #16388) Reported by: parisioa Patches: valgrind.supp + uloaded by atis (license 242) Tested by: atis, parisioa ........ + +2009-12-07 23:30 +0000 [r233475-233614] David Vossel <dvossel@digium.com> + + * /, main/utils.c: Merged revisions 233611 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 | + dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines + fixes incorrect logic in ast_uri_encode issue #16299 ........ + + * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009) + | 15 lines Merged revisions 233471 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) + | 9 lines fixes missing Contact header angle brackets (closes + issue #16298) Reported by: mgernoth Patches: + reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested + by: dvossel ........ ................ + +2009-12-07 16:16 +0000 [r233397] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 | + mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8 + lines Do not reject SDP packets describing only non audio + streams. (closes issue #16387) Reported by: zalex1953 Patches: + media-level-c-fix1.diff uploaded by mnicholson (license 96) + Tested by: mnicholson, zalex1953 ........ + +2009-12-04 21:56 +0000 [r233284] David Vossel <dvossel@digium.com> + + * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600 + (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) + | 7 lines clarify requirecalltoken option in iax.sample.conf + (closes issue #16223) Reported by: bklang Patches: + clarify-iax-requirecalltoken.patch uploaded by bklang (license + 919) ........ ................ + +2009-12-04 20:29 +0000 [r233236] Matthias Nick <mnick@digium.com> + + * /, pbx/pbx_config.c: Merged revisions 233093 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 | + mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines + Parse global variables or expressions in hint extensions Parse + global variables or expressions in hint extensions. Like: exten + => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166) + Reported by: rmudgett Tested by: mnick, rmudgett ........ + +2009-12-04 20:11 +0000 [r233230] Russell Bryant <russell@digium.com> + + * /: unblock a rev. + +2009-12-04 17:39 +0000 [r233167] David Vossel <dvossel@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600 + (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) + | 6 lines document and rename strip_control() in app_voicemail + (closes issue #16291) Reported by: wdoekes ........ + ................ + +2009-12-04 17:20 +0000 [r233112] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 233100 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009) + | 14 lines Merged revisions 233092 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009) + | 7 lines Only do frame payload check for HOLD frames. This code + was added for helping to debug the source of invalid HOLD frames. + However, a side effect of this is that it will incorrectly report + errors for frames that have an integer payload. Make the check + for this block specific to the HOLD frame case. ........ + ................ + +2009-12-04 15:46 +0000 [r233047] Matthias Nick <mnick@digium.com> + + * main/dsp.c, /: Merged revisions 233046 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) | + 17 lines Merged revisions 233014 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | + 11 lines Warning message gets displayed only once Added + additional field 'int display_inband_dtmf_warning', which when + set to '1' displays the warning ('Inband DTMF is not supported on + codec %s. Use RFC2833'), and when set to '0' doesn't display the + warning. Otherwise you would get hundreds of warnings every + second. (closes issue #15769) Reported by: falves11 Patches: + patch_15769_14.txt uploaded by mnick (license 874) Tested by: + mnick, falves11 ........ ................ + +2009-12-03 21:03 +0000 [r232864] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600 + (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) + | 8 lines Deprecate "cz" in favor of "cs". Also, change the use + of language codes so that language registers as a prefix, rather + than an exact match. (closes issue #16272) Reported by: patrol-cz + Patches: 20091203__issue16272.diff.txt uploaded by tilghman + (license 14) ........ ................ + +2009-12-03 14:47 +0000 [r232811] David Ruggles <thedavidfactor@gmail.com> + + * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 | + diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12 + lines Prevent double closing of FDs by EIVR This caused a problem + when asterisk was under heavy load and running both AGI and EIVR + applications. EIVR would close an FD at which point it would be + considered freed and be used by a new AGI instance the second + close would then close the FD now in use by AGI. (closes issue + #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec + Review: https://reviewboard.asterisk.org/r/436/ ........ + +2009-12-03 00:32 +0000 [r232699] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 232660-232661 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r232660 | tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 + Dec 2009) | 19 lines Fix multiple issues with musiconhold, which + led to classes not getting destroyed properly. * Classes are now + tracked past removal from the core container, and module removal + is actively prevented until all references are freed. * A hanging + reference stored in the channel has been removed. This could have + caused a mismatch and the music state not properly cleared, if + two or more reloads occurred between MOH being stopped and MOH + being restarted. * In certain circumstances, duplicate classes + were possible. * A race existed at reload time between a process + being killed and the thread responsible for reading from the + related pipe respawning that process. * Several reference counts + have also been corrected. At least one could have caused deleted + classes to stick around forever, consuming resources. This + originally manifested as MOH external processes that were not + killed at reload time. (closes issue #16279, closes issue #16207) + Reported by: parisioa, dcabot Patches: + 20091202__issue16279__2.diff.txt uploaded by tilghman (license + 14) Tested by: parisioa, tilghman ........ r232661 | tilghman | + 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove + debugging line ........ + +2009-12-02 22:03 +0000 [r232577-232583] Jeff Peeler <jpeeler@digium.com> + + * main/manager.c, /: Merged revisions 232582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009) + | 14 lines Merged revisions 232581 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009) + | 7 lines Send ack (response/message) after receiving manager + action userevent (closes issue #16264) Reported by: dimas + Patches: event-ack.patch uploaded by dimas (license 88) ........ + ................ + + * main/manager.c, /: Merged revisions 232576 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 | + jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines + Make manager response to "Action: events" finish with empty line + (closes issue #16275) Reported by: vnovy Patches: manager.c.diff + uploaded by vnovy (license 922) ........ + +2009-12-02 17:08 +0000 [r232357] Joshua Colp <jcolp@digium.com> + + * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) | + 12 lines Merged revisions 232355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 + lines Fix a bug where if you hung up very quickly after calling + AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG. + (closes issue #16239) Reported by: CGMChris ........ + ................ + +2009-12-02 17:03 +0000 [r232354] David Vossel <dvossel@digium.com> + + * /, main/acl.c: Merged revisions 232351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009) + | 12 lines Merged revisions 232350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009) + | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in + strace. (closes issue #16290) Reported by: wdoekes ........ + ................ + +2009-12-02 16:41 +0000 [r232346] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 | + file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add + support for handling the 415 Unsupported media type response like + we do for a 488 Not acceptable here response. (closes issue + #16186) Reported by: atis Patches: sip_t38_response_415.patch + uploaded by atis (license 242) ........ + +2009-12-02 15:45 +0000 [r232272] David Vossel <dvossel@digium.com> + + * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600 + (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) + | 9 lines fixes segfault in func_groupcount closes issue #16337) + Reported by: Parantido Patches: issue_16337.diff uploaded by + dvossel (license 671) Tested by: Parantido, dvossel ........ + ................ + +2009-12-02 00:49 +0000 [r232092] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600 + (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) + | 10 lines Do not modify the gain settings on data calls. (The + digital flag actually represents a data call.) (closes issue + #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt + uploaded by alecdavis (license 585) Tested by: alecdavis ........ + ................ + +2009-12-01 23:39 +0000 [r232009-232013] Russell Bryant <russell@digium.com> + + * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 | + russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines + Fix a build error on FreeBSD. ........ + + * /, main/file.c: Merged revisions 232008 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009) + | 9 lines Merged revisions 232007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) + | 2 lines Fix a warning pointed out by buildbot. ........ + ................ + +2009-12-01 21:57 +0000 [r231928] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /: Merged revisions 231927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009) + | 19 lines Merged revisions 231911 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) + | 12 lines Fix crash with invalid frame data The crash was + happening as a result of a frame containing an invalid data + pointer, but was set with data length of zero. The few times the + issue was reproduced it _seemed_ that the frame was queued + properly, that is the data pointer was set to NULL. I never could + reproduce the crash so as a last resort the crash has been fixed, + but a check in __ast_read has been added to give as much + information about the source of problematic frames in the future. + (closes issue #16058) Reported by: atis ........ ................ + +2009-12-01 21:22 +0000 [r231879] David Vossel <dvossel@digium.com> + + * main/pbx.c, /: Merged revisions 231867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009) + | 9 lines Merged revisions 231853 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009) + | 3 lines WaitExten m option with no parameters generates frame + with zero datalen but non-null data ptr ........ ................ + +2009-12-01 15:51 +0000 [r231744] Matthew Nicholson <mnicholson@digium.com> + + * /, main/file.c: Merged revisions 231741 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec + 2009) | 9 lines Merged revisions 231740 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec + 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce() + and return an error if no know formats are found. ........ + ................ + +2009-11-30 21:52 +0000 [r231693] Kevin P. Fleming <kpfleming@digium.com> + + * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: + Merged revisions 231692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | + kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 + lines Another round of UDPTL stack fixes/improvements: 1) Allow + users of UDPTL stack to associate a character-string tag with a + UDPTL session, so that log/error/debug messages generated by the + UDPTL stack can be 'connected' to the endpoint that caused them + to be generated. 2) Improve comments (and process) of calculating + the far end's maximum IFP size when redundancy mode is in use for + error correction. 3) When an IFP larger than the calculated 'far + max IFP' size is presented for writing, truncate it rather than + putting in the buffer and allowing the buffer to overflow; this + will cause the ends to retrain to a lower bit rate that produces + IFPs of an appropriate size if possible, and if not possible, the + FAX transfer will fail completely. In these cases, it is due to + the one endpoint supplying a T38FaxMaxDatagram value that is + improperly calculated and is too low to be of use; we have + configuration options available to override this behavior. 4) + Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no + longer needed. ........ + +2009-11-30 21:37 +0000 [r231691] Matthew Nicholson <mnicholson@digium.com> + + * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c, + main/app.c: Merged revisions 231688 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov + 2009) | 15 lines Merged revisions 231614 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov + 2009) | 8 lines Remove duplicate entries from voicemail format + lists. This prevents app_voicemail from entering an infinite loop + when the same format is specified twice in the format list. + (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson + Review: https://reviewboard.asterisk.org/r/429/ ........ + ................ + +2009-11-30 20:45 +0000 [r231603] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 | + file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines + When receiving SDP that matches the version of the last one do + not treat it as a fatal error. (closes issue #16238) Reported by: + seandarcy ........ + +2009-11-30 18:58 +0000 [r231517-231560] David Vossel <dvossel@digium.com> + + * /, apps/app_queue.c: Merged revisions 231556 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 | + dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines + app_queue crashes randomly, often during call-transfers This + patch adds a ref to the queue_ent object's parent call_queue in + queue_exec() so the call_queue won't be destroyed while the the + queue_ent still holds a pointer to it. (closes issue 0015686) + Tested by: dvossel, aragon ........ + + * main/rtp.c, /: Merged revisions 231491 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009) + | 17 lines Merged revisions 231441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009) + | 11 lines fixes crash caused by RTP comfort noise payload + greater than 24 bytes AST-2009-010 (closes issue #16242) Reported + by: amorsen Patches: issue16242.diff uploaded by oej (license + 306) Tested by: amorsen, oej, dvossel ........ ................ + +2009-11-25 22:34 +0000 [r231300] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 231299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009) + | 9 lines Merged revisions 231298 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) + | 2 lines After a frame duplication failure, unlock the channel + before returning. ........ ................ + +2009-11-25 15:48 +0000 [r231192] Matthew Nicholson <mnicholson@digium.com> + + * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 | + mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4 + lines Load pbx_lua with global symbols to allow linking with + other lua libraries. Found by Maxim Litnitskiy. ........ + +2009-11-24 18:53 +0000 [r231096] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: Merged revisions 231095 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 | + jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines + Fix erroneous hangup extension execution ast_spawn_extension + behaves differently from 1.4 in that hangups and extensions that + do not exist do not return an error, whereas in 1.6 it does. This + is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN + flag gets set properly. (closes issue #16106) Reported by: + ajohnson Tested by: ajohnson ........ + +2009-11-23 15:46 +0000 [r230882] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 230881 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | + file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines + Change fax detection in chan_sip so it behaves as one would + expect. Internally the way T.38 is negotiated has changed and the + option no longer reflects a behavior that is valid. It will now + look for a CNG tone on received calls and if present send the + call to the 'fax' extension. It is then up to the application or + channel to request the switch over to T.38. ........ + +2009-11-23 15:35 +0000 [r230782-230878] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov + 2009) | 9 lines Merged revisions 230839 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov + 2009) | 1 line Correct fix for issue #16268... the reporter's + original patch was very close to correct. ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov + 2009) | 12 lines Merged revisions 230772 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov + 2009) | 5 lines Ensure that SDP parsing does not ignore the last + line of the SDP. (closes issue #16268) Reported by: sgimeno + ........ ................ + +2009-11-20 22:37 +0000 [r230729] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 230726 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009) + | 7 lines fixes iax2 show cache locking error, thanks alecdavis! + (closes issue #16094) Reported by: alecdavis Patches: + bug16094.diff.txt uploaded by alecdavis (license 585) Tested by: + alecdavis, dvossel ........ + +2009-11-20 21:09 +0000 [r230631] Matthew Nicholson <mnicholson@digium.com> + + * /, main/features.c: Merged revisions 230628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov + 2009) | 15 lines Merged revisions 230627 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov + 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR + if it exists. This is necessary for the recordagentcalls option + in chan_agent to store the recorded file name in the bridge CDR. + (closes issue #14590) Reported by: msetim Patches: + queue_agent_userfield.patch uploaded by Laureano (license 265) + Tested by: Laureano, mnicholson ........ ................ + +2009-11-20 17:37 +0000 [r230512-230587] David Vossel <dvossel@digium.com> + + * /, include/asterisk/audiohook.h, main/audiohook.c: Merged + revisions 230583 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 | + dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines + audiohook signal trigger on every status change (issue #14618) + Review: https://reviewboard.asterisk.org/r/434/ ........ + + * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600 + (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) + | 10 lines fixes MixMonitor thread not exiting when + StopMixMonitor is used (closes issue #16152) Reported by: AlexMS + Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license + 671) Tested by: dvossel, AlexMS Review: + https://reviewboard.asterisk.org/r/424/ ........ ................ + +2009-11-30 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.19 + + * AST-2009-010 + + * SDP parser regression fix (issue #16268, issue #16238) + +2009-11-18 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.18 + +2009-11-13 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.18-rc3 + +2009-11-13 15:56 +0000 [r229913] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 | + file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix + T.38 negotiation regression introduced with the SDP parser + changes. ........ + +2009-11-12 16:49 +0000 [r229673] David Vossel <dvossel@digium.com> + + * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600 + (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009) + | 6 lines fixes merging error, datastore was being freed in the + wrong function. (closes issue #16219) Reported by: aragon + ........ ................ + +2009-11-11 19:51 +0000 [r229475-229500] David Brooks <dbrooks@digium.com> + + * main/pbx.c: Merged revisions 229499 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009) + | 15 lines Merged revisions 229498 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009) + | 8 lines Solaris doesn't like NULL going to ast_log Solaris will + crash if NULL is passed to ast_log. This simple patch simply uses + S_OR to get around this. (closes issue #15392) Reported by: + yrashk ........ ................ + + * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009) + | 7 lines Flags not initialized in app_softhangup.c, causing + undefined behavior Trivial patch [kobaz] to initialize an + ast_flags = {0} (closes issue #16129) Reported by: kobaz ........ + +2009-11-10 22:17 +0000 [r229363] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 229361 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009) + | 19 lines Merged revisions 229360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) + | 12 lines If two pattern classes start with the same digit and + have the same number of characters, they will compare equal. The + example given in the issue report is that of [234] and [246], + which have these characteristics, yet they are clearly not + equivalent. The code still uses these two characteristics, yet + when the two scores compare equal, an additional check will be + done to compare all characters within the class to verify + equality. (closes issue #15421) Reported by: jsmith Patches: + 20091109__issue15421__2.diff.txt uploaded by tilghman (license + 14) Tested by: jsmith, thedavidfactor ........ ................ + +2009-11-10 22:03 +0000 [r229357] David Ruggles <thedavidfactor@gmail.com> + + * doc/externalivr.txt: Merged revisions 229356 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov + 2009) | 16 lines Merged revisions 229355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov + 2009) | 9 lines Fix ExternalIVR Documentation Remove + documentation for event that doesn't function (closes issue + #16220) Reported by: thedavidfactor Patches: + externalivr.txt.20091110.1622.patch uploaded by thedavidfactor + (license 903) ........ ................ + +2009-11-10 21:31 +0000 [r229352] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 | + tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines + When GOSUB is invoked within an AGI, it may not exit correctly. + (closes issue #16216) Reported by: atis Patches: + 20091110__atis_work.diff.txt uploaded by tilghman (license 14) + Tested by: atis ........ + +2009-11-10 20:07 +0000 [r229283] Joshua Colp <jcolp@digium.com> + + * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) | + 15 lines Merged revisions 229281 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8 + lines Remove broken support for direct transcoding between G.726 + RFC3551 and G.726 AAL2. On some systems the translation core + would actually consider g726aal2 -> g726 -> signed linear to be a + quicker path then g726aal2 -> signed linear which exposed this + problem. (closes issue #15504) Reported by: globalnetinc ........ + ................ + +2009-11-10 17:54 +0000 [r229234] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 229168 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600 + (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009) + | 9 lines don't crash on log message in solaris AST-2009-006 + (closes issue #16206) Reported by: bklang Tested by: bklang + ........ ................ + +2009-11-10 17:37 +0000 [r229229] David Ruggles <thedavidfactor@gmail.com> + + * doc/externalivr.txt: Merged revisions 229228 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov + 2009) | 18 lines Merged revisions 229191 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov + 2009) | 11 lines Document ExternalIVR event tag collision + ExternalIVR uses the D tag for two different event types. This + documents that behavior and how to differentiate between the two + cases. Also includes a minor spelling fix and clarification + (closes issue #16211) Reported by: thedavidfactor Patches: + externalivr.txt.20091109.1507.patch uploaded by thedavidfactor + (license 903) ........ ................ + +2009-11-10 15:41 +0000 [r229100] Matthew Nicholson <mnicholson@digium.com> + + * channels/chan_sip.c: Reverted revision 202006. (closes issue + #16175) Reported by: paul-tg + +2009-11-10 11:19 +0000 [r229057] Gavin Henry <ghenry@suretecsystems.com> + + * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10 + Nov 2009) | 20 lines Schema file additions * Added + AsteriskDialplan, AsteriskAccount and AsteriskMailbox + objectClasses to allow standalone dialplan, account and mailbox + entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage, + AstAccountTransport, AstAccountPromiscRedir, - + AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap, + - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed + redundant IPaddr (there's already IPAddress) - Gives more + configuration Flags for SIP-Users available (tested) - Allows to + create Asterisk Attributes in defined Asterisk ObjectClasses + without extensibleObject (which really should be the last + resort); gives also additional possibilities for LDAP-filter + (closes issue #15874) Reported by: Medozas Patches: + asterisk.ldap-schema.patch uploaded by Medozas (license 41) + Tested by: Medozas, suretec ........ + +2009-11-09 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.18-rc2 + +2009-11-09 15:39 +0000 [r228898] 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 14:58 +0000 [r228861] 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 22:38 +0000 [r228696] 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 20:42 +0000 [r228651] 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 18:40 +0000 [r228549] 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 1.6.0.18-rc1 + +2009-11-06 17:53 +0000 [r228479-228500] 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. ........ + + * channels/chan_sip.c: Fix a logic flaw I introduced when I was + testing stuff out. + +2009-11-06 17:10 +0000 [r228423] David Vossel <dvossel@digium.com> + + * /, 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 16:56 +0000 [r228411-228415] Joshua Colp <jcolp@digium.com> + + * channels/chan_sip.c: Fix a crash caused by freeing a dialog + directly instead of using dialog_unref. (closes issue #16097) + Reported by: steinwej Patches: no_RTP.diff uploaded by steinwej + (license 841) + + * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) | + 14 lines Merged revisions 228409 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 + lines Fix a bug caused by a partially invalid frame (from the + jitterbuffer) passing through the Asterisk core. (closes issue + #15560) Reported by: jvandal (closes issue #15709) Reported by: + covici ........ ................ + +2009-11-06 15:44 +0000 [r228271-228342] 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 21:24 +0000 [r228190] 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 19:39 +0000 [r228146] 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 19:17 +0000 [r228081] 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 ........ ................ + +2009-11-04 23:52 +0000 [r227946] 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 21:15 +0000 [r227763-227833] 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 19:27 +0000 [r227717-227743] 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 20:00 +0000 [r227373] 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 [r227362-227369] 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 18:05 +0000 [r227278] 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 15:37 +0000 [r227168] 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 15:24 +0000 [r227163] 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 11:21 +0000 [r227102] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 227091 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15 + lines Merged revisions 227088 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 + lines Use proper response code when violating Contact ACL's. + https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a + quick review. (EDVX-003) ........ ................ + +2009-11-02 21:05 +0000 [r226975-226976] 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/ + + * /: 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 18:09 +0000 [r226891] 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 17:16 +0000 [r226813] 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 18:14 +0000 [r226533] Joshua Colp <jcolp@digium.com> + + * channels/chan_local.c, /, doc/tex/localchannel.tex: Merged + revisions 226532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | + 13 lines Merged revisions 226531 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 + lines Add an option to enabling passing music on hold start and + stop requests through instead of acting on them in chan_local. + (closes issue #14709) Reported by: dimas ........ + ................ + +2009-10-28 20:17 +0000 [r226381-226387] 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 18:05 +0000 [r226167-226306] 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 ........ ................ + +2009-10-26 23:13 +0000 [r226019] 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 15:46 +0000 [r225869] 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. (issue #16127) + +2009-10-23 14:05 +0000 [r225583] 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 21:53 +0000 [r225486] 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 17:13 +0000 [r225361] 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 22:10 +0000 [r225310-225311] 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/iax.conf.sample, /, channels/chan_sip.c, + configs/sip.conf.sample, channels/chan_iax2.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/ ........ ................ + +2009-10-21 03:15 +0000 [r224933] Russell Bryant <russell@digium.com> + + * include/asterisk/translate.h, main/dsp.c, main/frame.c, /, + main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c, + include/asterisk/frame.h: Merged revisions 224932 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r224932 | russell | 2009-10-20 22:09:04 -0500 + (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) + | 5 lines Isolate frames returned from a DSP instance or codec + translator. The reasoning for these changes are the same as what + I wrote in the commit message for rev 222878. ........ + ................ + +2009-10-20 22:10 +0000 [r224857] Tilghman Lesher <tlesher@digium.com> + + * /, main/audiohook.c: Merged revisions 224856 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) + | 12 lines Merged revisions 224855 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) + | 5 lines Pay attention to the return value of the manipulate + function. While this looks like an optimization, it prevents a + crash from occurring when used with certain audiohook callbacks + (diagnosed with SVN trunk, backported to 1.4 to keep the source + consistent across versions). ........ ................ + +2009-10-20 17:48 +0000 [r224775] 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 ........ ................ + +2009-10-19 23:50 +0000 [r224672] 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 19:50 +0000 [r224568] 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 00:12 +0000 [r224449] 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. This corrects an issue reported on the -users list. + ........ + +2009-10-17 02:03 +0000 [r224332-224337] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: fix typo, sorry + + * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500 + (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) + | 13 lines Fix stale caller id data from being reported in AMI + NewChannel event The problem here is that chan_dahdi is designed + in such a way to set certain values in the dahdi_pvt only once. + One of those such values is the configured caller id data in + chan_dahdi.conf. For PRI, the configured caller id data could be + overwritten during a call. Instead of saving the data and + restoring, it was decided that for all non-analog channels it was + simply best to not set the configured caller id in the first + place and also clear it at the end of the call. (closes issue + #15883) Reported by: jsmith ........ ................ + +2009-10-16 20:48 +0000 [r224262] 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 15:57 +0000 [r224179] 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 23:50 +0000 [r223833] 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 21:07 +0000 [r223759] 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 14:28 +0000 [r223653] 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 17:27 +0000 [r223488] 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 23:08 +0000 [r223404] 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 20:59 +0000 [r223331] 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 18:36 +0000 [r223276] 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 18:20 +0000 [r223226] 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 17:57 +0000 [r223210] 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 [r223172] 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 17:11 +0000 [r223091-223135] 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 19:54 +0000 [r222881] Russell Bryant <russell@digium.com> + + * include/asterisk/file.h, main/frame.c, /, main/file.c, + include/asterisk/frame.h: Merged revisions 222880 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r222880 | russell | 2009-10-08 14:52:03 -0500 + (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) + | 44 lines Make filestream frame handling safer by isolating + frames before returning them. This patch is related to a number + of issues on the bug tracker that show crashes related to freeing + frames that came from a filestream. A number of fixes have been + made over time while trying to figure out these problems, but + there re still people seeing the crash. (Note that some of these + bug reports include information about other problems. I am + specifically addressing the filestream frame crash here.) I'm + still not clear on what the exact problem is. However, what is + _very_ clear is that we have seen quite a few problems over time + related to unexpected behavior when we try to use embedded frames + as an optimization. In some cases, this optimization doesn't + really provide much due to improvements made in other areas. In + this case, the patch modifies filestream handling such that the + embedded frame will not be returned. ast_frisolate() is used to + ensure that we end up with a completely mallocd frame. In + reality, though, we will not actually have to malloc every time. + For filestreams, the frame will almost always be allocated and + freed in the same thread. That means that the thread local frame + cache will be used. So, going this route doesn't hurt. With this + patch in place, some people have reported success in not seeing + the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) + Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt + uploaded by russell (license 2) Tested by: aragon, russell + (closes issue #15817) Reported by: zerohalo Tested by: zerohalo + (closes issue #15845) Reported by: marhbere Review: + https://reviewboard.asterisk.org/r/386/ ........ ................ + +2009-10-08 19:42 +0000 [r222876] 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 16:47 +0000 [r222693-222800] 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 18:35 +0000 [r222605] Sean Bright <sean@malleable.com> + + * main/pbx.c: Properly initialize ast_devstate_aggregate so we + don't crash sporadically. This looks like it was just missed + during a merge. (closes issue #15841) Reported by: amorsen + Patches: + ast_devstate_aggregate_init-in-ast_extension_state2.patch + uploaded by amorsen (license 676) Tested by: amorsen (closes + issue #15852) Reported by: amorsen Tested by: amorsen, farisraouf + +2009-10-07 17:47 +0000 [r222546] 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) ........ ................ + +2009-10-07 17:32 +0000 [r222541] Tilghman Lesher <tlesher@digium.com> + + * res/ael/pval.c: Small typo (thanks, jpeeler) + +2009-10-06 23:57 +0000 [r222302-222464] 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) (closes issue #15683) Reported by: alecdavis ........ + ................ + + * channels/chan_dahdi.c: Fix potential crash when entire span + request is received. The variable index used in this scenario for + accessing the dahdi_pvts was wrong and was most likely copied + from the several other places it is used correctly. (closes issue + #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch + uploaded by tsearle (license 373) Modified: + branches/1.4/channels/chan_dahdi.c + + * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009) + | 9 lines Fix 222298 (crash during destruction of second channel + when variable set with setvar). I mistakenly reasoned that setvar + would be used on all channels. Since it can be set per channel, + give each dahdi channel a copy of the variable. (related to + #15899) ........ + + * 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 19:19 +0000 [r222279] Tilghman Lesher <tlesher@digium.com> + + * res/ael/pval.c, /: Recorded merge of revisions 222273 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r222273 | tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 + Oct 2009) | 5 lines When we call a gosub routine, the variables + should be scoped to avoid contaminating the caller. This affected + the ~~EXTEN~~ hack, where a subroutine might have changed the + value before it was used in the caller. Patch by myself, tested + by ebroad on #asterisk ........ + +2009-11-04 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.17 + + * AST-2009-008 and AST-2009-009 + +2009-10-06 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.16-rc2 + +2009-10-06 01:33 +0000 [r222111-222185] Kevin P. Fleming <kpfleming@digium.com> + + * main/astobj2.c, /, funcs/func_dialgroup.c, + include/asterisk/astobj2.h, res/res_phoneprov.c, + channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c, + channels/chan_iax2.c: Merged revisions 222176 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct + 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 + Oct 2009) | 20 lines Fix ao2_iterator API to hold references to + containers being iterated. See Mantis issue for details of what + prompted this change. Additional notes: This patch changes the + ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum + instead of a macro, with a name that fits our naming policy; + also, it is now necessary to call ao2_iterator_destroy() on any + iterator that has been created. Currently this only releases the + reference to the container being iterated, but in the future this + could also release other resources used by the iterator, if the + iterator implementation changes to use additional resources. + (closes issue #15987) Reported by: kpfleming Review: + https://reviewboard.asterisk.org/r/383/ ........ ................ + + * main/udptl.c, /, channels/chan_sip.c, configs/udptl.conf.sample, + UPGRADE.txt, configs/sip.conf.sample: Recorded merge of 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 17:37 +0000 [r222038] 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 17:00 +0000 [r221972] 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. ........ ................ + +2009-10-02 13:04 +0000 [r221963] Sean Bright <sean@malleable.com> + + * funcs/func_strings.c: Revert XML docs that ended up in the 1.6.0 + and 1.6.1 branches during a merge. + +2009-10-02 03:05 +0000 [r221921] Tilghman Lesher <tlesher@digium.com> + + * /, 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 01:20 +0000 [r221853] 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 00:03 +0000 [r221778] 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 ........ + ................ + +2009-10-01 21:04 +0000 [r221745] 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 20:34 +0000 [r221742] Tilghman Lesher <tlesher@digium.com> + + * /, 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 20:19 +0000 [r221712] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c: Fixes issue with non dynamic hosts not being + set for peers + +2009-10-01 17:09 +0000 [r221662] 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 16:18 +0000 [r221598] 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 23:08 +0000 [r221486] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 221432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep + 2009) | 17 lines Merged revisions 221360 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep + 2009) | 10 lines Fix SRV lookup and Request-URI generation in + chan_sip. This patch adds a new field "portinuri" to the sip + dialog struct and the sip peer struct. That field is used during + RURI generation to determine if the port should be included in + the RURI. It is also used in some places to determine if an SRV + lookup should occur. (closes issue #14418) Reported by: klaus3000 + Tested by: klaus3000, mnicholson Review: + https://reviewboard.asterisk.org/r/369/ ........ ................ + +2009-09-30 20:02 +0000 [r221369] Matthias Nick <mnick@digium.com> + + * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged + revisions 221368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) | + 23 lines Merged revisions 221153,221157,221303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | + 2 lines check bounds - prevents for buffer overflow ........ + r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | + 8 lines added a new dialplan function 'CSV_QUOTE' and changed the + cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr + Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: + mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, + 30 Sep 2009) | 2 lines changed the prototype definition of + csv_quote ........ ................ + +2009-09-30 18:50 +0000 [r221301] Terry Wilson <twilson@digium.com> + + * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h, + configs/sip.conf.sample: Merged revisions 221266 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 + (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) + | 25 lines Change the SSRC by default when our media stream + changes Be default, change SSRC when doing an audio stream + changes Asterisk doesn't honor marker bit when reinvited to + already-bridged RTP streams,resulting in far-end stack discarding + packets with "old" timestamps that areactually part of a new + stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is + a reinvite, unless the 'constantssrc' is set to true in sip.conf. + The original issue reported to Digium support detailed the + following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based + Application Server Call comes in fromITSP, Asterisk dials the app + server which sends a re-invite back toAsterisk--not to negotiate + to send media directly to the ITSP, but to indicatethat it's + changing the stream it's sending to Asterisk. The app + servergenerates a new SSRC, sequence numbers, timestamps, and + sets the marker bit on the new stream. Asterisk passes through + the teimstamp of the new stream, butdoes not reset the SSRC, + sequence numbers, or set the marker bit. When the timestamp on + the new stream is older than the timestamp on the originalstream, + the ITSP (which doesn't know there has been any change) discards + the newframes because it thinks they are too old. This patch + addresses this by changing the SSRC on a stream update unless + constantssrc=true is set in sip.conf. Review: + https://reviewboard.asterisk.org/r/374/ ........ ................ + +2009-09-30 16:57 +0000 [r221202] 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 14:52 +0000 [r221087] 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 04:34 +0000 [r220976-221045] 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 + ........ + + * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) + | 16 lines Merged revisions 220873 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) + | 9 lines Reduce CPU usage related to building a peer merely for + devicestates. This fixes a 100% CPU problem in the SIP driver, + found by profiling the driver while the problem was occurring. + (closes issue #14309) Reported by: pkempgen Patches: + 20090924__issue14309.diff.txt uploaded by tilghman (license 14) + Tested by: pkempgen, vrban ........ ................ + +2009-09-29 20:25 +0000 [r220940] 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 + +2009-09-29 17:04 +0000 [r220834] 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 [r220723] 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. ........ + ................ + +2009-09-26 15:11 +0000 [r220587] 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 20:42 +0000 [r220372] 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 [r220290] 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 [r220101-220219] 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 08:37 +0000 [r220029] 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 ........ ................ + +2009-09-22 21:47 +0000 [r219819] 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 17:03 +0000 [r219724] 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 18:20 +0000 [r219663] 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 03:06 +0000 [r219588] 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 23:23 +0000 [r219454-219523] 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 [r219413] 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 22:36 +0000 [r219365] 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 22:01 +0000 [r219305] 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 [r219265] 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 1.6.0.16-rc1 + +2009-09-17 15:44 +0000 [r219198] 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 [r219064] 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 19:26 +0000 [r218935] 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 18:44 +0000 [r218931] 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 18:11 +0000 [r218869] 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 13:36 +0000 [r218800] 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 22:39 +0000 [r218732] 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 19:31 +0000 [r218690] 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 16:21 +0000 [r218601] 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 16:05 +0000 [r218580] 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 15:42 +0000 [r218505-218573] 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 15:03 +0000 [r218501] 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 22:49 +0000 [r218431] 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 ........ ................ + +2009-09-14 19:49 +0000 [r218362] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /, configs/voicemail.conf.sample, + sounds/Makefile: 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 15:22 +0000 [r218244] 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 19:10 +0000 [r218216] 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 + +2009-09-12 13:10 +0000 [r218108] Michiel van Baak <michiel@vanbaak.info> + + * main/rtp.c: Use the ip for the new 'rtp set debug ip <foo>'. + Since 1.6.X still has the deprecated 'rtp debug ip <foo>' this + patch is different from the fix that went into trunk (closes + issue #15711) Reported by: davidw Patches: + 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) + Tested by: davidw + +2009-09-11 05:58 +0000 [r217920-218051] Tilghman Lesher <tlesher@digium.com> + + * /, 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) ........ + ................ + + * contrib/scripts/iax-friends.sql, /, channels/chan_sip.c, + channels/chan_iax2.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 22:31 +0000 [r217858-217913] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c: 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. Thanks to + oej for finding the issue and suggesting solutions. Review: + https://reviewboard.asterisk.org/r/355/ + + * /, 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 19:53 +0000 [r217736] 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 12:16 +0000 [r217596] 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 20:15 +0000 [r217484] 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 + http://svn.digium.com/svn/asterisk/trunk + +2009-09-09 11:33 +0000 [r217405] 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 21:45 +0000 [r217281] Kevin P. Fleming <kpfleming@digium.com> + + * configure: Commit regenerated configure script that I missed + earlier. + +2009-09-08 20:31 +0000 [r217209] 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 16:38 +0000 [r217075] Kevin P. Fleming <kpfleming@digium.com> + + * /, 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 15:35 +0000 [r217034] 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 14:28 +0000 [r216996] 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 16:38 +0000 [r216645-216843] 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 ........ + + * configs/sip.conf.sample: fix documentation so it agrees with code + + * channels/chan_sip.c, CHANGES: Add doc and turn off premature + media filter by default + + * apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c, + apps/app_disa.c, configs/sip.conf.sample: Merged revisions 216438 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, + 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 + lines Make apps send PROGRESS control frame for early media and + fix too early media issue in SIP The issue at hand is that some + legacy (dying) PBX systems send empty media frames on PRI links + *before* any call progress. The SIP channel receives these frames + and by default signals 183 Session progress and starts sending + media. This will cause phones to play silence and ignore the + later 180 ringing message. A bad user experience. The fix is + twofold: - We discovered that asterisk apps that support early + media ("noanswer") did not send any PROGRESS frame to indicate + early media. Fixed. - We introduce a setting in chan_sip so that + users can disable any relay of media frames before the outbound + channel actually indicates any sort of call progress. In 1.4, + 1.6.0 and 1.6.1, this will be disabled for backward + compatibility. In later versions of Asterisk, this will be + enabled. We don't assume that it will change your Asterisk phone + experience - only for the better. We encourage third-party + application developers to make sure that if they have + applications that wants to send early media, add a PROGRESS + control frame transmission to make sure that all channel drivers + actually will start sending early media. This has not been the + default in Asterisk previous to this patch, so if you got + inspiration from our code, you need to update accordingly. Sorry + for the trouble and thanks for your support. This code has been + running for a few months in a large scale installation (over 250 + servers with PRI and/or BRI links to old PBX systems). That's no + proof that this is an excellent patch, but, well, it's tested :-) + ........ ................ + +2009-09-04 19:32 +0000 [r216595] 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 17:32 +0000 [r216548] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /, UPGRADE-1.6.txt: Merged revisions 216547 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04 + Sep 2009) | 3 lines Enable turning off the application delimiter + warning with the 'dontwarn' option. Suggested on the -dev list, + and implemented in an alternate way by me. ........ + +2009-09-04 15:07 +0000 [r216507] 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 ........ ................ + +2009-09-04 10:49 +0000 [r216265] Russell Bryant <russell@digium.com> + + * 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 06:11 +0000 [r216223] 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 19:42 +0000 [r216011-216097] Russell Bryant <russell@digium.com> + + * UPGRADE.txt: tweak + + * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009) + | 16 lines Merged revisions 216085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r216085 | russell | 2009-09-03 14:36:46 -0500 + (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 + Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt. + ........ ................ ................ + + * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r216009 | russell | 2009-09-03 13:45:54 -0500 + (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r216008 | russell | 2009-09-03 13:44:58 -0500 + (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.2 + ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 + Sep 2009) | 2 lines Add IAX2 security document related to + AST-2009-006. ........ ................ ................ + +2009-09-03 18:40 +0000 [r216003] David Vossel <dvossel@digium.com> + + * 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, + channels/chan_iax2.c: Merged revisions 215955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | + dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines + Merge code associated with AST-2009-006 (closes issue #12912) + Reported by: rathaus Tested by: tilghman, russell, dvossel, + dbrooks ........ + +2009-09-03 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.0.15 released + + * AST-2009-006 + +2009-08-28 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.0.14 released + +2009-08-11 Tilghman Lesher <tlesher@digium.com> + + * Released 1.6.0.14-rc1 + +2009-08-10 19:51 +0000 [r211551-211587] 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 + ........ ................ + + * res/res_config_curl.c, apps/app_waitforring.c, + channels/chan_misdn.c, funcs/func_channel.c, apps/app_macro.c, + pbx/pbx_config.c, apps/app_chanspy.c, apps/app_mixmonitor.c, + res/res_odbc.c, main/asterisk.c, apps/app_voicemail.c, + doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c, + main/utils.c, cdr/cdr_pgsql.c, res/res_musiconhold.c, + apps/app_followme.c, channels/misdn_config.c, utils/frame.c, + main/channel.c, main/cdr.c, res/ael/pval.c, funcs/func_enum.c, + channels/chan_phone.c, apps/app_osplookup.c, + apps/app_setcallerid.c, main/manager.c, funcs/func_odbc.c, + apps/app_minivm.c, res/res_agi.c, res/res_config_ldap.c, + apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, + funcs/func_dialplan.c, main/dnsmgr.c, channels/chan_sip.c, + res/res_limit.c, apps/app_waitforsilence.c, agi/eagi-test.c, + apps/app_waituntil.c, main/acl.c, apps/app_queue.c, + channels/chan_oss.c, agi/eagi-sphinx-test.c, + channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c, + apps/app_sms.c, utils/extconf.c, apps/app_verbose.c, + apps/app_stack.c, apps/app_dahdibarge.c, funcs/func_rand.c, + apps/app_readfile.c, main/frame.c, /, apps/app_record.c, + funcs/func_strings.c, cdr/cdr_adaptive_odbc.c, + apps/app_alarmreceiver.c, channels/chan_iax2.c, + main/indications.c, main/config.c, main/cli.c, + pbx/pbx_loopback.c, channels/chan_dahdi.c, pbx/pbx_spool.c, + res/res_smdi.c, channels/chan_skinny.c, main/features.c, + main/http.c, main/pbx.c, apps/app_privacy.c, + codecs/codec_speex.c, funcs/func_math.c, channels/chan_agent.c, + apps/app_morsecode.c, apps/app_disa.c, funcs/func_cut.c, + channels/iax2-provision.c, pbx/dundi-parser.c, + apps/app_talkdetect.c: AST-2009-005 + +2009-08-10 14:10 +0000 [r211348] 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 [r211233-211276] 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 20:14 +0000 [r211114] 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 18:18 +0000 [r211044] 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 13:08 +0000 [r210993] 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 21:46 +0000 [r210909-210915] 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 17:47 +0000 [r210818] 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 20:07 +0000 [r210647] 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 18:57 +0000 [r210568] Leif Madsen <lmadsen@digium.com> + + * doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 + (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) + | 11 lines Update imapstorage.txt documentation. Updated the + imapstorage.txt documentation to reflect that issues with + c-client versions older than 2007 seem to cause crashing issues + that are not seen with more recent versions. Documentation has + been updated to reflect this. (closes issue #14496) Reported by: + vbcrlfuser Patches: __20090727-imap-documentation-patch.txt + uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, + dbrooks ........ ................ + +2009-08-04 14:54 +0000 [r210239] 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 ........ ................ + +2009-08-01 11:31 +0000 [r209840-209896] 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 ........ + ................ + +2009-08-01 01:13 +0000 [r209762] Kevin P. Fleming <kpfleming@digium.com> + + * channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile, + channels/misdn/ie.c, main/event.c: Merged revisions 209760-209761 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 + (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul + 2009) | 7 lines Minor changes inspired by testing with latest + GCC. The latest GCC (what will become 4.5.x) has a few new + warnings, that in these cases found some either downright buggy + code, or at least seriously poorly designed code that could be + improved. ........ ................ r209761 | kpfleming | + 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert + accidental Makefile change. ................ + +2009-07-31 21:56 +0000 [r209712] 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 16:37 +0000 [r209555-209587] David Brooks <dbrooks@digium.com> + + * channels/chan_dahdi.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 ........ + + * include/asterisk/abstract_jb.h, + contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, + codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions + 209554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | + dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines + Fixes numerous spelling errors. Patch submitted by alecdavis. + (closes issue #15595) Reported by: alecdavis ........ + +2009-07-28 12:01 +0000 [r209394] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_fax.c: Correct error in backport of latest app_fax + fixes. + +2009-07-28 00:19 +0000 [r209325] 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 [r209259-209280] 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 20:23 +0000 [r209221] David Brooks <dbrooks@digium.com> + + * channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /, + include/asterisk/module.h, main/features.c, res/res_agi.c, + res/res_jabber.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 20:16 +0000 [r209133-209198] 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 16:06 +0000 [r209061] David Brooks <dbrooks@digium.com> + + * 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 15:39 +0000 [r209057] 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 01:21 +0000 [r208925] Jeff Peeler <jpeeler@digium.com> + + * /, main/translate.c, channels/chan_iax2.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 ........ ................ + +2009-07-25 06:24 +0000 [r208752] Jeff Peeler <jpeeler@digium.com> + + * /, channels/chan_skinny.c, main/translate.c, + channels/chan_iax2.c: Merged revisions 208749 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) + | 13 lines Merged revisions 208746 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) + | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly + trivial changes, but I did not know of any other way to fix the + "dereferencing type-punned pointer will break strict-aliasing + rules" error without creating a tmp variable in chan_skinny. + ........ ................ + +2009-07-24 18:49 +0000 [r208594] 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 18:31 +0000 [r208589] 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 15:04 +0000 [r208468-208549] 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. ........ + + * UPGRADE.txt: Use correct formatting for T.38 change note in + UPGRADE.txt + + * 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, include/asterisk/frame.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 19:35 +0000 [r208389] 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 19:23 +0000 [r208384] 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 [r208264-208316] 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 ........ ................ + +2009-07-21 22:47 +0000 [r207947] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500 + (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) + | 8 lines Force an error if a blank is passed to QUOTE (because + the documentation states the argument is not optional). This + change makes URIENCODE and QUOTE behave similarly, since the + documentation states that the argument is not optional, for both. + (closes issue #15439) Reported by: pkempgen Patches: + 20090706__issue15439.diff.txt uploaded by tilghman (license 14) + ........ ................ + +2009-07-21 20:27 +0000 [r207783-207860] 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 r207636, this approach could + potentially block for an unacceptable amount of time. + +2009-07-21 14:30 +0000 [r207725] 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 13:39 +0000 [r207683] Kevin P. Fleming <kpfleming@digium.com> + + * funcs/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile, + Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, /, + main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, + Makefile.rules, pbx/Makefile, res/Makefile, channels/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 04:38 +0000 [r207636] 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 19:55 +0000 [r207425] 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 16:37 +0000 [r207362] 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 01:35 +0000 [r207286] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn/isdn_lib.c, channels/misdn_config.c, + channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, + doc/tex/misdn.tex, channels/chan_misdn.c, main/callerid.c, + configs/misdn.conf.sample: Merged revisions 145293,158010 from + https://origsvn.digium.com/svn/asterisk/branches/1.4 to make + merging easier. These changes are already on trunk. + ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 + (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c + channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk + to make merging easier later. ........ r145200 | rmudgett | + 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * + Miscellaneous formatting changes to make v1.4 and trunk more + merge compatible in the mISDN area. channels/chan_misdn.c * + Eliminated redundant code in cb_events() EVENT_SETUP ........ + r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) + | 9 lines improved helptext of misdn_set_opt. ........ r142181 | + rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line + Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 + 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines + channels/chan_misdn.c * Made bearer2str() use + allowed_bearers_array[] * Made use the causes.h defines instead + of hardcoded numbers. * Made use Asterisk presentation indicator + values if either of the mISDN presentation or screen options are + negative. * Updated the misdn_set_opt application option + descriptions. * Renamed the awkward Caller ID presentation + misdn_set_opt application option value not_screened to + restricted. Deprecated the not_screened option value. + channels/misdn/isdn_lib.c * Made use the causes.h defines instead + of hardcoded numbers. * Fixed some spelling errors and typos. * + Added all defined facility code strings to fac2str(). + channels/misdn/isdn_lib.h * Added doxygen comments to struct + misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen + comments to struct misdn_stack. channels/misdn_config.c + configs/misdn.conf.sample * Updated the mISDN presentation and + screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) + * Updated the misdn_set_opt application option descriptions. * + Fixed some spelling errors and typos. ................ r158010 | + rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines + Merged revision 157977 from + https://origsvn.digium.com/svn/asterisk/team/group/issue8824 + ........ Fixes JIRA ABE-1726 The dial extension could be empty if + you are using MISDN_KEYPAD to control ISDN provider features. + ................ + +2009-07-17 19:38 +0000 [r207097-207157] 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 17:53 +0000 [r206871-207032] 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 ........ ................ + + * 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 16:52 +0000 [r206809] 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 22:06 +0000 [r206775] 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 21:34 +0000 [r206762] 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 20:21 +0000 [r206705] 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 16:02 +0000 [r206637] 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 20:22 +0000 [r206585] Tilghman Lesher <tlesher@digium.com> + + * /, contrib/scripts/meetme.sql: Recorded merge of revisions 206567 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 + Jul 2009) | 6 lines Document all meetme realtime fields, and in + the process, make some field lengths more consistent. (closes + issue #15493) Reported by: lasko Patches: meetme.diff uploaded by + lasko (license 833) ........ + +2009-07-14 18:17 +0000 [r206555] 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 14:54 +0000 [r206387] 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 01:25 +0000 [r206369] 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 + ........ ................ + +2009-07-10 22:50 +0000 [r206017] 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 18:44 +0000 [r205940] 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 17:44 +0000 [r205879-205880] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Fix build. + + * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul + 2009) | 30 lines Merged revisions 205877 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 + (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 + (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul + 2009) | 10 lines Ensure that outbound NOTIFY requests are + properly routed through stateful proxies. With this change, we + make note of Record-Route headers present in any SUBSCRIBE + request that we receive so that our outbound NOTIFY requests will + have the proper Route headers in them. (closes issue #14725) + Reported by: ibc ........ ................ ................ + ................ + +2009-07-10 16:48 +0000 [r205843] 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 15:57 +0000 [r205777] 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 15:35 +0000 [r205771] 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 23:46 +0000 [r205729] 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 21:26 +0000 [r205697] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c, apps/app_fax.c, include/asterisk/frame.h: + 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 16:21 +0000 [r205597-205608] 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. ........ ................ + + * main/rtp.c, /, channels/chan_iax2.c, include/asterisk/frame.h: + 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 08:32 +0000 [r205533] 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 22:17 +0000 [r205415] David Vossel <dvossel@digium.com> + + * include/asterisk/devicestate.h, main/pbx.c, /, + main/devicestate.c, include/asterisk/pbx.h: Merged revisions + 205412 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) + | 12 lines Merged revisions 205409 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) + | 6 lines moving ast_devstate_to_extenstate to pbx.c from + devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This + change fixes a compile time error with chan_vpb as well. ........ + ................ + +2009-07-08 19:27 +0000 [r205351] 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 18:20 +0000 [r205296] 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 17:01 +0000 [r205224] Tilghman Lesher <tlesher@digium.com> + + * main/say.c: oops, fixing build + +2009-07-08 16:56 +0000 [r205220] 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 16:28 +0000 [r205200] 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 15:56 +0000 [r205139-205152] 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. ........ + + * main/asterisk.c, /, main/Makefile, res/res_crypto.c, main/ssl.c + (added), include/asterisk/_private.h, res/res_jabber.c: 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. ........ + +2009-07-08 14:35 +0000 [r205117] David Vossel <dvossel@digium.com> + + * channels/chan_sip.c, include/asterisk/sched.h: SIP Dialog ref + counting This patch adds reference counting for sip dialogs into + 1.6.0. When proc_session_timer() is called from the scheduler + thread it has no guarantee the session timer's dialog won't be + freed from underneath it. Now the session timer holds a reference + to the dialog, preventing it from being destroyed during the + middle of proc_session_timer(). (closes issue #13623) Reported + by: Nik Soggia Review: https://reviewboard.asterisk.org/r/302/ + +2009-07-06 15:17 +0000 [r204980] Tilghman Lesher <tlesher@digium.com> + + * main/say.c: Restore Hungarian (mistakenly removed during merge) + +2009-07-06 13:39 +0000 [r204949] 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 22:03 +0000 [r204836] 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 18:07 +0000 [r204652-204754] 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/ ........ ................ + + * channels/chan_sip.c: removes fake dialog_unref and dialog_ref + function calls. dialog_unref() and dialog_ref() in 1.6.0 where + only place holders for reference counting once it was + implemented. The functions did nothing but return the pointer on + ref and NULL on unref. These calls have been removed to make way + for a patch that actually does dialog ref counting. + +2009-06-30 21:21 +0000 [r204581] 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 18:50 +0000 [r204476] 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 [r204471] 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 ........ + ................ + +2009-06-29 22:52 +0000 [r204248-204302] 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 01:14 +0000 [r203910] 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 22:12 +0000 [r203855] 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 21:25 +0000 [r203780-203818] 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 20:16 +0000 [r203722] 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 19:54 +0000 [r203717] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: reverse whitespace change 203711 that was + based on looking at sig_analog (which has about a 1000 line + indentation change that is not worth doing here) + +2009-06-26 19:49 +0000 [r203716] 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 19:47 +0000 [r203711] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: whitespace fix + +2009-06-26 19:29 +0000 [r203701] Joshua Colp <jcolp@digium.com> + + * main/rtp.c, main/channel.c, main/frame.c, /, channels/chan_sip.c, + apps/app_fax.c, configs/sip.conf.sample, + include/asterisk/frame.h: Merged revisions 203699 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 + lines Improve T.38 negotiation by exchanging session parameters + between application and channel. ........ + +2009-06-26 19:25 +0000 [r203698] 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 ........ + +2009-06-25 21:47 +0000 [r203447] 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 21:18 +0000 [r203387] 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 21:06 +0000 [r203117-203377] 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 ........ ................ + + * /, 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 21:18 +0000 [r203044] 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 18:29 +0000 [r202968] 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 18:09 +0000 [r202926] 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 22:09 +0000 [r202763] 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 21:26 +0000 [r202754] Ryan Brindley <rbrindley@digium.com> + + * main/config.c, /: Merged revisions 202753 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 | + rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9 + lines If we delete the info, lets also delete the lines (closes + issue #14509) Reported by: timeshell Patches: + 20090504__bug14509.diff.txt uploaded by tilghman (license 14) + Tested by: awk, timeshell ........ + +2009-06-23 16:40 +0000 [r202675] 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 20:12 +0000 [r202498] 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 ........ ................ + +2009-06-22 16:30 +0000 [r202471] 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 16:06 +0000 [r202416] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) + | 9 lines Merged revisions 202414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) + | 2 lines Make Polycom subscription type override check more + explicit. ........ ................ + +2009-06-22 15:05 +0000 [r202338-202344] 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 16:14 +0000 [r202259-202263] 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 [r202184] 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 21:07 +0000 [r202006] 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 20:27 +0000 [r201997] 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 00:44 +0000 [r201786-201830] Tilghman Lesher <tlesher@digium.com> + + * /, 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 16:58 +0000 [r201682] David Vossel <dvossel@digium.com> + + * channels/misdn/isdn_lib.c, main/asterisk.c, utils/conf2ael.c, + main/ast_expr2.c, utils/stereorize.c, + codecs/gsm/src/gsm_destroy.c, /, channels/h323/ast_h323.cxx, + main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c, + utils/extconf.c, pbx/pbx_config.c, res/res_config_ldap.c: 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 15:32 +0000 [r201612] 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) + ........ ................ + +2009-06-17 20:10 +0000 [r201459-201463] 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 19:55 +0000 [r201449] David Vossel <dvossel@digium.com> + + * 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 19:35 +0000 [r201443] 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 12:05 +0000 [r201263] 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 22:31 +0000 [r201226] 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 19:34 +0000 [r201093] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c, + main/autoservice.c, main/frame.c, /, apps/app_meetme.c, + configure, main/slinfactory.c, autoconf/ast_gcc_attribute.m4, + configure.ac, include/asterisk/linkedlists.h, main/file.c, + include/asterisk/channel.h, include/asterisk/frame.h: Merged + revisions 201056,201090 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/ ........ + ................ 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. ................ + +2009-06-16 17:11 +0000 [r200992] 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 16:34 +0000 [r200986] Kevin P. Fleming <kpfleming@digium.com> + + * /, 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 16:02 +0000 [r200945] 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 01:33 +0000 [r200724-200767] 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, configs/sip.conf.sample: Merged revisions + 165180,200689 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | + mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 + lines This patch adds a new 'ignoresdpversion' option to + sip.conf. When this is enabled (either globally or for a specific + peer), chan_sip will treat any SDP data it receives as new data + and update the media stream accordingly. By default, Asterisk + will only modify the media stream if the SDP session version + received is different from the current SDP session version. This + option is required to interoperate with devices that have + non-standard SDP session version implementations (observed by toc + on the bug tracker with Microsoft OCS which always uses 0 as the + session version). http://reviewboard.digium.com/r/94/ (closes + issue #13958) Reported by: toc Tested by: toc ........ r200689 | + kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 + lines Accept T.38 re-INVITE responses with invalid SDP versions. + This commit changes the 'incoming SDP version' check logic a bit + more; when 'ignoresdpversion' is *not* set for a peer, if we + initiate a re-INVITE to switch to T.38, we'll always accept the + peer's SDP response, even if they don't properly increment the + SDP version number as they should. If this situation occurs, a + warning message will be generated suggesting that the peer's + configuration be changed to include the 'ignoresdpversion' + configuration option (although ideally they'd fix their SIP + implementation to be RFC compliant). AST-221 ........ + +2009-06-15 15:22 +0000 [r200515] 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 ........ ................ + +2009-06-12 19:08 +0000 [r200362] 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 ........ ................ + +2009-06-11 22:42 +0000 [r200228] Sean Bright <sean@malleable.com> + + * Makefile, /: Merged revisions 199781 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 | + seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 + lines Fix all of the parallel build warnings issued when running + make -j#. ........ + +2009-06-11 21:18 +0000 [r200149] 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 12:16 +0000 [r200040] Leif Madsen <lmadsen@digium.com> + + * /, build_tools/make_version_c, build_tools/make_version_h: 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 20:29 +0000 [r199994] 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 20:20 +0000 [r199975] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: The 1.6.0 branch was missing all + invite_branch logic. It has now been added. + +2009-06-10 16:13 +0000 [r199858] Sean Bright <sean@malleable.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 20:54 +0000 [r199821] 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 ........ + +2009-06-08 19:39 +0000 [r199632] Sean Bright <sean@malleable.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. ........ ................ + +2009-06-05 21:37 +0000 [r199301] David Vossel <dvossel@digium.com> + + * main/pbx.c, /: Merged revisions 199298 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) + | 21 lines Merged revisions 199297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) + | 14 lines Fixes issue with hints giving unexpected results. + Hints with two or more devices that include ONHOLD gave + unexpected results. (closes issue #15057) Reported by: + p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel + (license 671) pbx.c.1.4.patch uploaded by p (license 558) + devicestate.c.trunk.patch uploaded by p (license 671) Tested by: + p_lindheimer, dvossel Review: + https://reviewboard.asterisk.org/r/254/ ........ ................ + +2009-06-05 13:51 +0000 [r199228] 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 19:16 +0000 [r199142] 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 ........ + ................ + +2009-08-11 Tilghman Lesher <tlesher@digium.com> + + * Asterisk 1.6.0.13 released + + * channels/chan_sip.c: Bad merge from 1.6.0 branch + +2009-08-10 Tilghman Lesher <tlesher@digium.com> + + * Asterisk 1.6.0.12 released + + * AST-2009-005 + +2009-06-05 Leif Madsen <lmadsen@digium.com> + + * Asterisk 1.6.0.10 released + +2009-06-04 David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: Additional updates for AST-2009-001 + +2009-06-04 David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001 + +2009-04-06 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.9 + +2009-04-03 16:27 -0500 [r186517] Mark Michelson <mmichelson@digium.com> + + * Remove an invalid call to free memory. + + A bad merge from trunk to 1.6.0 meant freeing memory that + should not be freed. In trunk, pkt->data is an ast_str, but + in 1.6.0, it is allocated in the same chunk of memory as the + sip_pkt. This only affects 1.6.0. + + (closes issue #14819) + Reported by: cwolff09 + +2009-04-02 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.8 + +2009-04-02 Tilghman Lesher <tlesher@digium.com> + + * Fix for security issue AST-2009-003 + +2009-03-30 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.7 + +2009-03-19 Leif Madsen <lmadsen@digium.com> + + * Release Asterisk 1.6.0.7-rc2 + +2009-03-19 15:40 +0000 [r183066-183109] 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 [r183029] 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 14:24 +0000 [r182945] 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 20:51 +0000 [r182723] Jeff Peeler <jpeeler@digium.com> + + * 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, channels/h323/chan_h323.h: Merged + revisions 182722 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | + jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines + Allow H.323 Plus library to be used in addition to the OpenH323 + library Chan_h323 can now be compiled against both the previously + supported versions of OpenH323 as well as the current H.323 Plus + (version 1.20.2). The configure script has been modified to look + in the default install location of h323 to hopefully help avoid + using the environment variables OPENH323DIR and PWLIBDIR. Also, + the CLI command "h323 show version" has been added which + indicates which version of h323 is in use. (closes issue #11261) + Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch + uploaded by jthurman (license 614) ........ + +2009-03-17 15:27 +0000 [r182569] 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 15:00 +0000 [r182526-182532] 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 05:53 +0000 [r182451] 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 ........ ................ + +2009-03-16 17:52 +0000 [r182283] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 182282 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 + (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) + | 7 lines Randomize IAX2 encryption padding The 16-32 byte random + padding at the beginning of an encrypted IAX2 frame turns out to + not be all that random at all. This patch calls ast_random to + fill the padding buffer with random data. The padding is + randomized at the beginning of every encrypted call and for every + encrypted retransmit frame. Review: + http://reviewboard.digium.com/r/193/ ........ ................ + +2009-03-16 17:36 +0000 [r182212-182279] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_env.c: Merged revisions 182278 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182278 | + tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines + Fix an off-by-one error in the FILE() function, and extend + FILE()'s length parameter to work like variable substitution. + Previously, FILE() returned one less character than specified, + due to the terminating NULL. Both the offset and length + parameters now behave identically to the way variable + substitution offsets and lengths also work. (closes issue #14670) + Reported by: BMC ........ + + * channels/chan_local.c, /: Merged revisions 182211 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r182211 | tilghman | 2009-03-16 10:50:55 -0500 + (Mon, 16 Mar 2009) | 14 lines Merged revisions 182208 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) + | 7 lines Fixup glare detection, to fix a memory leak of a local + pvt structure. (closes issue #14656) Reported by: caspy Patches: + 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) + Tested by: caspy ........ ................ + +2009-03-16 13:59 +0000 [r182172] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 182171 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 | + file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix + a memory leak in the ast_answer / __ast_answer API call. For a + channel that is not yet answered this API call will wait until a + voice frame is received on the channel before returning. It does + this by waiting for frames on the channel and reading them in. + The frames read in were not freed when they should have been. + ........ + +2009-03-13 21:26 +0000 [r182064-182122] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 182121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 | + mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 + lines Change faulty comparison used when announcing average hold + minutes and seconds (closes issue #14227) Reported by: caspy + ........ + + * /, main/features.c: Merged revisions 182029 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar + 2009) | 41 lines Merged revisions 181990 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar + 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and + peer when interpreting DTMF. Dynamic features defined in the + applicationmap section of features.conf allow one to specify + whether the caller, callee, or both have the ability to use the + feature. The documentation in the features.conf.sample file could + be interpreted to mean that one only needs to set the + DYNAMIC_FEATURES channel variable on the calling channel in order + to allow for the callee to be able to use the features which he + should have permission to use. However, the DYNAMIC_FEATURES + variable would only be read from the channel of the participant + that pressed the DTMF sequence to activate the feature. The + result of this was that the callee was unable to use dynamic + features unless the dialplan writer had taken measures to be sure + that the DYNAMIC_FEATURES variable was set on the callee's + channel. This commit changes the behavior of + ast_feature_interpret to concatenate the values of + DYNAMIC_FEATURES from both parties involved in the bridge. The + features themselves determine who has permission to use them, so + there is no reason to believe that one side of the bridge could + gain the ability to perform an action that they should not have + the ability to perform. Kevin Fleming pointed out on the + asterisk-users list that the typical way that this was worked + around in the past was by setting _DYNAMIC_FEATURES on the + calling channel so that the value would be inherited by the + called channel. While this works, the documentation alone is not + enough to figure out why this is necessary for the callee to be + able to use dynamic features. In this particular case, changing + the code to match the documentation is safe, easy, and will + generally make things easier for people for future installations. + This bug was originally reported on the asterisk-users list by + David Ruggles. (closes issue #14657) Reported by: mmichelson + Patches: 14657.patch uploaded by mmichelson (license 60) ........ + ................ + +2009-03-13 17:28 +0000 [r182036] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | + file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix + an issue with requesting a T38 reinvite before the call is + answered. The code responsible for sending the T38 reinvite did + not check if an INVITE was already being handled. This caused + things to get confused and the call to fail. The code now defers + sending the T38 reinvite until the current INVITE is done being + handled. (issue AST-191) ........ + +2009-03-12 21:44 +0000 [r181770-181848] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 181846 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 | + mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 + lines Run the macro on the queue member's channel when he + answers, not the caller's channel. ........ + + * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar + 2009) | 28 lines Merged revisions 181768 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar + 2009) | 22 lines Properly send a 487 on an INVITE we have not + responded to if we receive a BYE. If we receive an INVITE from an + endpoint and then later receive a BYE from that same endpoint + before we have sent a final response for the INVITE, then we need + to respond to the INVITE with a 487. There was logic in the code + prior to this commit which seemed to exist solely to handle this + situation, but there was one condition in an if statement which + was incorrect. The only way we would send a 487 was if the + sip_pvt had no owner channel. This made no sense since we created + the owner channel when we received the INVITE, meaning that the + majority of the time we would never send the 487. The 487 being + sent should not rely on whether we have created a channel. Its + delivery should be dependent on the current state of the initial + INVITE transaction. With this commit, that logic is now correctly + in place. (closes issue #14149) Reported by: legranjl Patches: + 14149.patch uploaded by mmichelson (license 60) Tested by: + legranjl ........ ................ + +2009-03-12 17:58 +0000 [r181732] Tilghman Lesher <tlesher@digium.com> + + * /, configure, main/translate.c: Merged revisions 181731 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 + Mar 2009) | 9 lines Adjust translation table column widths based + upon the translation times. Previously, only 5 columns were + displayed, and if a translation time exceeded 99,999 useconds, it + would be displayed as 0, instead of its actual time. (closes + issue #14532) Reported by: pj Patches: + 20090311__bug14532.diff.txt uploaded by tilghman (license 14) + Tested by: pj ........ + +2009-03-12 16:57 +0000 [r181613-181666] Joshua Colp <jcolp@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, + 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 + lines Fix incorrect usage of strncasecmp... I really meant to use + strcasecmp. ........ ................ + + * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, + 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 + lines Fix another scenario where depending on configuration the + stream would not get read. For custom commands we don't know + whether the audio is coming from a stream or not so we are going + to have to read the data despite no channels. (closes issue + #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 + 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in + previous commit. ........ ................ + + * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, + 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | + 10 lines Fix issue with streaming MOH failing if nobody is + listening. When a music class is setup to actually provide music + on hold from a stream we need to constantly read audio from it + since it will constantly be providing audio. This is now done + despite there being no channels listening to it. (closes issue + #14416) Reported by: caspy ........ ................ + + * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | + file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix + crash when sleep and retries argument was not given to RetryDial + application. (closes issue #14647) Reported by: sherpya ........ + +2009-03-12 01:04 +0000 [r181543] Richard Mudgett <rmudgett@digium.com> + + * /, build_tools/make_version: Merged revisions 181542 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) + | 1 line Use the correct branch integrated property when + generating the version string ........ + +2009-03-11 23:19 +0000 [r181509] Michiel van Baak <michiel@vanbaak.info> + + * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk Provide + correct hint to debug SIP trouble in the default config (closes + issue #14646) Reported by: strk + +2009-03-11 22:22 +0000 [r181450] Jason Parker <jparker@digium.com> + + * /, configure, configure.ac: Merged revisions 181444 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181444 | qwell | 2009-03-11 17:20:13 -0500 + (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | + 4 lines Allow prefix to set localstatedir (when used and + different from the default). This is similar to the /etc change + that was made for the non-FreeBSD case. ........ ................ + +2009-03-11 22:15 +0000 [r181425-181429] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 181428 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | + russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines + Make handling of the BRIDGEPVTCALLID variable thread-safe. + ........ + + * main/channel.c, /: Merged revisions 181424 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) + | 17 lines Merged revisions 181423 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) + | 9 lines Make code that updates BRIDGEPEER variable thread-safe. + It is not safe to read the name field of an ast_channel without + the channel locked. This patch fixes some places in channel.c + where this was being done, and lead to crashes related to + masquerades. (closes issue #14623) Reported by: guillecabeza + ........ ................ + +2009-03-11 17:37 +0000 [r181372] David Vossel <dvossel@digium.com> + + * channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions + 181371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) + | 17 lines Merged revisions 181340 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) + | 11 lines encrypted IAX2 during packet loss causes decryption to + fail on retransmitted frames If an iax channel is encrypted, and + a retransmit frame is sent, that packet's iseqno is updated while + it is encrypted. This causes the entire frame to be corrupted. + When the corrupted frame is sent, the other side decrypts it and + sends a VNAK back because the decrypted frame doesn't make any + sense. When we get the VNAK, we look through the sent queue and + send the same corrupted frame causing a loop. To fix this, + encrypted frames requiring retransmission are decrypted, updated, + then re-encrypted. Since key-rotation may change the key held by + the pvt struct, the keys used for encryption/decryption are held + within the iax_frame to guarantee they remain correct. (closes + issue #14607) Reported by: stevenla Tested by: dvossel Review: + http://reviewboard.digium.com/r/192/ ........ ................ + +2009-03-11 17:28 +0000 [r181297-181352] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | + 21 lines Merged revisions 181328 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | + 14 lines Fix issue where an attended transfer could not be + completed under a rare scenario. When completing an attended + transfer chan_sip does a check to make sure the extension in the + URI portion of the Refer-To header is a local valid extension. We + don't actually need to check this since we know for sure the + other channel is already up and talking to the extension. Some + devices do not put the extension in the Refer-To header either, + which can cause the extension check to fail. We now no longer do + this check if it is an attended transfer. (closes issue #14628) + Reported by: sverre Patches: 14628.diff uploaded by file (license + 11) ........ ................ + + * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | + 16 lines Merged revisions 181295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 + lines Fix a problem with inband DTMF detection on outgoing SIP + calls when dtmfmode=auto. When dtmfmode was set to auto the + inband DTMF detector was not setup on outgoing SIP calls. This + caused inband DTMF detection to fail. The inband DTMF detector is + now setup for both dtmfmode inband and auto. (closes issue + #13713) Reported by: makoto ........ ................ + +2009-03-11 15:54 +0000 [r181137-181284] Jeff Peeler <jpeeler@digium.com> + + * channels/h323/ast_h323.cxx: add missing header file + + * utils/extconf.c: Fix merge oops from 181137 + + * utils/Makefile, include/asterisk/utils.h, + include/asterisk/astmm.h, /, channels/chan_sip.c, + channels/h323/ast_h323.cxx, utils/extconf.c: Merged revisions + 181135 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | + jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines + Fix malloc debug macros to work properly with h323. The main + problem here was that cstdlib was undefining free thereby causing + the proper debug macros to not be used. ast_h323.cxx has been + changed to call ast_free instead to avoid the issue. A few other + issues were addressed: - There were a few instances of functions + improperly passing ast_free instead of ast_free_ptr. - Some clean + up was done to avoid the debug macros intentionally being + redefined. (copied below from Kevin's commit, appreciate the + help) - disable astmm.h from doing anything when STANDALONE is + defined, which is used by the tools in the utils/ directory that + use parts of Asterisk header files in hackish ways; also ensure + that utils/extconf.c and utils/conf2ael.c are compiled with + STANDALONE defined. (closes issue #13593) Reported by: pj + ........ + +2009-03-11 00:52 +0000 [r181034] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 181032-181033 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 + (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar + 2009) | 9 lines Fix incorrect tag checking on transfers when + pedantic=yes is enabled. (closes issue #14611) Reported by: + klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt + uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ + r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar + 2009) | 3 lines Remove unused variables. ........ + ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 + (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC + 3891 ................ + +2009-03-10 22:05 +0000 [r180946] Jason Parker <jparker@digium.com> + + * /, configure, configure.ac, autoconf/ast_prog_sed.m4, + autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r180944 | qwell | 2009-03-10 17:03:41 -0500 + (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar + 2009) | 1 line Make things happier when using autoconf 2.62+ + ........ ................ + +2009-03-10 14:41 +0000 [r180718-180801] Joshua Colp <jcolp@digium.com> + + * main/manager.c, /: Merged revisions 180800 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | + file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines + Reset the thread local string buffer when handling the UserEvent + action. (closes issue #14593) Reported by: JimDickenson ........ + + * channels/chan_sip.c: If a port is specified when dialing a peer + then use it. (closes issue #14626) Reported by: acunningham + + * channels/chan_sip.c: Ensure that the new outgoing dialog to a + peer is able to set the socket details, even if the default is + present. (closes issue #14480) Reported by: jon + +2009-03-06 18:26 +0000 [r180582] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600 + (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, + 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when + IMAP storage is enabled. ........ ................ + +2009-03-06 Leif Madsen <lmadsen@digium.com> + + * Release 1.6.0.7-rc1 + +2009-03-06 17:28 +0000 [r180535] David Vossel <dvossel@digium.com> + + * main/enum.c, /: Merged revisions 180534 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) + | 15 lines Merged revisions 180532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) + | 9 lines Fix handling of backreferences for ENUM lookups enum.c + did not handle regex backtraces correctly. The '\1' in the regex + is a backreference that requires a pattern match to be inserted. + The way the code used to work is that it would find the + backreference and insert the entire input string minus the '+'. + This is incorrect. The regexec() function takes in a variable + called pmatch which is an array of structs containing the start + and end indexes for each backreference substring. The original + code actually passed the pmatch array pointer into regexec but + never did anything with it. Now when a backtrace is found, the + backtrace number is looked up in the pmatch array and the correct + substring is inserted. (closes issue #14576) Reported by: + chris-mac Review: http://reviewboard.digium.com/r/187/ ........ + ................ + +2009-03-05 23:28 +0000 [r180404-180466] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600 + (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar + 2009) | 16 lines [IMAP] Fix message retrieval issues when + identical mailbox names were defined in separate contexts. There + was a fix put in a while back so that an X-Asterisk-VM-Context + message header was added to stored IMAP voicemails. This would + allow for us to differentiate if the same mailbox name was used + in multiple contexts. The problem still left was that not all + places where messages were retrieved actually attempted to use + this header for information when retrieving messages. This commit + fixes that so that MWI and message retrieval from VoiceMailMain + work as expected. (closes issue #13853) Reported by: vicks1 + Patches: 13853_v2.patch uploaded by mmichelson (license 60) + Tested by: lmadsen ........ ................ + + * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged + revisions 180383 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar + 2009) | 31 lines Merged revisions 180380 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar + 2009) | 25 lines Fix broken mailbox parsing when searchcontexts + option is enabled. When using the searchcontexts option in + voicemail.conf, the code made the assumption that all mailbox + names defined were unique across all contexts. However, the code + did nothing to actually enforce this assumption, nor did it do + anything to alert a user that he may have created an ambiguity in + his voicemail.conf file by defining the same mailbox name in + multiple contexts. With this change, we now will issue a nice + long warning if searchcontexts is on and we encounter the same + mailbox name in multiple contexts and ignore any duplicates after + the first box. Whether searchcontexts is enabled or not, if we + come across a duplicate mailbox in the same context, then we will + issue a warning and ignore the duplicated mailbox. I have also + added a small note to voicemail.conf.sample in the explanation + for searchcontexts explaining that you cannot define the same + mailbox in multiple contexts if you have enabled the option. + (closes issue #14599) Reported by: lmadsen Patches: 14599.patch + uploaded by mmichelson (license 60) (with slight modification) + Tested by: lmadsen ........ ................ + +2009-03-05 18:36 +0000 [r180377] Kevin P. Fleming <kpfleming@digium.com> + + * main/rtp.c, main/frame.c, /, include/asterisk/frame.h: Merged + revisions 180373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar + 2009) | 15 lines Merged revisions 180372 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar + 2009) | 9 lines Fix problems when RTP packet frame size is + changed During some code analysis, I found that calling + ast_rtp_codec_setpref() on an ast_rtp session does not work as + expected; it does not adjust the smoother that may on the RTP + session, in fact it summarily drops it, even if it has data in + it, even if the current format's framing size has not changed. + This is not good. This patch changes this behavior, so that if + the packetization size for the current format changes, any + existing smoother is safely updated to use the new size, and if + no smoother was present, one is created. A new API call for + smoothers, ast_smoother_reconfigure(), was required to implement + these changes. Review: http://reviewboard.digium.com/r/184/ + ........ ................ + +2009-03-04 19:25 +0000 [r180121-180196] Joshua Colp <jcolp@digium.com> + + * /, main/callerid.c: Merged revisions 180195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | + 11 lines Merged revisions 180194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 + lines Look for the number in a callerid string starting from the + end. This way a value using <> can exist in the name portion. + (issue #AST-194) ........ ................ + + * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 | + file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines + Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) + Reported by: alecdavis Patches: app_dial.optionk.diff.txt + uploaded by alecdavis (license 585) ........ + +2009-03-03 23:35 +0000 [r180078] David Vossel <dvossel@digium.com> + + * main/channel.c, include/asterisk/app.h, apps/app_read.c, /, + main/app.c: Merged revisions 180032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | + dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines + app_read does not break from prompt loop with user terminated + empty string In app.c, ast_app_getdata is called to stream the + prompts and receive DTMF input. If ast_app_getdata() receives an + empty string caused by the user inputing the end of string + character, in this case '#', it should break from the prompt loop + and return to app_read, but instead it cycles through all the + prompts. I've added a return value for this special case in + ast_readstring() which uses an enum I've delcared in apps.h. This + enum is now used as a return value for ast_app_getdata(). (closes + issue #14279) Reported by: Marquis Patches: fix_app_read.patch + uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded + by dvossel (license 671) Tested by: Marquis, dvossel Review: + http://reviewboard.digium.com/r/177/ ........ + +2009-03-03 23:26 +0000 [r180058] Steve Murphy <murf@digium.com> + + * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile, + utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y, + main/ast_expr2f.c: Merged revisions 179973 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | + 33 lines Merged revisions 179807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some + work to do to port these changes to trunk; the check_expr stuff + hasn't been updated here for quite some time, it appears. I added + some more tests to the check_expr2 suite. I had to play around + with the makefile a bit, etc. I added STANDALONE2 #ifdefs to + ast_expr2.y so as not to conflict structure with aelparse. + ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar + 2009) | 19 lines These changes allow AEL to better check ${} + constructs within $[...], that are concatenated with text. I + modified and added rules in ast_expr2.fl to better handle the + concatenations. I added some default routines to ast_expr2.y so + the standalone would compile. It also looks like I haven't run + this thru bison since 2.1, so it's good to get this updated. The + Makefile has comments added now for check_expr2 and check_expr to + explain what they are for, and how to run them. The testexpr2s + stuff has been removed, in favor of check_expr2. expr2.testinput + has been updated to include the two expressions that inspired + these changes (from mcnobody on #asterisk this morning) The + regression has been run and all looks well. ........ + ................ + +2009-03-03 22:49 +0000 [r179971-180008] Mark Michelson <mmichelson@digium.com> + + * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions + 180007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar + 2009) | 22 lines Merged revisions 180006 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar + 2009) | 17 lines Clarify some documentation of queues.conf.sample + It had always been possible to explicitly specify a "blank" value + for a sound file in queues.conf and have no sound played back. + The problem with this is that it would result in some ugly CLI + warnings from file.c. This commit introduces a check when playing + a file in app_queue to see if the name of the file is zero-length + and return early if that is the case. Also, the ability to + specify the blank sound files in queues.conf is now mentioned + more clearly in queues.conf.sample (closes issue #14227) Reported + by: caspy ........ ................ + + * apps/app_queue.c: Fix a memory leak when updating a realtime + member field. This was discovered while looking at issue #14353 + +2009-03-03 18:29 +0000 [r179842] Joshua Colp <jcolp@digium.com> + + * /, main/features.c: Merged revisions 179841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | + 16 lines Merged revisions 179840 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 + lines Do not assume that the bridge_cdr is still attached to the + channel when the 'h' exten is finished executing. It is possible + for a masquerade operation to occur when the 'h' exten is + operating. This operation moves the CDR records around causing + the bridge_cdr to no longer exist on the channel where it is + expected to. We can not safely modify it afterwards because of + this, so don't even try. (closes issue #14564) Reported by: meric + ........ ................ + +2009-03-03 16:48 +0000 [r179743] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 179742 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) + | 14 lines Merged revisions 179741 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) + | 6 lines Ensure chan->fdno always gets reset to -1 after + handling a channel fd event. Since setting fdno to -1 had to be + moved, a couple of other code paths that do process an fd event + return early and do not pass through the code path where it was + moved to. So, set it to -1 in a few other places, too. ........ + ................ + +2009-03-03 14:40 +0000 [r179673] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 179672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | + 10 lines Merged revisions 179671 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 + lines Move where fdno is set to the default value to *after* the + read callback of the channel driver is called. We have to do this + as the underlying channel driver may need the fdno value to + determine what to read. ........ ................ + +2009-03-03 13:55 +0000 [r179610] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 179609 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) + | 17 lines Merged revisions 179608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) + | 9 lines Make it easier to detect an improper call to + ast_read(). When you call ast_waitfor() on a channel, the index + into the channel fds array that holds the file descriptor that + poll() determines has input available is stored in fdno. This + patch clears out this value after a call to ast_read() and also + reports errors if ast_read() is called without an fdno set. From + a discussion on the asterisk-dev list. ........ ................ + +2009-03-03 00:03 +0000 [r179538] Jeff Peeler <jpeeler@digium.com> + + * main/channel.c, /: Merged revisions 179537 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) + | 21 lines Merged revisions 179536 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) + | 15 lines Fix bridging regression from commit 176701 This fixes + a bad regression where the bridge would exit after an attended + transfer was made. The problem was due to nexteventts getting set + after the masquerade which caused the bridge to return + AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: + tim_ringenbach ........ ................ + +2009-03-02 23:38 +0000 [r179534] Russell Bryant <russell@digium.com> + + * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) + | 48 lines Merged revisions 179532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) + | 40 lines Move ast_waitfor() down to avoid the results of the + API call becoming stale. This call to ast_waitfor() was being + done way too soon in this section of code. Specifically, there + was code in between the call to waitfor and the code that uses + the result that puts the channel in autoservice. By putting the + channel in autoservice, the previous results of ast_waitfor() + become meaningless, as the autoservice thread will do it's own + ast_waitfor() and ast_read() on the channel. So, when we came + back out of autoservice and eventually hit the block of code that + calls ast_read() on the channel, there may not actually be any + input on the channel available. Even though the previous call to + ast_waitfor() in app_meetme said there was input, the autoservice + thread has since serviced the channel for some period of time. + This bug manifested itself while dvossel was doing some testing + of MeetMe in Asterisk trunk. He was using the timerfd timing + module. When the code hit ast_read() erroneously, it determined + that it must have been called because of input on the timer fd, + as chan->fdno was set to AST_TIMING_FD, since that was the cause + of the last legitimate call to ast_read() done by autoservice. In + this test, an IAX2 channel was calling into the MeetMe + conference. It was _much_ more likely to be seen with an IAX2 + channel because of the way audio is handled. Every audio frame + that comes in results in a call to ast_queue_frame(), which then + uses ast_timer_enable_continuous() to notify the channel thread + that a frame is waiting to be handled. So, the chances of + ast_waitfor() indicating that a channel needs servicing due to a + timer event on an IAX2 event is very high. Finally, it is + interesting to note that if a different timing interface was + being used, this bug would probably not be noticed. When + ast_read() is called and erroneously thinks that there is a timer + event to handle, it calls the ast_timer_ack() function. The + pthread and dahdi timing modules handle the ack() function being + called when there is no event by simply ignoring it. In the case + of the timerfd module, it results in a read() on the timer fd + that will block forever, as there is no data to read. This caused + Asterisk to lock up very quickly. Thanks to dvossel and + mmichelson for the fun debugging session. :-) ........ + ................ + +2009-03-02 23:15 +0000 [r179473] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 151464 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r151464 | + mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 + lines Make the sip_standard_port function more granular by + allowing separate type and port arguments. This is necessary + because when building our From and Contact headers, we need to be + absolutely sure that we are placing our source port there and not + the peer's source port. (closes issue #12761) Reported by: + asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by + asbestoshead (license 455) ........ + +2009-03-02 23:11 +0000 [r179470] Tilghman Lesher <tlesher@digium.com> + + * /, main/app.c: Merged revisions 179469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) + | 17 lines Merged revisions 179468 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) + | 10 lines When ending a recording with silence detection, + remember to reduce the duration. The end of the recording is + correspondingly trimmed, but the duration was not trimmed by the + number of seconds trimmed, so the saved duration was necessarily + longer than the actual soundfile duration. (closes issue #14406) + Reported by: sasargen Patches: 20090226__bug14406.diff.txt + uploaded by tilghman (license 14) Tested by: sasargen ........ + ................ + +2009-03-02 23:02 +0000 [r179463] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 179462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) + | 16 lines Merged revisions 179461 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) + | 8 lines Ensure that only one thread is calling ast_settimeout() + on a channel at a time. For example, with an IAX2 channel, you + can have both the channel thread and the chan_iax2 processing + threads calling this function, and doing so twice at the same + time is a bad thing. (Found in a debugging session with dvossel + and mmichelson) ........ ................ + +2009-03-02 20:17 +0000 [r179402] Jason Parker <jparker@digium.com> + + * /, main/editline/configure, main/editline/np/unvis.c, + main/editline/sys.h, main/editline/configure.in: Merged revisions + 179396 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | + 9 lines Merged revisions 179395 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | + 1 line Remove several silly warnings in editline. One about a + broken preprocessor directive, and another about strlcpy/strlcat. + (closes issue #14264) Reported by: dimas ........ + ................ + +2009-03-02 17:58 +0000 [r179360-179363] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c: KeepAlive application no longer exists, so fix + gosub implementation to not use it. (closes issue #14571) + Reported by: zktech Patches: 20090302__bug14571.diff.txt uploaded + by tilghman (license 14) Tested by: tilghman + + * cdr/cdr_sqlite3_custom.c: If cdr registration somehow succeeds + without a config file, don't crash. (closes issue #14563) + Reported by: alerios + +2009-03-01 22:07 +0000 [r179220-179222] Mark Michelson <mmichelson@digium.com> + + * apps/app_queue.c: Add error checking when updating the "paused" + field of a realtime queue member. This code already existed in + trunk and 1.6.1, but was not in 1.6.0 prior to this commit. + (closes issue #14338) Reported by: fiddur Patches: 14338.patch + uploaded by mmichelson (license 60) Tested by: fiddur + + * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | + mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 + lines Properly free memory and remove scheduler entries when a + transmission failure occurs. Previously, only the "data" field of + the sip_pkt created during __sip_reliable_xmit was freed when + XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was + called, this inevitably resulted in the reading and writing of + freed memory. XMIT_FAILURE 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 21:33 +0000 [r179162] Tilghman Lesher <tlesher@digium.com> + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) + | 3 lines If config file is blank, don't load module. (Closes + issue #14563) ........ + +2009-02-27 19:05 +0000 [r179058] Jason Parker <jparker@digium.com> + + * /, doc/tex/channelvariables.tex: Merged revisions 179057 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb + 2009) | 8 lines Update documentation for DIALEDTIME and + ANSWEREDTIME variables. (closes issue #14566) Reported by: + klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by + klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by + klaus3000 (license 65) ........ + +2009-02-27 03:52 +0000 [r178987] Steve Murphy <murf@digium.com> + + * configs/features.conf.sample, /, main/features.c: Merged + revisions 178986 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | + 26 lines Merged revisions 178956 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 In this + case, it's just a matter of reducing the default timeouts from + 2000 to 1000 msec, as the max def feature digit timeout is no + longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 + -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default + feature digit timeout to 1000 ms from the previous default of + 500. As per bug 14515, a dev discussion arrived at a "mediated + concensus" of a default feature digit timeout of 1.0 sec. Some + voted for 1300; ctooley thought 1500 for distracted phone users + in phone booths; kpfleming put his foot down at 1.0 sec. Users + who found the previous default max delay of 250 msec perfect, are + welcome to override the new default. Notice that I said that 250 + msec was the default; wait a minute, you might say, the config + file said it was 500 msec!; well, because of the bug fix for + 14515, we found that 500 msec was actually enforcing a max of + 250. The bug fix would restore 500 msec, but we felt even that + was a bit tight for most users... 2000 msec was pushed earlier by + mmichelson, so that reduces to 1000 msec after the bug fix. + Enjoy! ........ ................ + +2009-02-26 17:50 +0000 [r178874] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 178871 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) + | 6 lines IAX2 prune realtime, minor tweak to last fix A return + statement was missing which caused unexpected cli output. issue + #14479 ........ + +2009-02-26 17:29 +0000 [r178866] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 178828 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | + 34 lines Merged revisions 178804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | + 28 lines This patch prevents the feature detection timeout from + being cut in half. Because the ast_channel_bridge() call will + return 0 and pass a frame pointer for both DTMF_BEGIN and + DTMF_END, the feature_timer field in hte config struct is getting + decremented twice, which effectively cuts the digittimeout in + half. I added conditions to the if statement to only let DTMF_END + frames to flow thru, which solved the problem. Also, when the + frame pointer is null, let control flow thru-- this usually + happens on timeouts. I added a comment to the code to explain + what's going on and why. Many thanks to sodom for reporting this + problem. Personnally, it always seemed like something was wrong + with the featuredigittimeout, but I never could quite decide + what... and was too busy to investigate. This bug forced the + issue, and now we know. Sodom had other issues in 14515, but I + couldn't reproduce them. If he still has problems, and wants to + get them solved, he is welcome to reopen 14515. (closes issue + #14515) Reported by: sodom Patches: 14515.patch uploaded by murf + (license 17) Tested by: murf, sodom ........ ................ + +2009-02-26 16:01 +0000 [r178768] David Vossel <dvossel@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 178767 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) + | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. + If "iax2 prune realtime all" was called, it would appear like the + command was successful, but in reality nothing happened. This is + because the reload that was supposed to take place checks the + config files, sees no changes, and does nothing. If there had + been a change in the the config file, the realtime users would + have been marked for deletion and everything would have been + fine. Now prune_users() and prune_peers() are called instead of + reload_config() to prune all users/peers that are realtime. These + functions remove all users/peers with the rtfriend and delme + flags set. iax2_prune_realtime() also lacked the code to properly + delete a single friend. For example. if iax2 prune realtime + <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/ ........ + +2009-02-25 12:46 +0000 [r178510] Russell Bryant <russell@digium.com> + + * main/asterisk.c, /: Merged revisions 178509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) + | 10 lines Merged revisions 178508 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) + | 2 lines Update the copyright year for the main page of the + doxygen documentation. ........ ................ + +2009-02-24 23:28 +0000 [r178382-178447] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 178446 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600 + (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) + | 5 lines Add section about the #exec command in configuration + files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, + with additional notes by tilghman (license 14) ........ + ................ + + * main/asterisk.c, /: Merged revisions 178381 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 | + tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines + Apparently, a void cast doesn't override warn_unused_result. + ........ + +2009-02-24 20:43 +0000 [r178378] Russell Bryant <russell@digium.com> + + * main/rtp.c, /: Merged revisions 178374 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) + | 14 lines Merged revisions 178373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) + | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset + to 0 properly. (issue #14460) Reported by: moliveras Tested by: + russell ........ ................ + +2009-02-24 20:40 +0000 [r178343-178376] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, /: Merged revisions 178375 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 | + tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines + The 3 possible errors with pipe(2) are all impossible in this + situation. ........ + + * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 + Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of + depending upon the astcanary process being inherited by init. + ........ + +2009-02-24 18:05 +0000 [r178306] Terry Wilson <twilson@digium.com> + + * apps/app_dahdiras.c: Change include order to make compile on + Centos 5 with DAHDI If BIT_TYPES_DEFINED gets defined before + linux/types.h is included, the __s32 type doesn't get defined + +2009-02-24 17:53 +0000 [r178304] Tilghman Lesher <tlesher@digium.com> + + * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 | + tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines + Cause astcanary to exit if Asterisk exits abnormally and doesn't + kill astcanary. Also, add some documentation supporting the use + of astcanary. (closes issue #14538) Reported by: KNK Patches: + asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545) + ........ + +2009-02-24 15:20 +0000 [r178224] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | + 16 lines Merged revisions 178205 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 + lines Skip check for extension when subscribing for MWI. Since + the remote side is not actually subscribing to a specific + extension when subscribing for MWI just skip the check to see if + the extension exists. They can't use it to specify the mailbox + either since we require configuration of that in sip.conf (closes + issue #14531) Reported by: festr ........ ................ + +2009-02-23 23:17 +0000 [r178145] Russell Bryant <russell@digium.com> + + * main/rtp.c, /: Merged revisions 178142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) + | 22 lines Merged revisions 178141 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) + | 14 lines Fix infinite DTMF when a BEGIN is received without an + END. This commit is related to rev 175124 of 1.4 where a previous + attempt was made to fix this problem. The problem with the + previous patch was that the inserted code needed to go _before_ + setting the lastrxts to the current timestamp. Because those were + the same, the dtmfcount variable was never decremented, and so + the END was never sent. In passing, I removed the dtmfsamples + variable which was completed unused. I also removed a redundant + setting of the lastrxts variable. (closes issue #14460) Reported + by: moliveras ........ ................ + +2009-02-23 Leif Madsen <lmadsen@digium.com> + + * Released 1.6.0.6 + +2009-02-13 Leif Madsen <lmadsen@digium.com> + + * Released 1.6.0.6-rc1 + +2009-02-13 16:43 +0000 [r175550] Joshua Colp <jcolp@digium.com> + + * /, apps/app_record.c: Merged revisions 175549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | + file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add + an option to keep the recorded file upon hangup. (closes issue + #14341) Reported by: fnordian ........ + +2009-02-12 21:41 +0000 [r175369] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | + russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines + Remove useless string copy, and make sscanf safe again ........ + +2009-02-12 21:27 +0000 [r175347] Tilghman Lesher <tlesher@digium.com> + + * main/udptl.c, /: Merged revisions 175334 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) + | 16 lines Merged revisions 175311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) + | 9 lines Fix crashes when receiving certain T.38 packets. Also, + increase the maximum size of T.38 packets and warn users when + they try to set the limits above those maximums. (closes issue + #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt + uploaded by Corydon76 (license 14) Tested by: schern ........ + ................ + +2009-02-12 20:59 +0000 [r175299-175301] Jeff Peeler <jpeeler@digium.com> + + * main/features.c: Fix mistake in merging conflict from 175299. + + * /, main/features.c: Merged revisions 175298 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) + | 15 lines Merged revisions 175294 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) + | 9 lines Fix ParkedCall event information for From field in the + case of a blind transfer If the parker information can not be + obtained from the peer, try and see if the BLINDTRANSFER channel + variable has been set. Previously, a blind transfer to the + ParkAndAnnounce app would return nothing for the From. Closes + AST-189 ........ ................ + +2009-02-12 20:46 +0000 [r175256-175296] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | + russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines + Avoid using ast_strdupa() in a loop. ........ + + * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) + | 4 lines Don't enable something by default that has a dependency + on something _not_ enabled by default. menuselect was not happy + with this. ........ + +2009-02-12 18:00 +0000 [r175189] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c: Merged revisions 175188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) + | 12 lines Merged revisions 175187 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) + | 6 lines Fix crash in event of failed attempt to transfer to + parking The peer may not necessarily exist, such as in the case + of a transfer to ParkAndAnnounce. In this case don't try to play + a sound to it. ........ ................ + +2009-02-12 17:03 +0000 [r175126] Russell Bryant <russell@digium.com> + + * main/rtp.c, /: Merged revisions 175125 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) + | 35 lines Merged revisions 175124 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) + | 27 lines Don't send DTMF for infinite time if we do not receive + an END event. I thought that this was going to end up being a + pretty gnarly fix, but it turns out that there was actually + already a configuration option in rtp.conf, dtmftimeout, that was + intended to handle this situation. However, in between Asterisk + 1.2 and Asterisk 1.4, the code that processed the option got + lost. So, this commit brings it back to life. The default timeout + is 3 seconds. However, it is worth noting that having this be + configurable at all is not really the recommended behavior in RFC + 2833. From Section 3.5 of RFC 2833: Limiting the time period of + extending the tone is necessary to avoid that a tone "gets + stuck". Regardless of the algorithm used, the tone SHOULD NOT be + extended by more than three packet interarrival times. A slight + extension of tone durations and shortening of pauses is generally + harmless. Three seconds will pretty much _always_ be far more + than three packet interarrival times. However, that behavior is + not required, so I'm going to leave it with our legacy behavior + for now. Code from svn/asterisk/team/russell/issue_14460 (closes + issue #14460) Reported by: moliveras ........ ................ + +2009-02-12 16:33 +0000 [r175122] Mark Michelson <mmichelson@digium.com> + + * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions + 175121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | + mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 + lines 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 14:27 +0000 [r175059-175090] Philippe Sultan <philippe.sultan@gmail.com> + + * /, channels/chan_gtalk.c: Merged revisions 175089 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009) + | 6 lines Issue a warning message if our candidate's IP is the + loopback address. (closes issue #13985) Reported by: jcovert + Tested by: phsultan ........ + + * /, channels/chan_gtalk.c: Merged revisions 175058 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r175058 | phsultan | 2009-02-12 11:31:36 +0100 + (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) + | 12 lines Set the initiator attribute to lowercase in our + replies when receiving calls. This attribute contains a JID that + identifies the initiator of the GoogleTalk voice session. The + GoogleTalk client discards Asterisk's replies if the initiator + attribute contains uppercase characters. (closes issue #13984) + Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by + jcovert (license 551) Tested by: jcovert ........ + ................ + +2009-02-11 23:04 +0000 [r174765-174949] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 174948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb + 2009) | 35 lines 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: Merged revisions 174945 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb + 2009) | 29 lines 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 ........ + + * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 | + mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11 + lines 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, /: Merged revisions 174764 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 | + mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21 + lines 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 20:16 +0000 [r174711] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 | + file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines + Only decrease inringing count if above zero. (issue #13238) + Reported by: kowalma ........ + +2009-02-10 18:19 +0000 [r174596] Matthew Nicholson <mnicholson@digium.com> + + * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb + 2009) | 25 lines 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 15:39 +0000 [r174544] Joshua Colp <jcolp@digium.com> + + * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 | + file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines + 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 05:06 +0000 [r174439] Steve Murphy <murf@digium.com> + + * apps/app_rpt.c: For some strange reason, I didn't think 1.6.0 + needed this fix. I was wrong. Here it is. + +2009-02-09 17:28 +0000 [r174322-174328] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 | + mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 + lines Fix something I messed up in the merge I just did ........ + + * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb + 2009) | 20 lines 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 14:50 +0000 [r174220] Joshua Colp <jcolp@digium.com> + + * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon, + 09 Feb 2009) | 11 lines 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 16:17 +0000 [r174151] Russell Bryant <russell@digium.com> + + * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009) + | 10 lines 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 23:59 +0000 [r174085] Dwayne M. Hubbard <dhubbard@digium.com> + + * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) + | 13 lines 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 19:29 +0000 [r173986-174042] Joshua Colp <jcolp@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 + lines 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 173974 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | + 15 lines 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 16:33 +0000 [r173963] Matthew Nicholson <mnicholson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb + 2009) | 14 lines 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 ........ + ................ + +2009-02-05 23:51 +0000 [r173774-173777] Mark Michelson <mmichelson@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 173776 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, + 05 Feb 2009) | 14 lines 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, /: Merged revisions 173773 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb + 2009) | 7 lines 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 21:04 +0000 [r173698] Jeff Peeler <jpeeler@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600 + (Thu, 05 Feb 2009) | 18 lines 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 20:34 +0000 [r173590-173694] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 173693 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb + 2009) | 20 lines 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 + ........ ................ + + * apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600 + (Thu, 05 Feb 2009) | 11 lines 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 173589 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600 + (Thu, 05 Feb 2009) | 33 lines 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 ........ + ................ + +2009-02-05 16:23 +0000 [r173554] Jeff Peeler <jpeeler@digium.com> + + * build_tools/menuselect-deps.in: fix WORKING_FORK detection + +2009-02-05 00:11 +0000 [r173548] Tilghman Lesher <tlesher@digium.com> + + * build_tools/menuselect-deps.in: regenerate with bootstrap.sh + +2009-02-04 23:44 +0000 [r173546-173547] Jeff Peeler <jpeeler@digium.com> + + * /: I messed up and accidentally reverted the trunk-merged prop + before committing 173546. Added it manually. + + * main/features.c: Merged revisions 173500 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) + | 23 lines 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 22:23 +0000 [r173534] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 173507 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 | + mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 + lines 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 18:55 +0000 [r173460] Tilghman Lesher <tlesher@digium.com> + + * main/tcptls.c, /: Merged revisions 173458 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 | + tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines + 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 17:46 +0000 [r173355-173398] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb + 2009) | 11 lines 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 173393 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb + 2009) | 11 lines 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: Merged revisions 173354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 | + mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 + lines 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 00:45 +0000 [r173312] Tilghman Lesher <tlesher@digium.com> + + * main/pbx.c, /: Merged revisions 173311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 | + tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 + lines 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 23:41 +0000 [r173250] David Vossel <dvossel@digium.com> + + * channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing + of calls. 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. (issue #13468) Review: + http://reviewboard.digium.com/r/140/ + +2009-02-03 00:26 +0000 [r173111] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 173104 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600 + (Mon, 02 Feb 2009) | 12 lines 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 23:59 +0000 [r173068] Terry Wilson <twilson@digium.com> + + * /, main/features.c: Merged revisions 173067 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) + | 9 lines 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 18:15 +0000 [r172896] Leif Madsen <lmadsen@digium.com> + + * /, configs/res_ldap.conf.sample: Merged revisions 172894 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 + Feb 2009) | 7 lines 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 ........ + +2009-02-01 02:45 +0000 [r172707-172742] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) + | 4 lines Blank argument crashes Asterisk (closes issue #14377) + Reported by: amorsen ........ + + * /, funcs/func_strings.c: Merged revisions 172706 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009) + | 7 lines 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) ........ + +2009-01-31 00:06 +0000 [r172635-172637] Terry Wilson <twilson@digium.com> + + * configs/features.conf.sample, /: Merged revisions 172581 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30 + Jan 2009) | 2 lines Remove incorret line from sample config + ........ + + * configs/features.conf.sample, apps/app_dial.c, + main/global_datastores.c, /, main/features.c, + include/asterisk/global_datastores.h, CHANGES: Merged revisions + 172580 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) + | 44 lines 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 22:23 +0000 [r172604] Mark Michelson <mmichelson@digium.com> + + * /, include/asterisk/channel.h: Merged revisions 172598 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, + 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h + ........ + +2009-01-29 23:47 +0000 [r172503] Tilghman Lesher <tlesher@digium.com> + + * main/asterisk.c, apps/app_nbscat.c, /, autoconf/ast_func_fork.m4, + apps/app_festival.c, build_tools/menuselect-deps.in, configure, + apps/app_dahdiras.c, apps/app_mp3.c, res/res_agi.c, + apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c: + Merged revisions 172441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) + | 16 lines 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 21:35 +0000 [r172434] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged + revisions 172400 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 | + rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 + lines 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. + ........ + +2009-01-29 16:49 +0000 [r172316] Tilghman Lesher <tlesher@digium.com> + + * configs/func_odbc.conf.sample, /: Merged revisions 172315 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29 + Jan 2009) | 2 lines Better document mode=multirow, based upon a + conversation with Jared. ........ + +2009-01-29 13:51 +0000 [r172273] Leif Madsen <lmadsen@digium.com> + + * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 172271 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29 + Jan 2009) | 5 lines 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 09:56 +0000 [r172217] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 172173 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 + lines 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 20:41 +0000 [r172065] Steve Murphy <murf@digium.com> + + * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /, + main/features.c, include/asterisk/channel.h: Merged revisions + 172063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | + 52 lines 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 17:28 +0000 [r171965] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600 + (Wed, 28 Jan 2009) | 9 lines 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 13:18 +0000 [r171846] Olle Johansson <oej@edvina.net> + + * /, configs/sip.conf.sample: Merged revisions 171838 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, + 28 Jan 2009) | 10 lines 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. ........ + ................ + +2009-01-27 22:00 +0000 [r171619-171692] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 171691 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600 + (Tue, 27 Jan 2009) | 47 lines 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 171622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan + 2009) | 26 lines 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: Merged revisions 171618 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 | + mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 + lines 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 16:15 +0000 [r171594-171595] Matthew Fredrickson <creslin@digium.com> + + * main/ast_expr2.c, main/ast_expr2.h: Revert some changes that + shouldn't have made it in + + * main/ast_expr2.c, channels/chan_dahdi.c, main/ast_expr2.h: Make + sure we do not go into alarm on PTMP links with non persistent + D-channels + +2009-01-27 15:13 +0000 [r171529] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23 + lines 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 + ........ ................ + +2009-01-26 14:02 +0000 [r171327] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) | + 17 lines 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 ........ + ................ + +2009-01-26 00:03 +0000 [r171189] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009) + | 13 lines 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 13:38 +0000 [r170981] Sean Bright <sean.bright@gmail.com> + + * /, apps/app_page.c: Merged revisions 170980 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan + 2009) | 16 lines 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 02:50 +0000 [r170944] Russell Bryant <russell@digium.com> + + * include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) + | 6 lines 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. ........ + +2009-01-24 13:56 +0000 [r170838] Tilghman Lesher <tlesher@digium.com> + + * configs/res_odbc.conf.sample, /: Merged revisions 170837 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600 + (Sat, 24 Jan 2009) | 9 lines 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 23:52 +0000 [r170830] Richard Mudgett <rmudgett@digium.com> + + * /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 | + rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line + Fix asterisk.pdf generation if branch name has an underscore in + it. ........ + +2009-01-23 22:59 +0000 [r170791] Russell Bryant <russell@digium.com> + + * /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 | + russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines + Don't blow up if a branch name has an underscore in it ........ + +2009-01-23 20:56 +0000 [r170685-170721] Mark Michelson <mmichelson@digium.com> + + * configs/res_odbc.conf.sample, /: Merged revisions 170720 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600 + (Fri, 23 Jan 2009) | 16 lines 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 170677 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600 + (Fri, 23 Jan 2009) | 22 lines 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 20:19 +0000 [r170659] Joshua Colp <jcolp@digium.com> + + * main/channel.c, /: Merged revisions 170652 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | + 11 lines 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 Tilghman Lesher <tlesher@digium.com> + + * Released 1.6.0.5 + + * channels/chan_iax2.c: Regression fixes for security fix AST-2009-001 + +2009-01-06 Tilghman Lesher <tlesher@digium.com> + + * Released 1.6.0.3 + + * channels/chan_iax2.c: Security fix AST-2009-001 + +2008-12-03 Tilghman Lesher <tlesher@digium.com> + + * Released 1.6.0.3-rc1 + +2008-12-03 14:13 +0000 [r160482] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 160481 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) + | 14 lines Merged revisions 160480 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) + | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I + guess that having only ip-phones in mind is not a good approach. + Since it is possible to have a sip proxy connected to asterisk we + could receive a 407 (unauthorized) or 483 (too many hops) as + response and dialog ending would not be a good behavior." So + modified. ........ ................ + +2008-12-03 00:53 +0000 [r160427] Sean Bright <sean.bright@gmail.com> + + * Makefile: Fix some 'make menuselect' breakage introduced by + recent merges. + +2008-12-02 23:22 +0000 [r160386-160393] Tilghman Lesher <tlesher@digium.com> + + * apps/app_dial.c, /: Merged revisions 156388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) + | 12 lines Merged revisions 156386 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) + | 5 lines When using call limits under 1 second, infinite call + lengths are allowed, instead. (closes issue #13851) Reported by: + ruddy ........ ................ + + * /, apps/app_meetme.c: Merged revisions 156290 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008) + | 11 lines Merged revisions 156289 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) + | 3 lines For whatever reason, gcc only warned me about the + possible use of an uninitialized variable when compiling 1.6.1. + ........ ................ + + * /, apps/app_meetme.c: Merged revisions 156228 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008) + | 16 lines Merged revisions 156178 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) + | 8 lines (closes issue #13173) Reported by: pep This change adds + an announce_thread responsible for playing announcements to an + existing conference. This allows all announcing to be immediately + stopped if necessary but more importantly allows other threads + that need to play something to not block. There are multiple + benefits to this, but the actual bug is for solving the scenario + for a channel to be unusable after hang up for the entire + duration of the parting announcement. The parting announcement + can be extremely long depending on what the user recorded upon + joining the conference. Reviewed by Russell on Review Board: + http://reviewboard.digium.com/r/25/ ........ ................ + + * main/astobj2.c, main/asterisk.c, apps/app_while.c, + apps/app_dial.c, main/pbx.c, channels/chan_misdn.c, + main/manager.c, /, apps/app_meetme.c, channels/chan_sip.c, + channels/chan_skinny.c, include/asterisk/astobj2.h, + channels/chan_agent.c, channels/chan_h323.c, + channels/chan_iax2.c: Merged revisions + 152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r152969 | tilghman | 2008-10-30 15:35:46 -0500 + (Thu, 30 Oct 2008) | 10 lines Merged revisions 152958 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) + | 3 lines Cannot join detached threads. See + http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html + (Closes issue #13400) ........ ................ r153122 | + tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10 + lines Merged revisions 153114 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) + | 3 lines Turn off qualify on uncached realtime peers. (Closes + issue #13383) ........ ................ r154264 | tilghman | + 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines Recorded + merge of revisions 154263 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) + | 3 lines Make the monitor thread non-detached, so it can be + joined (suggested by Russell on -dev list). ........ + ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600 + (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) + | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state + when it receives the indication AST_CONTROL_RINGING. ........ + ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600 + (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) + | 9 lines On busy systems, it's possible for the values checked + within a single line of code to change, unless the structure is + locked to ensure a consistent state. (closes issue #13717) + Reported by: kowalma Patches: 20081102__bug13717.diff.txt + uploaded by Corydon76 (license 14) Tested by: kowalma ........ + ................ r155399 | tilghman | 2008-11-07 16:28:58 -0600 + (Fri, 07 Nov 2008) | 14 lines Merged revisions 155398 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) + | 7 lines Clarify error message. (closes issue #13809) Reported + by: denke Patches: 20081104__bug13809.diff.txt uploaded by + Corydon76 (license 14) Tested by: denke ........ ................ + r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov + 2008) | 22 lines Merged revisions 155861 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov + 2008) | 14 lines Channel drivers assume that when their indicate + callback is invoked, that the channel on which the callback was + called is locked. This patch corrects an instance in chan_agent + where a channel's indicate callback is called directly without + first locking the channel. This was leading to some observed + locking issues in chan_local, but considering that all channel + drivers operate under the same expectations, the generic fix in + chan_agent is the right way to go. AST-126 ........ + ................ r156166 | russell | 2008-11-12 11:38:20 -0600 + (Wed, 12 Nov 2008) | 15 lines Merged revisions 156164 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) + | 7 lines Move the sanity check that makes sure "always fork" is + not set along with the console option to be after the code that + reads options from asterisk.conf. This resolves a situation where + Asterisk can start taking up 100% when misconfigured. (Thanks to + Bryce Porter (x86 on IRC) for letting me log in to his system to + figure out what was causing the 100% CPU problem.) ........ + ................ r156295 | tilghman | 2008-11-12 13:28:22 -0600 + (Wed, 12 Nov 2008) | 13 lines Merged revisions 156294 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) + | 6 lines If the SLA thread is not started, then reload causes a + memory leak. (closes issue #13889) Reported by: eliel Patches: + app_meetme.c.patch uploaded by eliel (license 64) ........ + ................ r156690 | tilghman | 2008-11-13 15:30:41 -0600 + (Thu, 13 Nov 2008) | 14 lines Merged revisions 156688 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) + | 7 lines Provide more space for all the data which can appear in + an originating channel name. (closes issue #13398) Reported by: + bamby Patches: manager.c.diff uploaded by bamby (license 430) + ........ ................ r156756 | tilghman | 2008-11-13 + 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines Merged revisions + 156755 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) + | 6 lines ast_waitfordigit() requires that the channel be up, for + no good logical reason. This prevents While/EndWhile from working + within the "h" extension. Reported by: jgalarneau (for ABE C.2) + Fixed by: me ........ ................ r158066 | mmichelson | + 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged + revisions 158053 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov + 2008) | 12 lines Make sure to set the hangup cause on the calling + channel in the case that ast_call() fails. For incoming SIP + channels, this was causing us to send a 603 instead of a 486 when + the call-limit was reached on the destination channel. (closes + issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded + by putnopvut (license 60) Tested by: blitzrage ........ + ................ r158082 | mmichelson | 2008-11-20 11:54:31 -0600 + (Thu, 20 Nov 2008) | 24 lines Merged revisions 158071 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov + 2008) | 16 lines We don't handle 4XX responses to BYE well. + According to section 15 of RFC 3261, we should terminate a dialog + if we receive a 481 or 408 in response to our BYE. Since I am + aware of at least one phone manufacturer who may sometimes send a + 404 as well, I am being liberal and saying that any 4XX response + to a BYE should result in a terminated dialog. (closes issue + #12994) Reported by: pabelanger Patches: 12994.patch uploaded by + putnopvut (license 60) Closes AST-129 ........ ................ + r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) + | 10 lines Merged revisions 158539 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) + | 2 lines When compiling with DEBUG_THREADS, report the real + file/func/line for ao2_lock/ao2_unlock ........ ................ + r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) + | 12 lines Merged revisions 158600 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) + | 5 lines The passed extension may not be the same in the list as + the current entry, because we strip spaces when copying the + extension into the structure. Therefore, use the copied item to + place the item into the list. (found by lmadsen on -dev, fixed by + me) ........ ................ r159276 | tilghman | 2008-11-25 + 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions + 159269 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) + | 7 lines Don't try to send a response on a NULL pvt. (closes + issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch + uploaded by eliel (license 64) Tested by: barthpbx ........ + ................ + + * configs/features.conf.sample, apps/app_voicemail.c, + apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c, /, + channels/chan_sip.c, apps/app_queue.c: Merged revisions + 152216,152287,152369,152467,152569,152605 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r152216 | tilghman | 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008) + | 13 lines Merged revisions 152215 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008) + | 6 lines Inherit ALL elements of CallerID across a local + channel. (closes issue #13368) Reported by: Peter Schlaile + Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 + (license 14) ........ ................ r152287 | jpeeler | + 2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) | 10 lines Merged + revisions 152286 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008) + | 2 lines Buffer policy setting for half is not needed. ........ + ................ r152369 | tilghman | 2008-10-28 12:07:39 -0500 + (Tue, 28 Oct 2008) | 15 lines Merged revisions 152368 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) + | 8 lines Reset all DIAL variables back to blank, in case Dial is + called multiple times per call (which could otherwise lead to + inconsistent status reports). (closes issue #13216) Reported by: + ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 + (license 14) Tested by: ruddy ........ ................ r152467 | + tilghman | 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) | 10 + lines Merged revisions 152463 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008) + | 3 lines Quoting in the wrong direction (Fixes AST-107) ........ + ................ r152569 | russell | 2008-10-29 00:34:26 -0500 + (Wed, 29 Oct 2008) | 15 lines Merged revisions 152539 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) + | 7 lines Fix an incorrect usage of sizeof() (closes issue + #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch + uploaded by andrew53 (license 519) ........ ................ + r152605 | murf | 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) | + 22 lines Merged revisions 152538 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | + 14 lines A little documentation cross-ref between features and + dial and queue... I wasted some time (stupidly) trying to get the + one-touch parking stuff working, because it didn't occur to me + that I had to also have the corresponding options in the dial + command! Duh! (In all this time, I never set this up before!) So, + to keep some poor fool from suffering the same fate, I made the + features.conf.sample file mention the corresponding opts in + dial/queue; and the docs for dial/app specifically mention the + corresponding decls in the feature.conf file. I hope this doesn't + spoil some vast, eternal plan... ........ ................ + + * apps/app_speech_utils.c, apps/app_voicemail.c, Makefile, + channels/chan_dahdi.c, /, channels/chan_sip.c, + include/asterisk/audiohook.h, apps/app_waitforsilence.c, + main/features.c, main/audiohook.c, apps/app_queue.c: Merged + revisions + 147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r147518 | file | 2008-10-08 09:53:51 -0500 (Wed, + 08 Oct 2008) | 9 lines Merged revisions 147517 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 + lines If we receive DTMF make sure that the state of the speech + structure goes back to being not ready. (issue #LUMENVOX-8) + ........ ................ r147689 | kpfleming | 2008-10-08 + 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions + 147681 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct + 2008) | 3 lines when parsing a text configuration option, ensure + that the buffer on the stack is actually large enough to hold the + legal values of that option, and also ensure that sscanf() knows + to stop parsing if it would overrun the buffer (without these + changes, specifying "buffers=...,immediate" would overflow the + buffer on the stack, and could not have worked as expected) + ........ ................ r148000 | tilghman | 2008-10-09 + 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines Merged revisions + 147997 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) + | 4 lines When blank, callerid name and number should display + "unknown caller" in voicemail emails. (Closes issue #13643) + ........ ................ r148112 | mmichelson | 2008-10-09 + 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines Merged revisions + 146026 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | + 18 lines (closes issue #13579) Reported by: dwagner (closes issue + #13584) Reported by: dwagner Tested by: murf, putnopvut The + thought occurred to me that the res= from the extension spawn was + ending up being returned from the bridge. "Thou shalt not poison + the return value". Made the change and it appears to allow blind + xfers to work as normal. If I'm wrong, reopen the bugs. But it + looks good to me! Many thanks to putnopvut for helping me + reproduce this! ........ ................ r148268 | tilghman | + 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines Merged + revisions 148257 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) + | 7 lines User not notified of temporary greeting, if ODBC + storage is in use. (closes issue #13659) Reported by: moliveras + Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 + (license 14) Tested by: moliveras ........ ................ + r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) + | 11 lines Merged revisions 148916 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) + | 4 lines Ensure that mail headers are 7-bit clean, even when + UTF-8 characters are used in headers like 'Subject' and 'To'. + Closes AST-107. ........ ................ r148988 | tilghman | + 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines Merged + revisions 148987 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) + | 2 lines Some compilers warn, some don't. Fixing. ........ + ................ r149062 | tilghman | 2008-10-14 15:16:48 -0500 + (Tue, 14 Oct 2008) | 13 lines Merged revisions 149061 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) + | 6 lines Check correct values in the return of ast_waitfor(); + also, get rid of a possible memory leak. (closes issue #13658) + Reported by: explidous Patch by: me ........ ................ + r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct + 2008) | 15 lines Merged revisions 149130 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct + 2008) | 7 lines Don't allow reserved characters to be used in + register lines in sip.conf. (closes issue #13570) Reported by: + putnopvut ........ ................ r149201 | mmichelson | + 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines Merged + revisions 149200 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct + 2008) | 12 lines Update the queue with the correct number of + calls and whether the call was completed within the service level + when a transfer takes place. This way, we do not "break" the + leastrecent and fewestcalls strategies by not logging a call + until after the transferred call has ended. (closes issue #13395) + Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded + by Marquis (license 32) ........ ................ r149205 | + mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 + lines Merged revisions 149204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct + 2008) | 12 lines Add a tolerance period for sync-triggered + audiohooks so that if packetization of audio is close (but not + equal) we don't end up flushing the audiohooks over small + inconsistencies in synchronization. Related to issue #13005, and + solves the issue for most people who were experiencing the + problem. However, a small number of people are still experiencing + the problem on long calls, so I am not closing the issue yet + ........ ................ r149208 | mmichelson | 2008-10-14 + 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines Merged revisions + 149207 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct + 2008) | 9 lines Call register_peer_exten even in the case that + the peer's IP/port does not change. (closes issue #13309) + Reported by: dimas Patches: v2-13309.patch uploaded by dimas + (license 88) ........ ................ + + * channels/misdn/isdn_lib.c, Makefile, channels/chan_dahdi.c, + channels/chan_misdn.c, main/manager.c, /: Merged revisions + 115313,121770,123272,139624,140205,144257 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r115313 | tilghman | 2008-05-05 15:22:08 -0500 (Mon, 05 May 2008) + | 10 lines Merged revisions 115312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r115312 | tilghman | 2008-05-05 15:17:55 -0500 (Mon, 05 May 2008) + | 2 lines Reverse order, such that user configs override default + selections ........ ................ r121770 | crichter | + 2008-06-11 06:52:18 -0500 (Wed, 11 Jun 2008) | 9 lines Merged + revisions 121751 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008) + | 1 line fixed issue with previous commit, the find_free_channel + test for channels which where inuse was broken. ........ + ................ r123272 | russell | 2008-06-17 10:52:13 -0500 + (Tue, 17 Jun 2008) | 12 lines Merged revisions 123271 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) + | 4 lines Fix a memory leak in astobj2 that was pointed out by + seanbright. When a container got destroyed, the underlying bucket + list entry for each object that was in the container at that time + did not get free'd. ........ ................ r139624 | jpeeler | + 2008-08-22 16:57:32 -0500 (Fri, 22 Aug 2008) | 13 lines Merged + revisions 139621 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008) + | 5 lines (closes issue #13359) Reported by: Laureano Patches: + originate_channel_check.patch uploaded by Laureano (license 265) + ........ ................ r140205 | jpeeler | 2008-08-26 13:48:55 + -0500 (Tue, 26 Aug 2008) | 17 lines Merged revisions 140056 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26 Aug 2008) + | 9 lines (closes issue #12071) Reported by: tzafrir Patches: + dahdi_close.diff uploaded by tzafrir (license 46) Tested by: + tzafrir, jpeeler This patch fixes closing open file descriptors + in the case of an error. ........ ................ r144257 | + crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines + Merged revisions 144238 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24 Sep 2008) + | 1 line improved helptext of misdn_set_opt. ........ + ................ + +2008-12-02 18:05 +0000 [r160326-160337] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 160333 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008) + | 1 line remove duplicate comment that I accidentally merged + ........ + + * channels/chan_dahdi.c, /: Merged revisions 160319 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008) + | 7 lines (closes issue #13786) Reported by: tzafrir Readding + DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which + fixes not being able to make outgoing calls on some FXO adapters: + http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553 + ........ + +2008-12-02 18:01 +0000 [r160228-160322] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 160308 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) + | 17 lines Merged revisions 160297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) + | 10 lines When the text does not match exactly (e.g. RTP/SAVP), + then the %n conversion fails, and the resulting integer is + garbage. Thus, we must initialize the integer and check it + afterwards for success. (closes issue #14000) Reported by: folke + Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke + (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by + folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff + uploaded by folke (license 626) ........ ................ + + * include/asterisk/stringfields.h, apps/app_voicemail.c, + main/cli.c, main/pbx.c, main/frame.c, /, + channels/chan_features.c: Merged revisions 160208 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600 + (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) + | 3 lines Ensure that Asterisk builds with --enable-dev-mode, + even on the latest gcc and glibc. ........ ................ + +2008-12-01 23:41 +0000 [r160173] Sean Bright <sean.bright@gmail.com> + + * channels/chan_phone.c, main/manager.c, /, utils/smsq.c: Merged + revisions 160170-160172 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec + 2008) | 1 line Pay attention to the return value of system(), + even if we basically ignore it. ................ r160171 | + seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1 + line Silence a build warning. (chan_phone.c:810: warning: value + computed is not used) ................ r160172 | seanbright | + 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines Merged + revisions 159976 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) + | 3 lines Get rid of the useless format string and argument in + the Bogus/ manager channelname. Noted by kpfleming and name + Bogus/manager suggested by eliel ........ ................ + +2008-12-01 Tilghman Lesher <tlesher@digium.com> + + * Released 1.6.0.2 + +2008-12-01 21:45 +0000 [r160100] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 160097 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008) + | 2 lines Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or + bad things happen. ........ + +2008-12-01 21:07 +0000 [r160096] Sean Bright <sean.bright@gmail.com> + + * include/asterisk.h, /: Merged revisions 154919 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 | + seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2 + lines Fix a problem found while building res_snmp. ........ + +2008-12-01 17:39 +0000 [r160005] Russell Bryant <russell@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 160004 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r160004 | russell | 2008-12-01 11:34:31 -0600 + (Mon, 01 Dec 2008) | 14 lines Merged revisions 160003 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) + | 6 lines Apply some logic used in iax2_indicate() to + iax2_setoption(), as well, since they both have the potential to + send control frames in the middle of call setup. We have to wait + until we have received a message back from the remote end before + we try to send any more frames. Otherwise, the remote end will + consider it invalid, and we'll get stuck in an INVAL/VNAK storm. + ........ ................ + +2008-12-01 16:04 +0000 [r159974] Michiel van Baak <michiel@vanbaak.info> + + * main/manager.c, /: Merged revisions 159898 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008) + | 11 lines Merged revisions 159897 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008) + | 4 lines make manager compile on OpenBSD. The last (10th) + argument to ast_channel_alloc here should be a pointer and NULL + is not really a pointer. ........ ................ + +2008-12-01 14:56 +0000 [r159915] Russell Bryant <russell@digium.com> + + * .cleancount, /: Merged revisions 159911 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008) + | 10 lines Merged revisions 159900 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008) + | 2 lines Force a "make clean" to avoid a bizarre build issue ... + ........ ................ + +2008-11-29 18:37 +0000 [r159855] Kevin P. Fleming <kpfleming@digium.com> + + * utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c, Makefile, + include/asterisk/logger.h, include/asterisk/res_odbc.h, + main/srv.c, channels/chan_misdn.c, + include/asterisk/linkedlists.h, main/event.c, + include/asterisk/strings.h, utils/extconf.c, makeopts.in, + include/asterisk/stringfields.h, utils/check_expr.c, + channels/chan_vpb.cc, /, main/utils.c, res/res_config_sqlite.c, + utils/frame.c, channels/misdn_config.c, include/asterisk/astmm.h, + include/asterisk/compat.h, configure, channels/misdn/ie.c, + include/asterisk/module.h, main/features.c, main/dns.c, + funcs/Makefile, include/asterisk/devicestate.h, + include/asterisk/utils.h, channels/chan_sip.c, main/Makefile, + include/asterisk/dundi.h, include/asterisk/enum.h, configure.ac, + channels/chan_agent.c, utils/astman.c, include/asterisk/cli.h, + include/asterisk/channel.h, include/jitterbuf.h, + include/asterisk/manager.h: Merged revisions 159818 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov + 2008) | 18 lines incorporates r159808 from branches/1.4: + ------------------------------------------------------------------------ + r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov + 2008) | 7 lines update dev-mode compiler flags to match the ones + used by default on Ubuntu Intrepid, so all developers will see + the same warnings and errors since this branch already had some + printf format attributes, enable checking for them and tag + functions that didn't have them format attributes in a consistent + way + ------------------------------------------------------------------------ + in addition: move some format attributes from main/utils.c to the + header files they belong in, and fix up references to the + relevant functions based on new compiler warnings ........ + +2008-11-26 19:58 +0000 [r159558] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, /: Merged revisions 159554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 | + mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 + lines Add some necessary hangup commands in the case that + forwarding a call fails 1) Hang up the original destination if + the local channel cannot be requested. 2) Hang up the local + channel (in addition to the original destination) if ast_call + fails when calling the newly created local channel. This prevents + channels from sticking around forever in the case of a botched + call forward (e.g. to an extension which does not exist). (closes + issue #13764) Reported by: davidw Patches: 13764_v2.patch + uploaded by putnopvut (license 60) Tested by: putnopvut, davidw + ........ + +2008-11-26 19:18 +0000 [r159536] Kevin P. Fleming <kpfleming@digium.com> + + * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules, + Makefile.rules: Merged revisions 159534 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov + 2008) | 11 lines Merged revisions 159476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov + 2008) | 7 lines simplify (and slightly bug-fix) the recent + developer-oriented COMPILE_DOUBLE mode ensure that 'make clean' + removes dependency files for .i files that are created in + COMPILE_DOUBLE mode ........ ................ + +2008-11-26 18:40 +0000 [r159478] Tilghman Lesher <tlesher@digium.com> + + * main/udptl.c, /: Merged revisions 159475 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r159475 | + tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines + If the config file does not exist, then the first use crashes + Asterisk. (closes issue #13848) Reported by: klaus3000 Patches: + udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage + ........ + +2008-11-26 15:01 +0000 [r159439] Mark Michelson <mmichelson@digium.com> + + * channels/chan_agent.c: Merged revisions 159437 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r159437 | + mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov 2008) | 10 + lines Don't allow for configuration options to overwrite options + set via channel variables on a reload. (closes issue #13921) + Reported by: davidw Patches: 13921.patch uploaded by putnopvut + (license 60) Tested by: davidw ........ + +2008-11-25 23:09 +0000 [r159374] Steve Murphy <murf@digium.com> + + * main/cdr.c, /, channels/chan_iax2.c: Merged revisions 159360 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue, + 25 Nov 2008) | 23 lines Merged revisions 159316 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | + 15 lines (closes issue #12694) Reported by: yraber Patches: + 12694.2nd.diff uploaded by murf (license 17) Tested by: murf, + laurav Thanks to file (Joshua Colp) for his IAX fix. the change + to cdr.c allows no-answer to percolate up into CDR's, and feels + like the right place to locate this fix; if BUSY is done here, + no-answer should be, too. ........ ................ + +2008-11-25 22:28 +0000 [r159314] Mark Michelson <mmichelson@digium.com> + + * main/channel.c: I don't care what anyone says, this change is + going into 1.6.0. Otherwise, the simple act of logging an agent + in spams the CLI with warning messages about failed reads of the + alertpipe. + +2008-11-25 21:43 +0000 [r159248] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 159247 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r159247 | tilghman | 2008-11-25 15:42:42 -0600 + (Tue, 25 Nov 2008) | 21 lines Merged revisions 159246 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600 + (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) + | 7 lines Regression fix for last security fix. Set the iseqno + correctly. (closes issue #13918) Reported by: ffloimair Patches: + 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) + Tested by: ffloimair ........ ................ ................ + +2008-11-25 16:21 +0000 [r159024-159094] Terry Wilson <twilson@digium.com> + + * /, apps/app_festival.c: Merged revisions 159093 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 | + twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines + Add missing variable declaration for PPC code ........ + + * channels/chan_usbradio.c, /: Merged revisions 158992 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008) + | 2 lines Make chan_usbradio compile under dev mode ........ + +2008-11-21 22:40 +0000 [r158545] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 158484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) | + 19 lines Merged revisions 158483 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | + 11 lines (closes issue #13871) Reported by: mdu113 This one is + totally my fault. The code doesn't even create a bridge CDR if + the channel CDR has POST_DISABLED. I didn't check for that at the + end of the bridge. Fixed with a few small insertions. Tested. + Looks good. No cdr generated, no crash, no unnecc. data objects + created either. ........ ................ + +2008-11-21 22:13 +0000 [r158542] Russell Bryant <russell@digium.com> + + * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions + 158540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) + | 10 lines Merged revisions 158539 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) + | 2 lines When compiling with DEBUG_THREADS, report the real + file/func/line for ao2_lock/ao2_unlock ........ ................ + +2008-11-21 20:44 +0000 [r158451] Kevin P. Fleming <kpfleming@digium.com> + + * /, UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, + UPGRADE-1.6.txt, CHANGES: Merged revisions 158449 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov + 2008) | 3 lines as suggested by jtodd, document the purposes of + the CHANGES and UPGRADE files ........ + +2008-11-21 17:12 +0000 [r158376] Terry Wilson <twilson@digium.com> + + * cdr/cdr_csv.c, /: Merged revisions 158374 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158374 | + twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines + Reloading the config and having no changes still initialized some + settings to 0. Initialize settings after doing all of the cfg + checks. (closes issue #13942) Reported by: davidw Patches: + cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by: + davidw ........ + +2008-11-21 01:23 +0000 [r158231-158267] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 158265-158266 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, + 20 Nov 2008) | 4 lines Use some magic constants to get the right + size for this sscanf statement. Thanks Richard! ........ r158266 + | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 + lines Use a more expressive constant for a 64-bit scanned int + ........ + + * /, channels/chan_sip.c: Merged revisions 158262 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 | + mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 + lines Fix the build for 32-bit systems. %lu is only 32-bits on + 32-bit systems, so we need to use %llu instead. Of course %llu is + 128-bits on 64-bit systems, so we have to cast to unsigned long + long. No harm, but it's sure annoying. ........ + + * /, channels/chan_sip.c: Merged revisions 158230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 | + mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 + lines Change the remote user agent session version variable from + an int to a uint64_t. This prevents potential comparison problems + from happening if the version string exceeds INT_MAX. This was an + apparent problem for one user who could not properly place a call + on hold since the version in the SDP of the re-INVITE to place + the call on hold greatly exceeded INT_MAX. This also aligns with + RFC 2327 better since it recommends using an NTP timestamp for + the version (which is a 64-bit number). (closes issue #13531) + Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut + (license 60) Tested by: sgofferj ........ + +2008-11-20 19:42 +0000 [r158190] Sean Bright <sean.bright@gmail.com> + + * res/ael/pval.c, /: Merged revisions 158188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r158188 | + seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10 + lines Fix one case where the application argument was not + converted from a pipe to a comma. This was causing problems with + switch statements with empty expressions. (closes issue #13901) + Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by + seanbright (license 71) Tested by: seanbright Reviewed by: murf + ........ + +2008-11-20 00:12 +0000 [r157738-157976] Kevin P. Fleming <kpfleming@digium.com> + + * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree, + channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, res/ael, + channels, main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash, + codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules, + channels/misdn, main/db1-ast/mpool, Makefile.rules, res/snmp, + pbx/Makefile, res/Makefile: Merged revisions 157974 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r157974 | kpfleming | 2008-11-19 18:08:12 -0600 + (Wed, 19 Nov 2008) | 13 lines Merged revisions 157859 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov + 2008) | 7 lines the gcc optimizer frequently finds broken code + (use of uninitalized variables, unreachable code, etc.), which is + good. however, developers usually compile with the optimizer + turned off, because if they need to debug the resulting code, + optimized code makes that process very difficult. this means that + we get code changes committed that weren't adequately checked + over for these sorts of problems. with this build system change, + if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is + turned on, when a source file is compiled it will actually be + preprocessed (into a .i or .ii file), then compiled once with + optimization (with the result sent to /dev/null) and again + without optimization (but only if the first compile succeeded, of + course). while making these changes, i did some cleanup work in + Makefile.rules to move commonly-used combinations of flag + variables into their own variables, to make the file easier to + read and maintain ........ ................ + + * /, res/res_agi.c: Merged revisions 157743 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157743 | + kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line + correct small bug introduced during API conversion ........ + + * apps/app_stack.c, include/asterisk/agi.h, /, channels/chan_sip.c, + res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added): Merged + revisions 157706 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | + kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 + lines make some corrections to the ast_agi_register_multiple(), + ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to + be consistent with API guidelines also, move UPGRADE.txt to + UPGRADE-1.6.txt and make the new UPGRADE.txt contain information + about upgrading between Asterisk 1.6 releases ........ + +2008-11-19 00:33 +0000 [r157601] Sean Bright <sean.bright@gmail.com> + + * Makefile, /, build_tools/make_version, configure, configure.ac, + build_tools/make_buildopts_h, makeopts.in: Merged revisions + 157600 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157600 | + seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10 + lines Fix a few build problems on Solaris (and check for an md5 + utility in configure instead of the icky loop I was doing + before). (closes issue #13842) Reported by: snuffy Patches: + bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff + uploaded by seanbright (license 71) Tested by: snuffy ........ + +2008-11-18 22:59 +0000 [r157307-157541] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Merged revisions 157512 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov + 2008) | 21 lines Merged revisions 157503 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov + 2008) | 13 lines Add some missing invite state changes necessary + in the sip_write function. Not setting the invite state correctly + on the call was resulting in the Record application leaving empty + files. I also have updated the doxygen comment next to the + declaration of the INV_EARLY_MEDIA constant to reflect that we + also use this state when we *send* a 18X response to an INVITE. + (closes issue #13878) Reported by: nahuelgreco Patches: + sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco + (license 162) Tested by: putnopvut ........ ................ + + * channels/chan_sip.c: Once again, Russell to the rescue. Use the + builtin astobj1 lock of the sip_peer and sip_user instead of + adding a new one + + * /, channels/chan_sip.c: Merged revisions 157496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 | + mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 + lines Based on Russell's advice on the asterisk-dev list, I have + changed from using a global lock in update_call_counter to using + the locks within the sip_pvt and sip_peer structures instead. + ........ + + * /, channels/chan_sip.c: Merged revisions 157427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 | + mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 + lines * Add a lock to be used in the update_call_counter + function. * Revert logic to mirror 1.4's in the sense that it + will not allow the call counter to dip below 0. These two + measures prevent potential races that could cause a SIP peer to + appear to be busy forever. (closes issue #13668) Reported by: mjc + Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic + (license 586) ........ + + * apps/app_dial.c, channels/chan_local.c, /, main/features.c, + include/asterisk/channel.h, apps/app_followme.c: Merged revisions + 157306 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov + 2008) | 20 lines Merged revisions 157305 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov + 2008) | 12 lines Fix a crash in the end_bridge_callback of + app_dial and app_followme which would occur at the end of an + attended transfer. The error occurred because we initially stored + a pointer to an ast_channel which then was hung up due to a + masquerade. This commit adds a "fixup" callback to the + bridge_config structure to allow for end_bridge_callback_data to + be changed in the case that a new channel pointer is needed for + the end_bridge_callback. ........ ................ + +2008-11-18 18:10 +0000 [r157303] Steve Murphy <murf@digium.com> + + * main/config.c, /: Merged revisions 157302 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r157302 | + murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines + (closes issue #13420) Reported by: alex70 Patches: + 13420.13539.patch uploaded by murf (license 17) Tested by: murf, + awk This fixes two problems: a spurious linefeed insertion + probably left over from pre-precomment times. Only generated when + category had no previous comments. The other problem: Insertions + could get the line-numbering out of whack and generate negative + line numbers, causing chunks of line numbers to be emitted, on + the scale of the number of lines up to that point in the file. In + such cases, abort the looping, and all is well. ........ + +2008-11-15 19:47 +0000 [r157107-157165] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged + revisions 157164 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov + 2008) | 13 lines Merged revisions 157162-157163 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov + 2008) | 1 line dist-clean should remove dependency information + files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03 + +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory + dist-clean is run, run clean in that directory first, and when + running top-level dist-clean, do not run subdirectory clean + operations twice ........ ................ + + * /, contrib/asterisk-ng-doxygen: Merged revisions 157105 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 + Nov 2008) | 13 lines major update to doxygen configuration file: + 1) update to doxygen 1.5.x style file, as used in trunk 2) tell + doxygen where are header files are, so include-file processing + can be done 3) make all macros that are used to define + variables/functions be expanded, so that doxygen will properly + document the resulting variable/function 4) make all macros that + are used to provide the contents of a variable (structure) be + expanded, so that doxygen will be able to document the resulting + fields 5) suppress compiler attributes (__attribute__(xxx)) from + being seen by doxygen, so it will properly match up function + definition and usage (for an example of th effect of this, look + at the doxygen docs for ast_log() from before and afte this + commit) ........ + +2008-11-14 17:03 +0000 [r156912] Tilghman Lesher <tlesher@digium.com> + + * main/manager.c, /: Merged revisions 156911 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r156911 | + tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines + Ping is missing the standard double-newline after the event. + (closes issue #13903) Reported by: kebl0155 ........ + +2008-11-14 16:55 +0000 [r156818-156889] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/strings.h, apps/app_queue.c: This is the 1.6.0 + version of revision 156883 of trunk. This is different in that it + preserves the case-sensitiveness of processing queues from + configuration. closes issue #13703 + + * apps/app_voicemail.c, /: Merged revisions 156817 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r156817 | mmichelson | 2008-11-14 09:20:03 -0600 + (Fri, 14 Nov 2008) | 18 lines Merged revisions 156816 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov + 2008) | 10 lines If the prompt to reenter a voicemail password + timed out, it resulted in the password not being saved, even if + the input matched what you gave when first prompted to enter a + new password. This is because the return value of ast_readstring + was checked, but not checked properly. This bug was discovered by + Jared Smith during an Asterisk training course. Thanks for + reporting it! ........ ................ + +2008-11-13 19:26 +0000 [r156652-156653] Brandon Kruse <bkruse@digium.com> + + * main/manager.c: Update to Coding Guidelines + + * main/manager.c, /: Merged revisions 156017 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r156017 | + pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines + Patch by Ryan Brindley -- Make sure that manager refuses any + duplicate 'new category' requests in updateconfig (closes issue + #13539) ........ + +2008-11-12 19:56 +0000 [r156319] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 156299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) | + 26 lines Merged revisions 156297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | + 18 lines It turns out that the 0x0XX00 codes being returned for + N, X, and Z are off by one, as per conversation with jsmith on + #asterisk-dev; he was teaching a class and disconcerted that this + published rule was not being followed, with patterns _NXX, + _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should + have been. This change, tested on these 3 patterns now picks the + proper one. However, this change may surprise users who set up + dialplans based on previous behavior, which has been there for + what, 2 and half years or so now. ........ ................ + +2008-11-12 18:57 +0000 [r156251] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 156243 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r156243 | tilghman | 2008-11-12 12:55:18 -0600 + (Wed, 12 Nov 2008) | 18 lines Merged revisions 156229 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) + | 11 lines Revert revision 132506, since it occasionally caused + IAX2 HANGUP packets not to be sent, and instead, schedule a task + to destroy the iax2 pvt structure 10 seconds later. This allows + the IAX2 HANGUP packet to be queued, transmitted, and ACKed + before the pvt is destroyed. (closes issue #13645) Reported by: + dzajro Patches: 20081111__bug13645__3.diff.txt uploaded by + Corydon76 (license 14) Tested by: vazir Reviewed: + http://reviewboard.digium.com/r/51/ ........ ................ + +2008-11-12 17:47 +0000 [r156170] Mark Michelson <mmichelson@digium.com> + + * apps/app_dial.c, /: Merged revisions 156169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov + 2008) | 15 lines Merged revisions 156167 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov + 2008) | 7 lines When doing some tests, I was having a crash at + the end of every call if an attended transfer occurred during the + call. I traced the cause to the CDR on one of the channels being + NULL. murf suggested a check in the end bridge callback to be + sure the CDR is non-NULL before proceeding, so that's what I'm + adding. ........ ................ + +2008-11-11 21:28 +0000 [r156012] Russell Bryant <russell@digium.com> + + * apps/app_directory.c: Don't blow up if we get NULL when trying to + parse out the full name field (fixed for Jared in the training + room) + +2008-11-11 20:04 +0000 [r156007] Michiel van Baak <michiel@vanbaak.info> + + * /: remove prop that shouldn't be here + +2008-11-11 19:49 +0000 [r155815-156004] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_realtime.c: Merged revisions 155862 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155862 | + tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines + Make documentation of update method match documentation and + update update2 method to match. Reported by: atis, via -dev + mailing list. Fixed by: me ........ + + * doc/valgrind.txt, /: Merged revisions 155804 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155803 | + tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line + I got tired of saying this in every single bugnote referring to + this file. ........ + +2008-11-09 01:34 +0000 [r155555] Sean Bright <sean.bright@gmail.com> + + * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h, + apps/app_followme.c, apps/app_queue.c: Merged revisions 155554 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500 + (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov + 2008) | 6 lines Use static functions here instead of nested ones. + This requires a small change to the ast_bridge_config struct as + well. To understand the reason for this change, see the following + post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html + ........ ................ + +2008-11-07 23:42 +0000 [r155361-155468] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 155467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 | + mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 + lines Set the invite state to INV_CANCELLED in a place that makes + more sense. Where it was set before, it was impossible to + actually delay sending a CANCEL if we had not yet received a + provisional response to an INVITE. (closes issue #13626) Reported + by: atis Patches: 13626.patch uploaded by putnopvut (license 60) + Tested by: atis ........ + + * /, configs/voicemail.conf.sample: Merged revisions 155360 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri, + 07 Nov 2008) | 8 lines Remove one more instance of the sample + configuration lying about what's possible. The tz cannot be set + in a context like this. It can only be set in the general section + or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing + this out ........ + +2008-11-06 22:50 +0000 [r155123] Kevin P. Fleming <kpfleming@digium.com> + + * /, res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: Merged + revisions 155121 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r155121 | + kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3 + lines don't blindly assume that Darwin and Cygwin need + GLOB_ABORTED defined; only define it if it is not already defined + ........ + +2008-11-06 19:47 +0000 [r155013] Mark Michelson <mmichelson@digium.com> + + * /, configs/voicemail.conf.sample: Merged revisions 155012 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600 + (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov + 2008) | 8 lines The documentation listed the ability to set + 'maxmsg' per context. The truth is that you can only set this in + the general section or per mailbox. Thus I am updating the sample + config file to be more accurate. Thanks to sasargen on IRC for + bringing up this issue. ........ ................ + +2008-11-03 22:30 +0000 [r154062-154081] Tilghman Lesher <tlesher@digium.com> + + * /: Recorded merge of revisions 154072 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r154072 | tilghman | 2008-11-03 16:28:12 -0600 (Mon, 03 Nov 2008) + | 12 lines Merged revisions 154066 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008) + | 5 lines Attempting to expunge a mailbox when the mailstream is + NULL will crash Asterisk. (Closes issue #13829) Reported by: + jaroth Patch by: me (modified jaroth's patch) ........ + ................ + + * main/rtp.c, /: Merged revisions 154060 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) + | 3 lines Remove the potential for a division by zero error. + (Closes issue #13810) ........ + +2008-11-03 00:53 +0000 [r153743-153746] Kevin P. Fleming <kpfleming@digium.com> + + * /: record revisions that were manually merged + + * apps/app_stack.c, include/asterisk/agi.h, configure, + include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4, + configure.ac, include/asterisk/compiler.h: Merge revision 153709 + from trunk + ------------------------------------------------------------------------ + r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov + 2008) | 3 lines instead of trying to forcibly load res_agi when + app_stack is loaded (even if the administrator didn't want it + loaded), use GCC weak symbols to determine whether it was loaded + already or not; if it was loaded, then use it. + ------------------------------------------------------------------------ + + * channels/chan_oss.c, agi/eagi-sphinx-test.c, res/ael/ael_lex.c, + channels/chan_h323.c, main/file.c, apps/app_sms.c, + pbx/pbx_dundi.c, res/ael/ael.flex, pbx/pbx_config.c, + apps/app_chanspy.c, apps/app_stack.c, utils/streamplayer.c, + main/asterisk.c, apps/app_voicemail.c, utils/muted.c, + apps/app_authenticate.c, res/res_phoneprov.c, main/utils.c, + res/res_musiconhold.c, formats/format_wav_gsm.c, + res/res_jabber.c, channels/chan_iax2.c, utils/frame.c, + utils/stereorize.c, main/channel.c, channels/chan_dahdi.c, + main/manager.c, res/ael/ael.tab.c, funcs/func_odbc.c, + main/ast_expr2f.c, res/res_agi.c, main/logger.c, main/http.c, + formats/format_gsm.c, apps/app_adsiprog.c, apps/app_dial.c, + channels/chan_sip.c, formats/format_wav.c, apps/app_festival.c, + main/db1-ast/hash/hash_page.c, res/ael/ael.y, res/res_crypto.c, + agi/eagi-test.c, utils/astman.c, pbx/pbx_lua.c, + formats/format_ogg_vorbis.c, utils/astcanary.c, apps/app_queue.c: + port gcc 4.3.x warning fixes from trunk to this branch + +2008-10-31 21:49 +0000 [r153265] Terry Wilson <twilson@digium.com> + + * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h, + apps/app_followme.c, apps/app_queue.c: Merged revisions 153181 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 + Oct 2008) | 5 lines Recent CDR fixes moved execution of the 'h' + exten into the bridging code, so variables that were set after + ast_bridge_call was called would not show up in the 'h' exten. + Added a callback function to handle setting variables, etc. from + w/in the bridging code. Calls back into a nested function within + the function calling ast_bridge_call (closes issue #13793) + Reported by: greenfieldtech ........ + +2008-10-30 21:00 +0000 [r152994] Sean Bright <sean.bright@gmail.com> + + * /, bootstrap.sh: Merged revisions 152993 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r152993 | seanbright | 2008-10-30 16:59:17 -0400 (Thu, 30 Oct + 2008) | 10 lines Merged revisions 152992 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct + 2008) | 2 lines The -I argument to aclocal needs a space before + the include directory name. ........ ................ + +2008-10-30 16:54 +0000 [r152813] Kevin P. Fleming <kpfleming@digium.com> + + * main/cdr.c, /: Merged revisions 152812 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r152812 | kpfleming | 2008-10-30 11:54:29 -0500 (Thu, 30 Oct + 2008) | 9 lines Merged revisions 152811 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct + 2008) | 3 lines instead of comparing the string pointer to 0, + let's compare the value that was actually parsed out of the + string (found by sparse) ........ ................ + +2008-10-30 04:28 +0000 [r152772] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 152765 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r152765 | tilghman | 2008-10-29 23:26:34 -0500 (Wed, 29 + Oct 2008) | 5 lines Set up an example stdexten that preserves the + original context and extension in the CDR. (Related to issue + #13799) Reported by: davidw ........ + +2008-10-29 20:54 +0000 [r152647] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_directory.c: Merged revisions 152646 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r152646 | mmichelson | 2008-10-29 15:53:53 -0500 (Wed, 29 Oct + 2008) | 9 lines If there was no named defined in a voicemail.conf + mailbox entry, then app_directory would crash when attempting to + read that entry from the file. We now check for the NULL or empty + string properly so that there will be no crash. (closes issue + #13804) Reported by: bluecrow76 ........ + +2008-10-29 20:13 +0000 [r152644] Terry Wilson <twilson@digium.com> + + * apps/app_queue.c: Small modification to putnopvut's patch to fix + this issue. Thanks for all the help, putnopvut! (closes issue + #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch + uploaded by otherwiseguy (license 396) Tested by: otherwiseguy + +2008-10-28 21:39 +0000 [r152443] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_mgcp.c, /: Merged revisions 152442 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r152442 | tilghman | 2008-10-28 16:38:26 -0500 (Tue, 28 Oct 2008) + | 7 lines Only re-add the io port if it was closed, otherwise + reload causes a memory leak. (closes issue #13785) Reported by: + eliel Patches: chan_mgcp.c.patch uploaded by eliel (license 64) + ........ + +2008-10-27 16:33 +0000 [r152157] Tilghman Lesher <tlesher@digium.com> + + * apps/app_stack.c, /: Merged revisions 152134 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r152134 | + tilghman | 2008-10-27 11:24:11 -0500 (Mon, 27 Oct 2008) | 4 lines + Oops, only delete the ARG variables once upon release. The + following section would have removed them again (removing + variables from 2 stack frames, instead of just one). ........ + +2008-10-26 20:26 +0000 [r152062] Sean Bright <sean.bright@gmail.com> + + * /, funcs/func_strings.c: Merged revisions 152060 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r152060 | seanbright | 2008-10-26 16:25:08 -0400 + (Sun, 26 Oct 2008) | 15 lines Merged revisions 152059 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct + 2008) | 7 lines Since passing \0 as the second argument to strchr + is valid (and will match the trailing \0 of a string) we need to + check that first, otherwise we end up with incorrect results. Fix + suggested by reporter. (closes issue #13787) Reported by: + meitinger ........ ................ + +2008-10-23 16:12 +0000 [r151765] Russell Bryant <russell@digium.com> + + * channels/chan_sip.c: Fix some memory leaks. These issues are + 1.6.0 specific. - Freeing the peer got accidentally removed from + the peer's destructor. It is still needed for astobj, but not for + astobj2. - Fix some places that called find_user or find_peer, + but did not release the reference that was returned. (closes + issue #13331) Reported by: sergee Patches: + chan_sip-3leaks-16-r151244.diff uploaded by sergee (license 138) + Tested by: sergee + +2008-10-20 05:03 +0000 [r151244] Kevin P. Fleming <kpfleming@digium.com> + + * autoconf (added), autoconf/ast_check_pwlib.m4, + autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure, + autoconf/ast_gcc_attribute.m4, bootstrap.sh, + autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4, + autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4, + autoconf/ast_c_define_check.m4, autoconf/ast_prog_egrep.m4, + autoconf/ast_ext_tool_check.m4, autoconf/ast_check_mandatory.m4, + /, autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4, + configure.ac, acinclude.m4 (removed), autoconf/ast_prog_sed.m4: + Merged revisions 151242-151243 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r151242 | kpfleming | 2008-10-20 07:59:04 +0300 (Mon, 20 Oct + 2008) | 9 lines Merged revisions 151240 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct + 2008) | 3 lines break up acinclude.m4 into individual files, + which will make it easier to maintain, easier to add new macros + (less patching) and will ease maintenance of these macros across + Asterisk branches ........ ................ r151243 | kpfleming | + 2008-10-20 08:00:56 +0300 (Mon, 20 Oct 2008) | 9 lines Merged + revisions 151241 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct + 2008) | 2 lines rename this macro to properly reflect what it + does ........ ................ + +2008-10-18 02:35 +0000 [r150854] BJ Weschke <bweschke@btwtech.com> + + * main/manager.c, /: Merged revisions 150817 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r150817 | + bweschke | 2008-10-17 22:18:33 -0400 (Fri, 17 Oct 2008) | 8 lines + Using the GetVar handler in AMI is potentially dangerous + (insta-crash [tm]) when you use a dialplan function that requires + a channel and then you don't provide one or provide an invalid + one in the Channel: parameter. We'll handle this situation + exactly the same way it was handled in pbx.c back on r61766. + We'll create a bogus channel for the function call and destroy it + when we're done. If we have trouble allocating the bogus channel + then we're not going to try executing the function call at all + and run the risk of crashing. (closes issue #13715) reported by: + makoto patch by: bweschke ........ + +2008-10-17 00:19 +0000 [r150308-150313] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Instead of merging commit 150307 to 1.6.0, I + had meant to block it in 1.6.1...time to go home :) + + * /, channels/chan_sip.c: Merged revisions 150307 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r150307 | + mmichelson | 2008-10-16 19:13:35 -0500 (Thu, 16 Oct 2008) | 14 + lines After a long discussion on #asterisk-bugs, it seems kind of + odd that a channel would be named after the port on which it came + in on. For endpoints that always include ":5060" as part of the + From: header, it will mean that you have a ton of channels with + names like "SIP/5060-3ea38a8b." I am boldly moving forward with + this change in trunk, but I'm not touching other branches with + this one since this definitely would qualify as a behavior + change. If there is a problem with this commit, and I haven't + seen the obvious reason why you'd want to name the channel after + the port from which the call originated, then please feel free to + revert this ........ + +2008-10-16 16:10 +0000 [r150126] Richard Mudgett <rmudgett@digium.com> + + * channels/chan_misdn.c, /: Merged revisions 150125 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r150125 | rmudgett | 2008-10-16 11:04:45 -0500 + (Thu, 16 Oct 2008) | 9 lines Merged revisions 150124 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16 + Oct 2008) | 1 line Fix memory leak found by customer ........ + ................ + +2008-10-15 20:18 +0000 [r149757] BJ Weschke <bweschke@btwtech.com> + + * configs/agents.conf.sample, /: Merged revisions 149756 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r149756 | bweschke | 2008-10-15 16:14:20 -0400 + (Wed, 15 Oct 2008) | 10 lines Merged revisions 149683 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008) + | 4 lines An update to the documentation/example of + agents.conf.sample with the correct parameter for this feature as + defined in chan_agent.c (closes issue #13709) ........ + ................ + +2008-10-15 11:29 +0000 [r149495] Kevin P. Fleming <kpfleming@digium.com> + + * /, channels/chan_sip.c: Merged revisions 149487 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r149487 | kpfleming | 2008-10-15 13:26:36 +0200 (Wed, 15 Oct + 2008) | 9 lines Merged revisions 149452 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct + 2008) | 3 lines fix some problems when parsing SIP messages that + have the maximum number of headers or body lines that we support + ........ ................ + +2008-10-14 17:39 +0000 [r148914] Mark Michelson <mmichelson@digium.com> + + * channels/chan_local.c, /: Merged revisions 148913 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r148913 | mmichelson | 2008-10-14 12:38:06 -0500 + (Tue, 14 Oct 2008) | 17 lines Merged revisions 148912 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, 14 Oct + 2008) | 9 lines Deadlock prevention in chan_local. (closes issue + #13676) Reported by: tacvbo Patches: 13676.patch uploaded by + putnopvut (license 60) Tested by: tacvbo ........ + ................ + +2008-10-14 10:34 +0000 [r148613-148739] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /: Merged revisions 148738 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r148738 | kpfleming | 2008-10-14 12:33:14 +0200 (Tue, 14 Oct + 2008) | 9 lines Merged revisions 148736 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct + 2008) | 3 lines on Ubuntu (at least), recent versions of ld in + binutils delete all debugging symbols when -x is supplied; since + the reasons why -x is being passed are lost in the mists of time, + remove it so debugging will work properly ........ + ................ + + * /, main/translate.c: Merged revisions 148612 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r148612 | kpfleming | 2008-10-14 03:06:45 -0500 (Tue, 14 Oct + 2008) | 9 lines Merged revisions 148611 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct + 2008) | 3 lines it would be nice if this message printing code + had actually been tested before it was committed... ........ + ................ + +2008-10-10 21:18 +0000 [r148374] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 148373 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r148373 | + mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8 + lines Make sure that the inUse and inRinging fields for a sip + peer cannot go below zero. This is a regression from 1.4 and so + it will be applied to 1.6.0 as well. (closes issue #13668) + Reported by: mjc ........ + +2008-10-10 01:25 +0000 [r148201-148204] Sean Bright <sean.bright@gmail.com> + + * res/res_config_sqlite.c, apps/app_voicemail.c, + include/asterisk.h, /, main/tdd.c, main/cryptostub.c: Merged + revisions 148200 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r148200 | + seanbright | 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) | 12 + lines Don't include logger.h in asterisk.h by default as it is + causing problems building app_voicemail. Instead, include it + where it is needed. This turned out to be a relatively minor + issue because other headers include logger.h as well. Need to + test -addons before merging this back to 1.6.0. (closes issue + #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff + uploaded by seanbright (license 71) Tested by: mmichelson + ........ + + * apps/app_rpt.c: Somehow we got conflict markers checked in! Might + need a 1.6.0.1 sooner than we'd like. + +2008-10-09 23:31 +0000 [r148147] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 148144 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r148144 | mmichelson | 2008-10-09 18:30:47 -0500 (Thu, 09 Oct + 2008) | 10 lines Read the callerid in the correct order and make + sure to read the Urgent flag value from the IMAP headers. (closes + issue #13652) Reported by: jaroth Patches: imapheaders.patch + uploaded by jaroth (license 50) ........ + +2008-10-09 23:26 +0000 [r148124] Tilghman Lesher <tlesher@digium.com> + + * /, configs/res_ldap.conf.sample: Merged revisions 148120 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r148120 | tilghman | 2008-10-09 18:25:53 -0500 (Thu, 09 + Oct 2008) | 6 lines Fix example schema (closes issue #12860) + Reported by: flyn Patches: res_ldap.conf.patch uploaded by flyn + (license 503) ........ + +2008-10-09 17:51 +0000 [r147900] Michiel van Baak <michiel@vanbaak.info> + + * include/asterisk/endian.h, /: Merged revisions 147899 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r147899 | mvanbaak | 2008-10-09 19:48:53 +0200 (Thu, 09 + Oct 2008) | 5 lines only include this for OpenBSD. At least + FreeBSD is borked when including it (closes issue #13649) + Reported by: ys ........ + +2008-10-09 17:47 +0000 [r147897] Tilghman Lesher <tlesher@digium.com> + + * configs/extensions.conf.sample, /: Merged revisions 147896 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r147896 | tilghman | 2008-10-09 12:46:15 -0500 (Thu, 09 + Oct 2008) | 4 lines Remove "second form" of extensions, as it no + longer applies. Also, cleanup the grammar, formatting, and + introduce several clarifications to the text. (Closes issue + #13654) ........ + +2008-10-09 14:56 +0000 [r147809] Steve Murphy <murf@digium.com> + + * main/astobj2.c, channels/chan_oss.c, main/config.c, main/rtp.c, + main/cli.c, configure, channels/console_gui.c, utils/extconf.c, + main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES, /, + include/asterisk/autoconfig.h.in, main/translate.c, + channels/vcodecs.c, configure.ac, channels/console_video.c, + channels/chan_iax2.c: Merged revisions 147807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r147807 | + murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines + (closes issue #13557) Reported by: nickpeirson Patches: + pbx.c.patch uploaded by nickpeirson (license 579) + replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) + Tested by: nickpeirson, murf 1. replaced all refs to bzero and + bcopy to memset and memmove instead. 2. added a note to the + CODING-GUIDELINES 3. add two macros to asterisk.h to prevent + bzero, bcopy from creeping back into the source 4. removed bzero + from configure, configure.ac, autoconfig.h.in ........ + +2008-10-08 12:16 +0000 [r147458] Russell Bryant <russell@digium.com> + + * configs/chan_dahdi.conf.sample: Remove the sample configuration + for configuration sections in chan_dahdi.conf. This code was not + merged into 1.6.0. Reported by: angler (closes AST-119) + +2008-10-08 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0.1 released. + + * configs/chan_dahdi.conf.sample: Remove mention of configuration + sections for defining channels in chan_dahdi.conf. This code + is in 1.6.1, and was not merged into 1.6.0. + +2008-10-01 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0 released. + +2008-09-09 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-rc6 released. + +2008-09-09 15:44 +0000 [r142065] Russell Bryant <russell@digium.com> + + * /, main/features.c: Merged revisions 142064 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008) + | 13 lines Merged revisions 142063 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) + | 5 lines Ensure that the stored CDR reference is still valid + after the bridge before poking at it. Also, keep the channel + locked while messing with this CDR. (fixes crashes reported in + issue #13409) ........ ................ + +2008-09-09 12:34 +0000 [r141996-141999] Mark Michelson <mmichelson@digium.com> + + * channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 | + mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8 + lines Fix a memory leak in chan_oss (closes issue #13311) + Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel + (license 64) ........ + +2008-09-09 01:49 +0000 [r141950] Russell Bryant <russell@digium.com> + + * main/channel.c, /: Merged revisions 141949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 | + russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines + Modify ast_answer() to not hold the channel lock while calling + ast_safe_sleep() or when calling ast_waitfor(). These are + inappropriate times to hold the channel lock. This is what has + caused "could not get the channel lock" messages from chan_sip + and has likely caused a negative impact on performance results of + SIP in Asterisk 1.6. Thanks to file for pointing out this section + of code. (closes issue #13287) (closes issue #13115) ........ + +2008-09-08 21:07 +0000 [r141808] Russell Bryant <russell@digium.com> + + * main/pbx.c, /: Merged revisions 141807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008) + | 15 lines Merged revisions 141806 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) + | 7 lines When doing an async goto, detect if the channel is + already in the middle of a masquerade. This can happen when + chan_local is trying to optimize itself out. If this happens, + fail the async goto instead of bursting into flames. (closes + issue #13435) Reported by: geoff2010 ........ ................ + +2008-09-08 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-rc5 released. + +2008-09-08 20:19 +0000 [r141746] Jason Parker <jparker@digium.com> + + * Makefile, /, redhat (removed): Merged revisions 141745 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r141745 | qwell | 2008-09-08 15:18:17 -0500 + (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) | + 8 lines Remove RPM package targets from Makefile (and all + associated parts). This has never worked in 1.4, and we decided + that it makes no sense to be done here. There are many distros + out there that already have "proper" spec files that can be + (re)used. Closes issue #13113 Closes issue #10950 Closes issue + #10952 ........ ................ + +2008-09-08 17:14 +0000 [r141683] Sean Bright <sean.bright@gmail.com> + + * /, build_tools/make_buildopts_h: Merged revisions 141682 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon, + 08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on + various platforms doesn't choke on the special characters (like + ^). (closes issue #13417) Reported by: dougm Patches: + 13417.make_buildopts_h.patch uploaded by seanbright (license 71) + Tested by: dougm ........ + +2008-09-06 20:21 +0000 [r141567] Steve Murphy <murf@digium.com> + + * /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 + lines Merged revisions 141565 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 + line This fix comes from Joshua Colp The Brilliant, who, given + the trace, came up with a solution. This will most likely will + close 13235 and 13409. I'll wait till Monday to verify, and then + close these bugs. ........ ................ + +2008-09-06 15:40 +0000 [r141505-141508] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 141504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008) + | 12 lines Merged revisions 141503 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008) + | 4 lines Reverting behavior change (AGI should not exit non-zero + on SUCCESS) (closes issue #13434) Reported by: francesco_r + ........ ................ + +2008-09-05 22:06 +0000 [r141368-141426] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_agent.c: Merged revisions 141367 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500 + (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep + 2008) | 7 lines Agent's should not try to call a channel's + indicate callback if the channel has been hung up. It will likely + crash otherwise ABE-1159 ........ ................ + +2008-09-05 14:24 +0000 [r141116-141158] Steve Murphy <murf@digium.com> + + * main/channel.c, /: Merged revisions 141157 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 + lines Merged revisions 141156 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 + line A small change to prevent double-posting of CDR's; thanks to + Daniel Ferrer for bringing it to our attention ........ + ................ + + * pbx/ael/ael-test/ref.ael-vtest25 (added), /, + pbx/ael/ael-test/ael-vtest25/extensions.ael, + pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c, + pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged + revisions 141115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) | + 78 lines Merged revisions 141094 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | + 70 lines (closes issue #13357) Reported by: pj Tested by: murf + (closes issue #13416) Reported by: yarns Tested by: murf If you + find this message overly verbose, relax, it's probably not meant + for you. This message is meant for probably only two people in + the whole world: me, or the poor schnook that has to maintain + this code because I'm either dead or unavailable at the moment. + This fix solves two reports, both having to do with embedding a + function call in a ${} construct. It was tricky because the + funccall syntax has parenthesis () in it. And up till now, the + 'word' token in the flex stuff didn't allow that, because it + would tend to steal the LP and RP tokens. To be truthful, the + "word" token was the trickiest, most unstable thing in the whole + lexer. I was lucky it made this long without complaints. I had to + choose every character in the pattern with extreme care, and I + knew that someday I'd have to revisit it. Well, the day has come. + So, my brilliant idea (and I'm being modest), was to use the + surrounding ${} construct to make a state machine and capture + everything in it, no matter what it contains. But, I have to now + treat the word token like I did with comments, in that I turn the + whole thing into a state-machine sort of spec, with new contexts + "curlystate", "wordstate", and "brackstate". Wait a minute, + "brackstate"? Yes, well, it didn't take very many regression + tests to point out if I do this for ${} constructs, I also have + to do it with the $[] constructs, too. I had to create a separate + pcbstack2 and pcbstack3 because these constructs can occur inside + macro argument lists, and when we have two state machines + operating on the same structures we'd get problems otherwise. I + guess I could have stopped at pcbstack2 and had the brackstate + stuff share it, but it doesn't hurt to be safe. So, the pcbpush + and pcbpop routines also now have versions for "2" and "3". I had + to add the {KEYWORD} construct to the initial pattern for "word", + because previously word would match stuff like "default7", + because it was a longer match than the keyword "default". But, + not any more, because the word pattern only matches only one or + two characters now, and it will always lose. So, I made it the + winner again by making an optional match on any of the keywords + before it's normal pattern. I added another regression test to + make sure we don't lose this in future edits, and had to fix just + one regression, where it no longer reports a 'cascaded' error, + which I guess is a plus. I've given some thought as to whether to + apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I + decided to put it in 1.4 because one of the bug reports was + against 1.4; and it is unexpected that AEL cannot handle this + situation. It actually reduced the amount of useless "cascade" + error messages that appeared in the regressions (by one line, + ehhem). There is a possible side-effect in that it does now do + more careful checking of what's in those ${} constructs, as far + as matching parens, and brackets are concerned. Some users may + find a an insidious problem and correct it this way. This should + be exceedingly rare, I hope. ........ ................ + +2008-09-04 18:35 +0000 [r141086] Jeff Peeler <jpeeler@digium.com> + + * /, main/features.c, res/res_agi.c: Merged revisions 141039 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500 + (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) + | 7 lines (closes issue #11979) Fixes multiple parking problems: + Crash when executing a park on an extension dialed by AGI due to + not returning the proper return code. Crash when using a builtin + feature that was a subset of a enabled dynamic feature. Crash due + to always hanging up the peer despite the fact that the peer was + supposed to be parked. ........ ................ + +2008-09-03 20:18 +0000 [r140976] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 140975 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 | + mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4 + lines Fix some locking order issues in app_queue. This was + brought up by atis on IRC a while ago. ........ + +2008-09-03 Russell Bryant <russell@digium.com> + + * Asterisk 1.6.0-rc4 released. + +2008-09-03 14:17 +0000 [r140825-140827] Steve Murphy <murf@digium.com> + + * main/cdr.c, /: Merged revisions 140749 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) | + 11 lines Merged revisions 140747 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 + line I am turning the warnings generated in ast_cdr_free and + post_cdr into verbose level 2 messages. Really, they matter + little to end users. You either get the CDR's you wanted, or you + don't, and it is a bug. For trunk, I am going one step further. + These messages were pretty worthless even for debug, so I'm + completely removing them. ........ ................ + + * main/channel.c, /: Merged revisions 140692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | + 13 lines Merged revisions 140690 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 + line After reconsidering, with respect to 13409, ast_cdr_detach + should be OK, better in fact, than ast_cdr_free, which generates + lots of useless warnings that will undoubtably generate + complaints. Hmmm. It doesn't hush the useless warnings, but it + does allow control of posting via the detach and post routines, + for those possible situations, where you'd want to post + single-channel cdrs. ........ ................ + + * main/channel.c, main/pbx.c, /: Merged revisions 140691 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue, + 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | + 14 lines (closes issue #13409) Reported by: tomaso Patches: + asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license + 564) I basically spent the day, verifying that this patch solves + the problem, and doesn't hurt in non-problem cases. Why valgrind + did not plainly reveal this leak absolutely mystifies and stuns + me. Many, many thanks to tomaso for finding and providing the + fix. ........ ................ + +2008-09-03 13:27 +0000 [r140818] Russell Bryant <russell@digium.com> + + * main/poll.c, /: Merged revisions 140817 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008) + | 12 lines Merged revisions 140816 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008) + | 4 lines Don't freak out if the poll emulation receives NULL for + the pollfds array (closes issue #13307) Reported by: jcovert + ........ ................ + +2008-09-02 18:17 +0000 [r140607] Sean Bright <sean.bright@gmail.com> + + * /, channels/chan_iax2.c: Merged revisions 140606 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400 + (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep + 2008) | 8 lines Make sure to use the correct length of the + mohinterpret and mohsuggest buffers when copying configuration + values. (closes issue #13336) Reported by: + decryptus_proformatique Patches: + chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded + by decryptus (license 555) ........ ................ + +2008-09-02 15:12 +0000 [r140564-140567] Russell Bryant <russell@digium.com> + + * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions + 140566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 | + russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines + Update instructions for getting libresample ........ + +2008-08-27 20:15 +0000 [r140302-140304] Mark Michelson <mmichelson@digium.com> + + * channels/chan_sip.c: Revert commit 140302. Should not be merging + changes like that into a release-candidate branch + + * channels/chan_sip.c: Merged revisions 140301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug + 2008) | 19 lines Merged revisions 140299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug + 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when + in pedantic mode. The problem was that the wrong tags would be + compared depending on the direction of the call. (closes issue + #13353) Reported by: flefoll Patches: + chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll + (license 244) ........ ................ + +2008-08-26 18:12 +0000 [r140170] Russell Bryant <russell@digium.com> + + * Makefile, /: Merged revisions 140169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 | + russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines + Fix building menuselect-tree with PRINT_DIR set. We _must_ use + the --quiet flag here, or else some arbitrary text will end up in + the resulting menuselect-tree file and things will explode. + ........ + +2008-08-25 21:33 +0000 [r139918] Sean Bright <sean.bright@gmail.com> + + * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged + revisions 139915 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug + 2008) | 17 lines Merged revisions 139909 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug + 2008) | 9 lines Some versions of awk (nawk, for example) don't + like empty regular expressions so be slightly more verbose. + (closes issue #13374) Reported by: dougm Patches: 13374.diff + uploaded by seanbright (license 71) Tested by: dougm ........ + ................ + +2008-08-25 21:05 +0000 [r139872] Terry Wilson <twilson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008) + | 10 lines Merged revisions 139869 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008) + | 2 lines Make SIPADDHEADER() propagate indefinitely ........ + ................ + +2008-08-25 16:00 +0000 [r139774] Steve Murphy <murf@digium.com> + + * main/pbx.c, /, main/features.c: Merged revisions 139770 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon, + 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 + lines This patch reverts the changes made via 139347, and 139635, + as users are seeing adverse difference. I will un-close 13251. + Back to the drawing board/ concept/ beginning/ whatever! ........ + ................ + +2008-08-24 16:30 +0000 [r139705-139708] Tilghman Lesher <tlesher@digium.com> + + * /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 | + tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines + Memory leak ........ + +2008-08-22 22:35 +0000 [r139628-139671] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 139662 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) | + 14 lines Merged revisions 139635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 + lines I found some problems with the code I committed earlier, + when I merged them into trunk, so I'm coming back to clean up. + And, in the process, I found an error in the code I added to + trunk and 1.6.x, that I'll fix using this patch also. ........ + ................ + + * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions + 139627 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | + 59 lines Merged revisions 139347 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | + 47 lines (closes issue #13251) Reported by: sergee Tested by: + murf THis is a bold move for a static release fix, but I wouldn't + have made it if I didn't feel confident (at least a *bit* + confident) that it wouldn't mess everyone up. The reasoning goes + something like this: 1. We simply cannot do anything with CDR's + at the current point (in pbx.c, after the __ast_pbx_run loop). + It's way too late to have any affect on the CDRs. The CDR is + already posted and gone, and the remnants have been cleared. 2. I + was very much afraid that moving the running of the 'h' extension + down into the bridge code (where it would be now practical to do + it), would result in a lot more calls to the 'h' exten, so I + implemented it as another exten under another name, but found, to + my pleasant surprise, that there was a 1:1 correspondence to the + running of the 'h' exten in the pbx_run loop, and the new spot at + the end of the bridge. So, I ifdef'd out the current 'h' loop, + and moved it into the bridge code. The only difference I can see + is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this + is still an important decision point, I can replicate it if there + are complaints. To be perfectly honest, the KEEPALIVE situation + is not totally clear to me, and how it relates to a post-bridge + situation is less clear. I suspect the users will point out + everything in total clarity if this steps on anyone's toes! 3. I + temporarily swap the bridge_cdr into the channel before running + the 'h' exten, which makes it possible for users to edit the cdr + before it goes out the door. And, of course, with the + endbeforehexten config var set, the users can also get at the + billsec/duration vals. After the h exten finishes, the cdr is + swapped back and processing continues as normal. Please, all who + deal with CDR's, please test this version of Asterisk, and file + bug reports as appropriate! ........ I also made a little fix to + the app_dial's 'e' option, that is related to my updates. + ................ + +2008-08-22 20:21 +0000 [r139458-139564] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/threadstorage.h, /: Merged revisions 139554 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500 + (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug + 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is + selected (closes issue #13298) Reported by: snuffy Patches: + bug13298_20080822.diff uploaded by snuffy (license 35) ........ + ................ + + * /, channels/chan_iax2.c: Merged revisions 139469 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500 + (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug + 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........ + ................ + + * /, channels/chan_iax2.c: Merged revisions 139457 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500 + (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug + 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from + incorrect locking order between iax2_pvt and ast_channel + structures. AST-13 ........ ................ + +2008-08-21 23:46 +0000 [r139400] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500 + (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008) + | 3 lines Fixes loop that could possibly never exit in the event + of a channel never being able to be opened or specify after a + restart. (closes issue #11017) ........ ................ + +2008-08-21 10:02 +0000 [r139282] Philippe Sultan <philippe.sultan@gmail.com> + + * /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008) + | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel! + (closes issue #13310) Reported by: eliel Patches: + chan_gtalk.c.patch uploaded by eliel (license 64) ........ + +2008-08-20 Kevin P. Fleming <kpfleming@digium.com> + + * Asterisk 1.6.0-rc3 released. + +2008-08-20 22:17 +0000 [r139216] Russell Bryant <russell@digium.com> + + * apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008) + | 19 lines Merged revisions 139213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) + | 11 lines Fix a crash in the ChanSpy application. The issue here + is that if you call ChanSpy and specify a spy group, and sit in + the application long enough looping through the channel list, you + will eventually run out of stack space and the application with + exit with a seg fault. The backtrace was always inside of a + harmless snprintf() call, so it was tricky to track down. + However, it turned out that the call to snprintf() was just the + biggest stack consumer in this code path, so it would always be + the first one to hit the boundary. (closes issue #13338) Reported + by: ruddy ........ ................ + +2008-08-20 20:12 +0000 [r139155] Shaun Ruffell <sruffell@digium.com> + + * codecs/codec_dahdi.c: Fix bug where the samples were not accurate + when in G723 mode, which would cause the timestamp field of the + RTP header to be invalid. + +2008-08-20 17:30 +0000 [r139104] Steve Murphy <murf@digium.com> + + * main/cdr.c, /: Merged revisions 139083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) | + 20 lines Merged revisions 139074 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | + 12 lines (closes issue #13263) Reported by: brainy Tested by: + murf The specialized reset routine is tromping on the flags field + of the CDR. I made a change to not reset the DISABLED bit. This + should get rid of this problem. ........ ................ + +2008-08-20 15:39 +0000 [r138889-139017] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug + 2008) | 14 lines Merged revisions 139015 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug + 2008) | 6 lines sip_read should properly handle a NULL return + from sip_rtp_read. (closes issue #13257) Reported by: travishein + ........ ................ + + * apps/app_chanspy.c: Manually add revision 138887 from trunk to + the 1.6.0 branch. I had misunderstood the policy for when to + merge to 1.6.0 since it moved to rc status. + +2008-08-19 16:38 +0000 [r138846-138847] Steve Murphy <murf@digium.com> + + * utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y, + res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug + 2008) | 1 line Oops. put a decl in a generated file. My bad, but + fixed now. ........ + + * main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y, + res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 | + murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines + These changes are in regards to bug 13249, where users are being + surprised by the changes made to the Set app in trunk/1.6.x, as + they come from the 1.4 world. They are only bitten if they write + their AEL dialplan in the 1.4 world, and then carry it over to a + trunk/1.6.x installation where a "make samples" was executed, or + where they hand-edited the asterisk.conf file and added the + [compat] category with app_set = 1.6 (or higher). (this commit + does not totally solve 13249, at least not yet) The change + involves issueing a single warning while the AEL file is loading, + if: 1. app_set is present in the config file, and set to 1.6 or + higher. 2. there are double quotes in an assignment statement (eg + x = "hi there";) 3. the warning was not already issued. The + standalone app, aelparse, does not (yet) issue this warning. I'd + have to have it read in the asterisk.conf file, and that's a bit + of hassle. I'll add it if users request it, tho. ........ + +2008-08-19 00:15 +0000 [r138776-138781] Sean Bright <sean.bright@gmail.com> + + * /, channels/chan_sip.c: Merged revisions 138778-138780 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, + 18 Aug 2008) | 1 line While we're at it, make this machine + parseable too. ........ r138779 | seanbright | 2008-08-18 + 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we + don't need anymore. ........ r138780 | seanbright | 2008-08-18 + 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now, + too (woops) ........ + + * /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 | + seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3 + lines Change event header to RegistrationTime to be more + consistent (and avoid breaking existing frameworks). Pointed out + by Laureano on #asterisk-dev. ........ + +2008-08-18 20:23 +0000 [r138688-138695] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 138687 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug + 2008) | 18 lines Merged revisions 138685 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug + 2008) | 10 lines Change the inequalities used in app_queue with + regards to timeouts from being strict to non-strict for more + accuracy. (closes issue #13239) Reported by: atis Patches: + app_queue_timeouts_v2.patch uploaded by atis (license 242) + ........ ................ + +2008-08-18 15:54 +0000 [r138632] Jason Parker <jparker@digium.com> + + * Makefile, /: Merged revisions 138631 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 | + qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line + Remove option that isn't valid here. ........ + +2008-08-18 02:14 +0000 [r138519] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008) + | 1 line add missing define for SS7 in dahdi_restart ........ + +2008-08-17 14:14 +0000 [r138443-138483] Sean Bright <sean.bright@gmail.com> + + * /, main/features.c: Merged revisions 138482 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 | + seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6 + lines Move Uniqueid to the end of the event for those that rely + on the position of the name/value pairs, pointed out by + snuffy-home on #asterisk-commits. For those of you who rely on + the position of name/value pairs in manager events... stop... + that is why associative arrays were invented. ........ + + * /, main/features.c: Merged revisions 138479 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 | + seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7 + lines Add Uniqueid header to ParkedCall manager event. (closes + issue #13323) Reported by: srt Patches: + 13323_unique_id_for_parkedcalls_event.diff uploaded by srt + (license 378) ........ + + * main/rtp.c, /: Merged revisions 138476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 | + seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7 + lines Add missing colons to RTCPReceived and RTCPSent manager + events. (closes issue #13319) Reported by: srt Patches: + 13319_rtcp_manager_event_headers.diff uploaded by srt (license + 378) ........ + + * /, channels/chan_iax2.c: Merged revisions 138473 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug + 2008) | 7 lines Fix the output of the JitterBufStats manager + event. (closes issue #13324) Reported by: srt Patches: + 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt + (license 378) ........ + + * configs/cdr_tds.conf.sample, /: Merged revisions 138442 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat, + 16 Aug 2008) | 4 lines Since it's introduction in revision 3497, + cdr_tds has *never* read the port configuration option from + cdr_tds.conf. So go ahead and remove it from the sample config. + ........ + +2008-08-16 13:07 +0000 [r138410-138413] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008) + | 2 lines Fix compilation warnings (found with dev-mode) ........ + +2008-08-16 01:14 +0000 [r138333-138362] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500 + (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15 + Aug 2008) | 1 line fixes use count to properly decrement if an + active dahdi channel is destroyed allowing module to be unloaded + ........ ................ + + * channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500 + (Fri, 15 Aug 2008) | 20 lines Merged revisions + 138119,138151,138238 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) + | 4 lines Fixes the dahdi restart functionality. Dahdi restart + allows one to restart all DAHDI channels, even if they are + currently in use. This is different from unloading and then + loading the module since unloading requires the use count to be + zero. Reloading the module is different in that the signalling is + not changed from what it was originally configured. Also, this + fixes not closing all the file descriptors for D-channels upon + module unload (which would prevent loading the module + afterwards). (closes issue #11017) ........ r138151 | jpeeler | + 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared + static mutexes using AST_MUTEX_DEFINE_STATIC macro ........ + r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) + | 1 line initialize condition variable ss_thread_complete using + ast_cond_init ........ ................ + +2008-08-15 23:03 +0000 [r138207-138262] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 138260 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) + | 16 lines Merged revisions 138258 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) + | 8 lines More fixes for realtime peers. (closes issue #12921) + Reported by: Nuitari Patches: 20080804__bug12921.diff.txt + uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt + uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ + ................ + + * configs/extensions.conf.sample, main/pbx.c, /: Merged revisions + 138206 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 | + tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines + Remove deprecated syntax from sample config file (closes issue + #13314) Reported by: kue ........ + +2008-08-15 20:20 +0000 [r138156-138157] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to + dfd to match 1.4 (left over from DAHDI transition) + +2008-08-15 15:12 +0000 [r138029] Russell Bryant <russell@digium.com> + + * main/autoservice.c, /: Merged revisions 138028 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008) + | 17 lines Merged revisions 138027 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) + | 9 lines Ensure that when a hangup occurs in autoservice, that a + hangup frame gets properly deferred to be read from the channel + owner when it gets taken out of autoservice. (closes issue + #12874) Reported by: dimas Patches: v1-12874.patch uploaded by + dimas (license 88) ........ ................ + +2008-08-15 15:04 +0000 [r138025] Tilghman Lesher <tlesher@digium.com> + + * /, funcs/func_strings.c: Merged revisions 138024 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500 + (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008) + | 8 lines Additional check for more string specifiers than + arguments. (closes issue #13299) Reported by: adomjan Patches: + 20080813__bug13299.diff.txt uploaded by Corydon76 (license 14) + func_strings.c-sprintf.patch uploaded by adomjan (license 487) + Tested by: adomjan ........ ................ + +2008-08-14 22:43 +0000 [r137988] Russell Bryant <russell@digium.com> + + * /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 | + russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines + Fix a bashism that causes an error when trying to build the pdf + on ubuntu ........ + +2008-08-14 18:48 +0000 [r137934] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug + 2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes + issue #13304) Reported by: eliel Patches: sqlite.patch uploaded + by eliel (license 64) (Slightly modified by me) ........ + +2008-08-14 17:01 +0000 [r137849-137852] Tilghman Lesher <tlesher@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500 + (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) + | 9 lines When creating the secondary subchannel name, it is + necessary to compare to the existing channel name without the + "Zap/" or "DAHDI/" prefix, since our test string is also without + that prefix. (closes issue #13027) Reported by: dferrer Patches: + chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525) + (Slightly modified by me, to compensate for both names) ........ + ................ + +2008-08-14 Jason Parker <jparker@digium.com> + + * Asterisk 1.6.0-rc2 released. + +2008-08-14 15:37 +0000 [r137814] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 | + qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines + Make sure we set the socket port, so we don't try to use <ip + address>:0. (closes issue #13255) Reported by: falves11 Patches: + 13255-socketport.diff uploaded by qwell (license 4) Tested by: + falves11 ........ + +2008-08-14 15:20 +0000 [r137783] Russell Bryant <russell@digium.com> + + * /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r137732 | russell | 2008-08-14 09:15:50 -0500 + (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) + | 4 lines Comments in this config file were aligned only if your + tab size was set to 8. So, convert tabs to spaces so that things + should be aligned regardless of what tab size you use in your + editor. ........ ................ + +2008-08-14 15:05 +0000 [r137781] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 | + seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8 + lines If we detect that we are no longer connected, try to + reconnect a few times before giving up. This relies on the + timeout settings in the freetds.conf file and, unfortunately, on + a recent version of FreeTDS (0.82 or newer). I either need to + change the current execs to be non-blocking (which I do not want + to do) or we have to force people to run with the latest and + greatest of FreeTDS. I'm on the fence... ........ + +2008-08-14 02:04 +0000 [r137681] Kevin P. Fleming <kpfleming@digium.com> + + * /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug + 2008) | 9 lines Merged revisions 137679 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug + 2008) | 1 line forgot one module name that changed ........ + ................ + +2008-08-13 Kevin P. Fleming <kpfleming@digium.com> + + * Asterisk 1.6.0-rc1 released. + +2008-08-13 23:00 +0000 [r137631-137641] Kevin P. Fleming <kpfleming@digium.com> + + * /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug + 2008) | 1 line make this script actually work ........ + + * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions + 137627 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug + 2008) | 9 lines Merged revisions 137530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug + 2008) | 1 line add document describing what users will need to be + aware of when upgrading to this version and using DAHDI ........ + ................ + +2008-08-13 21:09 +0000 [r137497-137533] Jason Parker <jparker@digium.com> + + * /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 | + qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines + Correctly end locally ended calls. (closes issue #12170) Reported + by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff + uploaded by bbryant (license 36) Tested by: bbryant, pabelanger + ........ + + * /, apps/app_fax.c: Merged revisions 137496 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 | + qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines + Add FAXMODE variable with what fax transport was used. (closes + issue #13252) Patches: v1-13252.patch uploaded by dimas (license + 88) ........ + +2008-08-13 14:47 +0000 [r137350-137407] Sean Bright <sean.bright@gmail.com> + + * /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400 + (Wed, 13 Aug 2008) | 9 lines Merged revisions 137405 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, + 13 Aug 2008) | 1 line Update docs to reflect the change to + cdr_tds ........ ................ + + * cdr/cdr_tds.c, /: Merged revisions 137403 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 | + seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1 + line Use the ast_vasprintf macro instead of vasprintf directly. + ........ + +2008-08-12 19:48 +0000 [r137300-137302] Russell Bryant <russell@digium.com> + + * doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008) + | 2 lines Grammar hax from Qwell ........ + + * doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008) + | 3 lines Note that developer documentation belongs in doxygen, + and not integrated with the user manual stuff in doc/tex/. + ........ + +2008-08-11 16:15 +0000 [r137240] Russell Bryant <russell@digium.com> + + * Makefile, /: Merged revisions 137239 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 | + russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines + Make PRINT_DIR work as advertised. ........ + +2008-08-11 14:31 +0000 [r137217] Sean Bright <sean.bright@gmail.com> + + * cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon, + 11 Aug 2008) | 7 lines Log the userfield CDR variable like the + other CDR backends, assuming the column is actually there. If + it's not, we still log everything else as before. (closes issue + #13281) Reported by: falves11 ........ + +2008-08-11 00:27 +0000 [r137160] Tilghman Lesher <tlesher@digium.com> + + * res/res_odbc.c, /: Merged revisions 137150 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008) + | 13 lines Merged revisions 137138 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008) + | 5 lines Deallocate database connection handle on disconnect, as + we allocate another one on connect. (closes issue #13271) + Reported by: dveiga ........ ................ + +2008-08-09 15:27 +0000 [r136948] Tilghman Lesher <tlesher@digium.com> + + * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged + revisions 136947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008) + | 18 lines Merged revisions 136946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500 + (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) + | 2 lines Regression fixes for Solaris ........ ................ + ................ + +2008-08-09 01:16 +0000 [r136860] Tilghman Lesher <tlesher@digium.com> + + * /, res/res_agi.c: Merged revisions 136859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 | + tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines + Update documentation as to the behavior of AGI in 1.6.0 and + higher. Also, add an OOB message that answers the question of, if + AGI no longer shuts down the connection on hangup, how will + FastAGI know when to stop processing the call? ........ + +2008-08-08 15:33 +0000 [r136785] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug + 2008) | 3 lines Fix compilation for ODBC voicemail ........ + +2008-08-08 06:45 +0000 [r136778] Steve Murphy <murf@digium.com> + + * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, + pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, /, + pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h, + utils/ael_main.c: Merged revisions 136746 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) | + 40 lines Merged revisions 136726 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | + 32 lines (closes issue #13236) Reported by: korihor Wow, this one + was a challenge! I regrouped and ran a new strategy for setting + the ~~MACRO~~ value; I set it once per extension, up near the + top. It is only set if there is a switch in the extension. So, I + had to put in a chunk of code to detect a switch in the pval + tree. I moved the code to insert the set of ~~exten~~ up to the + beginning of the gen_prios routine, instead of down in the switch + code. I learned that I have to push the detection of the switches + down into the code, so everywhere I create a new exten in + gen_prios, I make sure to pass onto it the values of the + mother_exten first, and the exten next. I had to add a couple + fields to the exten struct to accomplish this, in the + ael_structs.h file. The checked field makes it so we don't repeat + the switch search if it's been done. I also updated the + regressions. ........ ................ + +2008-08-08 02:36 +0000 [r136753] Tilghman Lesher <tlesher@digium.com> + + * /: Merged revisions 136751 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 | + tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines + Removing bad properties ........ + +2008-08-07 23:42 +0000 [r136719-136724] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a + bunch of functions over one level during a merge. + + * apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug + 2008) | 3 lines Remove one last batch of debug messages ........ + + * apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug + 2008) | 18 lines Merging the imap_consistency_trunk branch to + trunk. For an explanation of what "imap_consistency" is, please + see svn revision 134223 to the 1.4 branch. Coincidentally, this + also fixes a recent bug report regarding the inability to save + messages to the new folder when using IMAP storage since they + will would be flagged as "seen" and not be recognized as new + messages. (closes issue #13234) Reported by: jaroth ........ + +2008-08-07 20:41 +0000 [r136672-136674] Shaun Ruffell <sruffell@digium.com> + + * codecs/codec_dahdi.c: Removing code that was commented out. + + * codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder + interface in the DAHDI. (Issue: DAHDI-42) + +2008-08-07 20:26 +0000 [r136632-136663] Mark Michelson <mmichelson@digium.com> + + * /, main/features.c: Merged revisions 136660 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 | + mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4 + lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears + once for every bridged call ........ + + * main/pbx.c, /: Merged revisions 136635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 | + mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5 + lines Don't allow Answer() to accept a negative argument. + Negative argument means an infinite delay and we don't want that. + ........ + + * main/channel.c, /: Merged revisions 136633 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 | + mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7 + lines Fix a calculation error I had made in the poll. The poll + would reset to 500 ms every time a non-voice frame was received. + The total time we poll should be 500 ms, so now we save the + amount of time left after the poll returned and use that as our + argument for the next call to poll ........ + + * main/channel.c, /: Merged revisions 136631 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 | + mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13 + lines Scrap the 500 ms delay when Asterisk auto-answers a + channel. Instead, poll the channel until receiving a voice frame. + The cap on this poll is 500 ms. The optional delay is still + allowable in the Answer() application, but the delay has been + moved back to its original position, after the call to the + channel's answer callback. The poll for the voice frame will not + happen if a delay is specified when calling Answer(). (closes + issue #12708) Reported by: kactus ........ + +2008-08-07 19:19 +0000 [r136598] Richard Mudgett <rmudgett@digium.com> + + * channels/misdn_config.c, channels/chan_misdn.c, /, + configs/misdn.conf.sample: Merged revisions 136594 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500 + (Thu, 07 Aug 2008) | 13 lines Merged revisions 136241 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008) + | 5 lines * The allowed_bearers setting in misdn.conf misspelled + one of its options: digital_restricted. * Fixed some other + spelling errors and typos. ........ ................ + +2008-08-07 17:44 +0000 [r136506-136543] Kevin P. Fleming <kpfleming@digium.com> + + * include/asterisk/doxyref.h, /: Merged revisions 136542 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500 + (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + ........ ................ + +2008-08-07 16:57 +0000 [r136490] Tilghman Lesher <tlesher@digium.com> + + * /, apps/app_queue.c: Merged revisions 136489 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008) + | 15 lines Merged revisions 136488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008) + | 7 lines Update persistent state on all exit conditions. (closes + issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt + uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon + ........ ................ + +2008-08-06 20:16 +0000 [r136113-136192] Tilghman Lesher <tlesher@digium.com> + + * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500 + (Wed, 06 Aug 2008) | 12 lines Merged revisions 136190 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008) + | 4 lines -C option takes a filename, not a directory path. + (closes issue #13007) Reported by: klaus3000 ........ + ................ + + * /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008) + | 7 lines Persist DIALGROUP() values in astdb (closes issue + #13138) Reported by: Corydon76 Patches: + 20080725__bug13138.diff.txt uploaded by Corydon76 (license 14) + Tested by: pj ........ + +2008-08-06 16:00 +0000 [r136064] Mark Michelson <mmichelson@digium.com> + + * main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500 + (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug + 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame + type, there are places where ast_rtp_new_source may be called + where the tech_pvt of a channel may not yet have an rtp structure + allocated. This caused a crash in chan_skinny, which was fixed + earlier, but now the same crash has been reported against + chan_h323 as well. It seems that the best solution is to modify + ast_rtp_new_source to not attempt to set the marker bit if the + rtp structure passed in is NULL. This change to + ast_rtp_new_source also allows the removal of what is now a + redundant pointer check from chan_skinny. (closes issue #13247) + Reported by: pj ........ ................ + +2008-08-06 13:59 +0000 [r136006] Olle Johansson <oej@edvina.net> + + * /, res/res_jabber.c: Merged revisions 136005 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 | + oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines - + Formatting - Changing debug messages from VERBOSE to DEBUG + channel - Adding a few todo's - Adding a few more "XMPP"'s to + compliment Jabber... ........ + +2008-08-06 03:56 +0000 [r135901-135951] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /: Merged revisions 135950 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008) + | 12 lines Merged revisions 135949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) + | 4 lines Fix a longstanding bug in channel walking logic, and + fix the explanation to make sense. (Closes issue #13124) ........ + ................ + + * /, main/translate.c: Merged revisions 135938 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008) + | 12 lines Merged revisions 135915 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008) + | 4 lines Since powerof() can return an error condition, it's + foolhardy not to detect and deal with that condition. (Related to + issue #13240) ........ ................ + + * include/asterisk/threadstorage.h, include/asterisk/utils.h, /: + Merged revisions 135900 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008) + | 12 lines Merged revisions 135899 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) + | 4 lines 1) Bugfix for debugging code 2) Reduce compiler + warnings for another section of debugging code (Closes issue + #13237) ........ ................ + +2008-08-06 00:31 +0000 [r135852] Mark Michelson <mmichelson@digium.com> + + * include/asterisk/abstract_jb.h, main/channel.c, /, + main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions + 135851 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug + 2008) | 48 lines Merged revisions 135841,135847,135850 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug + 2008) | 27 lines Merging the issue11259 branch. The purpose of + this branch was to take into account "burps" which could cause + jitterbuffers to misbehave. One such example is if the L option + to Dial() were used to inject audio into a bridged conversation + at regular intervals. Since the audio here was not passed through + the jitterbuffer, it would cause a gap in the jitterbuffer's + timestamps which would cause a frames to be dropped for a brief + period. Now ast_generic_bridge will empty and reset the + jitterbuffer each time it is called. This causes injected audio + to be handled properly. ast_generic_bridge also will empty and + reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE + frame since the change in audio source could negatively affect + the jitterbuffer. All of this was made possible by adding a new + public API call to the abstract_jb called ast_jb_empty_and_reset. + (closes issue #11259) Reported by: plack Tested by: putnopvut + ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, + 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel + that occurred when I was testing for a memory leak ........ + r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug + 2008) | 3 lines Remove properties that should not be here + ........ ................ + +2008-08-05 23:52 +0000 [r135822] Steve Murphy <murf@digium.com> + + * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c, + include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | + 42 lines Merged revisions 135799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | + 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf + I discovered that also, in the previous bug fixes and changes, + the cdr.conf 'unanswered' option is not being obeyed, so I fixed + this. And, yes, there are two 'answer' times involved in this + scenario, and I would agree with you, that the first answer time + is the time that should appear in the CDR. (the second 'answer' + time is the time that the bridge was begun). I made the necessary + adjustments, recording the first answer time into the peer cdr, + and then using that to override the bridge cdr's value. To get + the 'unanswered' CDRs to appear, I purposely output them, using + the dial cmd to mark them as DIALED (with a new flag), and + outputting them if they bear that flag, and you are in the right + mode. I also corrected one small mention of the Zap device to + equally consider the dahdi device. I heavily tested 10-sec-wait + macros in dial, and without the macro call; I tested hangups + while the macro was running vs. letting the macro complete and + the bridge form. Looks OK. Removed all the instrumentation and + debug. ........ ................ + +2008-08-05 21:38 +0000 [r135749] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500 + (Tue, 05 Aug 2008) | 17 lines Merged revisions 135747 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008) + | 9 lines In a conversion to use ast_strlen_zero, the meaning of + the flag IAX_HASCALLERID was perverted. This change reverts IAX2 + to the original meaning, which was, that the callerid set on the + client should be overridden on the server, even if that means the + resulting callerid is blank. In other words, if you set + "callerid=" in the IAX config, then the callerid should be + overridden to blank, even if set on the client. Note that there's + a distinction, even on realtime, between the field not existing + (NULL in databases) and the field existing, but set to blank + (override callerid to blank). ........ ................ + +2008-08-05 13:27 +0000 [r135599] Sean Bright <sean.bright@gmail.com> + + * main/cli.c, /: Merged revisions 135598 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug + 2008) | 9 lines Merged revisions 135597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug + 2008) | 1 line Use PATH_MAX for filenames ........ + ................ + +2008-08-04 20:15 +0000 [r135538] Russell Bryant <russell@digium.com> + + * configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135537 | russell | 2008-08-04 15:15:27 -0500 + (Mon, 04 Aug 2008) | 10 lines Merged revisions 135536 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) + | 2 lines fix a config sample typo ........ ................ + +2008-08-04 17:12 +0000 [r135478-135486] Tilghman Lesher <tlesher@digium.com> + + * contrib/init.d/rc.mandriva.asterisk (added), Makefile, + contrib/init.d/rc.mandrake.asterisk (removed), /, + contrib/init.d/rc.mandriva.zaptel (added), + contrib/init.d/rc.mandrake.zaptel (removed): Merged revisions + 135485 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 | + tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines + Rename Mandrake scripts to Mandriva (Closes issue #13221) + ........ + + * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500 + (Mon, 04 Aug 2008) | 11 lines Merged revisions 135482 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008) + | 2 lines Define ASTSBINDIR for script (Closes issue #13221) + ........ ................ + + * apps/app_voicemail.c, /: Merged revisions 135480 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500 + (Mon, 04 Aug 2008) | 14 lines Merged revisions 135479 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008) + | 6 lines Memory leak on unload (closes issue #13231) Reported + by: eliel Patches: app_voicemail.leak.patch uploaded by eliel + (license 64) ........ ................ + +2008-08-04 16:28 +0000 [r135440-135475] Russell Bryant <russell@digium.com> + + * configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135474 | russell | 2008-08-04 11:28:07 -0500 + (Mon, 04 Aug 2008) | 10 lines Merged revisions 135473 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) + | 2 lines Add a minor clarification to the documentation of + mohinterpret and mohsuggest ........ ................ + + * /, channels/chan_console.c: Merged revisions 135439 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008) + | 4 lines Be explicit that we don't want a result from this + callback. The callback would never indicate a match, so nothing + would have been returned anyway, but it was still a poor example + of proper usage. ........ + +2008-08-02 05:15 +0000 [r135266] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 135265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 | + murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines + (closes issue #13202) Reported by: falves11 Tested by: murf + falves11 == The changes I introduce here seem to clear up the + problem for me. However, if they do not for you, please reopen + this bug, and we'll keep digging. The root of this problem seems + to be a subtle memory corruption introduced when creating an + extension with an empty extension name. While valgrind cannot + detect it outside of DEBUG_MALLOC mode, when compiled with + DEBUG_MALLOC, this is certain death. The code in main/features.c + is a puzzle to me. On the initial module load, the code is + attempting to add the parking extension before the features.conf + file has even been opened! I just wrapped the offending call with + an if() that will not try to add the extension if the extension + name is empty. THis seems to solve the corruption, and let the + "memory show allocations" work as one would expect. But, really, + adding an extension with an empty name is a seriously bad thing + to allow, as it will mess up all the pattern matching algorithms, + etc. So, I added a statement to the add_extension2 code to return + a -1 if this is attempted. in 1.6.0, the changes to only + main/pbx.c were applicable, as apparently the code added to + main/features by jpeeler were not included in 1.6.0. ........ + +2008-08-01 19:30 +0000 [r135198] Sean Bright <sean.bright@gmail.com> + + * channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug + 2008) | 6 lines Remove some code that used to do something but + does not anymore, mainly to get rid of a shadow warning (but this + seemed legitimate enough to fix here instead of in my branch). + Thanks to putnopvut for taking a look as well. ........ + +2008-08-01 17:10 +0000 [r135127-135129] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 | + tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines + Picky, picky, buildbot ........ + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 135126 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 | + tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines + SIP should use the transport type set in the Moved Temporarily + for the next invite. (closes issue #11843) Reported by: + pestermann Patches: + 20080723__issue11843_302_ignores_transport_16branch.diff uploaded + by bbryant (license 36) + 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by + bbryant (license 36) Tested by: pabelanger ........ + +2008-08-01 14:43 +0000 [r135070] Mark Michelson <mmichelson@digium.com> + + * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged + revisions 135067-135068 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 | + mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13 + lines IMAP storage functioned under the assumption that folders + such as "Work" and "Family" would be subfolders of the INBOX. + This is an invalid assumption to make, but it could be desirable + to set up folders in this manner, so a new option for + voicemail.conf, "imapparentfolder" has been added to allow for + this. (closes issue #13142) Reported by: jaroth Patches: + parentfolder.patch uploaded by jaroth (license 50) ........ + r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug + 2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE + defines... ........ + +2008-08-01 12:18 +0000 [r135057-135062] Michiel van Baak <michiel@vanbaak.info> + + * /, apps/app_ices.c: Merged revisions 135059 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008) + | 10 lines Merged revisions 135058 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008) + | 2 lines make app_ices compile on OpenBSD. ........ + ................ + + * /, channels/chan_skinny.c: Merged revisions 135056 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200 + (Fri, 01 Aug 2008) | 16 lines Merged revisions 135055 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008) + | 8 lines fix some potential deadlocks in chan_skinny (closes + issue #13215) Reported by: qwell Patches: + 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7) + Tested by: mvanbaak ........ ................ + +2008-07-31 22:34 +0000 [r135034] Kevin P. Fleming <kpfleming@digium.com> + + * /, main/http.c: Merged revisions 135016 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul + 2008) | 11 lines Merged revisions 134983 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul + 2008) | 3 lines accomodate users who seem to lack a sense of + humor :-) ........ ................ + +2008-07-31 21:58 +0000 [r134926-134981] Tilghman Lesher <tlesher@digium.com> + + * sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions + 134980 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008) + | 16 lines Blocked revisions 134976 via svnmerge ........ r134976 + | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9 + lines Specify codecs in callfiles and manager, to allow video + calls to be set up from callfiles and AMI. (closes issue #9531) + Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt + uploaded by Corydon76 (license 14) + 20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license + 14) Tested by: Corydon76 ........ ................ + + * res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008) + | 2 lines Switch command order, to meet with current specs + ........ + +2008-07-31 19:54 +0000 [r134923] Steve Murphy <murf@digium.com> + + * /, main/features.c: Merged revisions 134922 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) | + 63 lines Merged revisions 134883 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | + 51 lines (closes issue #11849) Reported by: greyvoip Tested by: + murf OK, a few days of debugging, a bunch of instrumentation in + chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook + pages of notes later, I have made the small tweek necc. to get + the start time right on the second CDR when: A Calls B B answ. A + hits Xfer button on sip phone, A dials C and hits the OK button, + A hangs up C answers ringing phone B and C converse B and/or C + hangs up But does not harm the scenario where: A Calls B B answ. + B hits xfer button on sip phone, B dials C and hits the OK + button, B hangs up C answers ringing phone A and C converse A + and/or C hangs up The difference in start times on the second CDR + is because of a Masquerade on the B channel when the xfer number + is sent. It ends up replacing the CDR on the B channel with a + duplicate, which ends up getting tossed out. We keep a pointer to + the first CDR, and update *that* after the bridge closes. But, + only if the CDR has changed. I hope this change is specific + enough not to muck up any current CDR-based apps. In my defence, + I assert that the previous information was wrong, and this change + fixes it, and possibly other similar scenarios. I wonder if I + should be doing the same thing for the channel, as I did for the + peer, but I can't think of a scenario this might affect. I leave + it, then, as an exersize for the users, to find the scenario + where the chan's CDR changes and loses the proper start time. + ........ ................ + +2008-07-31 19:41 +0000 [r134918] Russell Bryant <russell@digium.com> + + * /, apps/app_ices.c: Merged revisions 134917 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008) + | 17 lines Merged revisions 134915 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008) + | 9 lines Get app_ices working again (closes issue #12981) + Reported by: dlogan Patches: + 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant + (license 36) 20080709__app_ices_v2_update_14.diff uploaded by + bbryant (license 36) Tested by: bbryant ........ ................ + +2008-07-31 16:53 +0000 [r134816] Russell Bryant <russell@digium.com> + + * channels/iax2-parser.c: Merged revisions 134815 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008) + | 15 lines Merged revisions 134814 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008) + | 7 lines In case we have some processing threads that free more + frames than they allocate, do not let the frame cache grow + forever. (closes issue #13160) Reported by: tavius Tested by: + tavius, russell ........ ................ + +2008-07-31 16:07 +0000 [r134760] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 134759 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul + 2008) | 24 lines Merged revisions 134758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul + 2008) | 16 lines Add more timeout checks into app_queue, + specifically targeting areas where an unknown and potentially + long time has just elapsed. Also added a check to try_calling() + to return early if the timeout has elapsed instead of potentially + setting a negative timeout for the call (thus making it have *no* + timeout at all). (closes issue #13186) Reported by: + miquel_cabrespina Patches: 13186.diff uploaded by putnopvut + (license 60) Tested by: miquel_cabrespina ........ + ................ + +2008-07-30 22:41 +0000 [r134651-134707] Tilghman Lesher <tlesher@digium.com> + + * main/sched.c, /, include/asterisk/sched.h: Merged revisions + 134703 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 | + tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines + Oops, wrong define ........ + + * /, configure, configure.ac: Merged revisions 134650 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500 + (Wed, 30 Jul 2008) | 12 lines Merged revisions 134649 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008) + | 4 lines Qwell pointed out, via IRC, that the previous fix only + worked when explicitly set. When nothing is set, and the option + is implied, it breaks, because configure sets the prefix to + 'NONE'. Fixing. ........ ................ + +2008-07-30 21:06 +0000 [r134599] Mark Michelson <mmichelson@digium.com> + + * /, apps/app_queue.c: Merged revisions 134598 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul + 2008) | 15 lines Merged revisions 134556 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | + mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 + lines Fix the parsing of the "reason" parameter in the Diversion: + header. (closes issue #13195) Reported by: woodsfsg ........ + ................ + +2008-07-30 20:39 +0000 [r134597] Russell Bryant <russell@digium.com> + + * /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008) + | 14 lines Merged revisions 134595 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008) + | 6 lines Reduce stack consumption by 12.5% of the max stack size + to fix a crash when compiled with LOW_MEMORY. (closes issue + #13154) Reported by: edantie ........ ................ + +2008-07-30 20:25 +0000 [r134561] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 134556 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | + mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 + lines Fix the parsing of the "reason" parameter in the Diversion: + header. (closes issue #13195) Reported by: woodsfsg ........ + +2008-07-30 19:56 +0000 [r134542] Russell Bryant <russell@digium.com> + + * funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008) + | 12 lines Merged revisions 134540 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008) + | 4 lines Fix a memory leak in func_curl. Every thread that used + this function leaked an allocation the size of a pointer. + (reported by jmls in #asterisk-dev) ........ ................ + +2008-07-30 19:49 +0000 [r134482-134539] Tilghman Lesher <tlesher@digium.com> + + * /, configure, configure.ac: Merged revisions 134538 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500 + (Wed, 30 Jul 2008) | 12 lines Merged revisions 134536 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008) + | 4 lines Only override sysconfdir and mandir when prefix=/usr + (closes issue #13093) Reported by: pabelanger ........ + ................ + + * /, apps/app_queue.c: Merged revisions 134483 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 | + tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines + Let "roundrobin" also reference rrmemory, for the 1.6 release (as + described in UPGRADE-1.4.txt) (Closes issue #13181) ........ + + * /, res/res_agi.c: Merged revisions 134481 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008) + | 13 lines Merged revisions 134480 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008) + | 5 lines launch_netscript sometimes returns -1, which fails to + set AGISTATUS. Map failure to -1, so that AGISTATUS is always + set. (closes issue #13199) Reported by: smw1218 ........ + ................ + +2008-07-30 18:33 +0000 [r134477] Mark Michelson <mmichelson@digium.com> + + * /, main/app.c: Merged revisions 134476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul + 2008) | 12 lines Merged revisions 134475 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul + 2008) | 4 lines Fix a spot where a function could return without + bringing a channel out of autoservice. ........ ................ + +2008-07-30 15:34 +0000 [r134356] Kevin P. Fleming <kpfleming@digium.com> + + * Makefile, /: Merged revisions 134355 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul + 2008) | 10 lines Merged revisions 134352 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul + 2008) | 2 lines use the proper method for building version.h + ........ ................ + +2008-07-29 22:29 +0000 [r134283] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /, + apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c: + Merged revisions 134260 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 | + kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2 + lines build against the now-typedef-free dahdi/user.h, and remove + some #ifdefs for features that will always be present in DAHDI + ........ + +2008-07-28 22:16 +0000 [r134164] Tilghman Lesher <tlesher@digium.com> + + * apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500 + (Mon, 28 Jul 2008) | 15 lines Merged revisions 134161 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008) + | 7 lines Detect when sox fails to raise the volume, because sox + can't read the file. (closes issue #12939) Reported by: + rickbradley Patches: 20080728__bug12939.diff.txt uploaded by + Corydon76 (license 14) Tested by: rickbradley ........ + ................ + +2008-07-28 19:55 +0000 [r134126] Mark Michelson <mmichelson@digium.com> + + * /, configure, main/Makefile, configure.ac, CHANGES: Merged + revisions 134125 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 | + mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27 + lines This commit compensates for buggy poll(2) implementations. + Asterisk has, for a long time, had its own implementation of + poll(2) which just used the input arguments to call select(2). In + 1.4, this internal implementation was used for Darwin systems. + This was removed in Asterisk trunk at some point, but it seems as + though this was not the right move to make. On Mac OS X, it + appears as though the poll used to gather CLI input does not + respond properly when connecting via a remote Asterisk console. + Reverting to the use of Asterisk's poll fixed the issue. Also, + there is now an option for the configure script, + --enable-internal-poll, which will allow for anyone to use + Asterisk's internal poll implementation in case they suspect that + their system's poll implementation is buggy. closes issue #11928) + Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded + by putnopvut (license 60) ........ + +2008-07-28 16:49 +0000 [r134087] Kevin P. Fleming <kpfleming@digium.com> + + * apps/app_parkandannounce.c, main/loader.c, sample.call, + contrib/scripts/autosupport, build_tools/cflags.xml, + main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c, + doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions + 134086 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 | + kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3 + lines remove remaining Zaptel references in various places + ........ + +2008-07-28 16:13 +0000 [r134052] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, + /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions + 134050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 | + mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 + lines merging the zap_and_dahdi_trunk branch up to trunk ........ + +2008-07-26 15:34 +0000 [r133942-133982] Russell Bryant <russell@digium.com> + + * main/asterisk.c, include/asterisk/doxyref.h, /: Include the + licensing page in 1.6.0 as well. Now, this page exists in 1.4, + trunk, and 1.6.0. + + * /: unblock 133575 + + * /, main/devicestate.c: Merged revisions 133945-133946 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26 + Jul 2008) | 6 lines ast_device_state() gets called in two + different ways. The first way is when called from elsewhere in + Asterisk to find the current state of a device. In that case, we + want to use the cached value if it exists. The other way is when + processing a device state change. In that case, we do not want to + check the cache because returning the last known state is counter + productive. ........ r133946 | russell | 2008-07-26 10:16:20 + -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache + argument ........ + +2008-07-25 22:09 +0000 [r133863-133905] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25 + Jul 2008) | 6 lines Update version (closes issue #13163) Reported + by: suretec Patches: asterisk.ldif uploaded by suretec (license + 70) ........ + +2008-07-25 19:37 +0000 [r133804-133806] Brandon Kruse <bkruse@digium.com> + + * /: Blocking revert of code changes in r133770 + + * main/http.c: Include the http_decode function from trunk to + replace the + with a space. + +2008-07-25 17:33 +0000 [r133694] Brandon Kruse <bkruse@digium.com> + + * /: Blocking a fix from trunk for the function http_decode. 1.6.0 + does not have this function. + +2008-07-25 17:26 +0000 [r133671] Tilghman Lesher <tlesher@digium.com> + + * main/channel.c, /, channels/chan_agent.c, main/devicestate.c: + Merged revisions 133665 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008) + | 16 lines Merged revisions 133649 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) + | 8 lines Fix some errant device states by making the devicestate + API more strict in terms of the device argument (only without the + unique identifier appended). (closes issue #12771) Reported by: + davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 + (license 14) Tested by: davidw, jvandal, murf ........ + ................ + +2008-07-25 15:01 +0000 [r133576-133580] Russell Bryant <russell@digium.com> + + * /, LICENSE: Merged revisions 133579 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008) + | 18 lines Merged revisions 133578 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r133578 | russell | 2008-07-25 10:00:31 -0500 + (Fri, 25 Jul 2008) | 10 lines Merged revisions 133577 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) + | 2 lines Fix the IAX2 URI for calling Digium ........ + ................ ................ + +2008-07-25 14:41 +0000 [r133571-133574] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul + 2008) | 15 lines Merged revisions 133572 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul + 2008) | 7 lines We need to make sure to null-terminate the "name" + portion of SIP URI parameters so that there are no bogus + comparisons. Thanks to bbryant for pointing this out. ........ + ................ + +2008-07-25 13:25 +0000 [r133567-133569] Russell Bryant <russell@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 | + russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines + Minor coding guidelines tweaks ... - Use ast_strlen_zero in one + place - check for successful string comparison the way most of + Asterisk code does it ........ + +2008-07-24 21:31 +0000 [r133524] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133509 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008) + | 11 lines Merged revisions 133488 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) + | 3 lines Fix rtautoclear and rtcachefriends (Closes issue + #12707) ........ ................ + +2008-07-24 20:41 +0000 [r133487] Russell Bryant <russell@digium.com> + + * /, channels/chan_agent.c: Merged revisions 133486 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008) + | 3 lines I made this change from DEVICE_STATE to + DEVICE_STATE_CHANGE, but I had it backwards, this is the right + event to subscribe to ... ........ + +2008-07-24 19:55 +0000 [r133449] Mark Michelson <mmichelson@digium.com> + + * /, main/logger.c: Merged revisions 133448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 | + mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12 + lines Print the correct PID in log messages. Prior to this + commit, only the logger thread's PID would be printed. (closes + issue #13150) Reported by: atis Patches: log_pid.diff uploaded by + putnopvut (license 60) Tested by: eliel ........ + +2008-07-24 05:21 +0000 [r133392-133405] Tilghman Lesher <tlesher@digium.com> + + * contrib/scripts/asterisk.logrotate, Makefile, /: Merged revisions + 133400 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133400 | + tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines + Build the logrotate script according to paths (Closes issue + #13147) ........ + + * Makefile, /: Merged revisions 133391 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133391 | + tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines + Optionally install logrotate file (Closes issue #13148) ........ + +2008-07-23 22:07 +0000 [r133300] Steve Murphy <murf@digium.com> + + * main/pbx.c, /: Merged revisions 133299 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 | + murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines + (closes issue #13144) Reported by: murf Tested by: murf For: J. + Geis The 'data' field in the ast_exten struct was being 'moved' + from the current dialplan to the replacement dialplan. This was + not good, as the current dialplan could have problems in the time + between the change and when the new dialplan is swapped in. So, I + modified the merge_and_delete code to strdup the 'data' field + (the args to the app call), and then it's freed as normal. I + improved a few messages; I added code to limit the number of + calls to the context_merge_incls_swits_igps_other_registrars() to + one per context. I don't think having it called multiple times + per context was doing anything bad, but it was inefficient. I + hope this fixes the problems Mr. Geiss was noting in + asterisk-users, see + http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html + ........ + +2008-07-23 21:50 +0000 [r133297] Jason Parker <jparker@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 133296 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r133296 | qwell | 2008-07-23 16:50:20 -0500 + (Wed, 23 Jul 2008) | 9 lines Merged revisions 133295 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul + 2008) | 1 line inbandrelease is gone - it's now inbanddisconnect + ........ ................ + +2008-07-23 20:39 +0000 [r133218] Brett Bryant <bbryant@digium.com> + + * /, channels/chan_sip.c: Merged revisions 133197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 | + bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines + Fix issue where tcp in sip is enabled by default, despite what it + says in the config sample file. Also fix "sip show settings" for + tcp connections. ........ + +2008-07-23 19:50 +0000 [r133042-133172] Mark Michelson <mmichelson@digium.com> + + * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, + /: Merged revisions 133171 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul + 2008) | 20 lines Merged revisions 133169 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul + 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN + in app_chanspy should be set at load time, not at compile time, + since dahdi_chan_name is determined at load time. Also changed + the next_unique_id_to_use to have the static qualifier. Also + added the dahdi_chan_name_len variable so that + strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for + the suggestion. ........ ................ + + * apps/app_chanspy.c, /: Merged revisions 133106 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul + 2008) | 13 lines Merged revisions 133104 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul + 2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is + twelve. The strncmp call in next_channel should account for this. + ........ ................ + + * apps/app_chanspy.c, /: Merged revisions 133102 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul + 2008) | 14 lines Merged revisions 133101 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul + 2008) | 6 lines Update the "last" channel in next_channel in + app_chanspy so that the same pseudo channel isn't constantly + returned. related to issue #13124 ........ ................ + + * channels/chan_dahdi.c, /: Merged revisions 133041 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r133041 | mmichelson | 2008-07-23 12:54:03 -0500 + (Wed, 23 Jul 2008) | 15 lines Merged revisions 133038 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul + 2008) | 7 lines Small cleanup. Move the declaration of the + DAHDI_SPANINFO variable to the block where it is used. This + allows one less #ifdef HAVE_PRI to clutter things up. Thanks to + Tzafrir for pointing this out on #asterisk-dev ........ + ................ + +2008-07-23 17:21 +0000 [r132978-132983] Tilghman Lesher <tlesher@digium.com> + + * /, channels/chan_iax2.c: Merged revisions 132981 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008) + | 6 lines Yet another conversion of '|' to ',' (closes issue + #13137) Reported by: eliel Patches: chan_iax2trunk-IAXPEER.patch + uploaded by eliel (license 64) ........ + + * contrib/scripts/asterisk.logrotate (added), /: Merged revisions + 132977 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132977 | + tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines + Add logrotate script for Asterisk (closes issue #13085) Reported + by: pabelanger Patches: logrotate uploaded by pabelanger (license + 224) ........ + +2008-07-23 16:42 +0000 [r132965-132967] Kevin P. Fleming <kpfleming@digium.com> + + * channels/misdn/isdn_lib.c, /: Merged revisions 132883,132966 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r132883 | crichter | 2008-07-23 07:07:15 -0500 + (Wed, 23 Jul 2008) | 9 lines Merged revisions 132826 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 + Jul 2008) | 1 line another Fix because of r119585, this commit + has broken high frequented BRI Ports, there was a possibility + that a channel, that was marked as in_use would be reused later, + the corresponding port could got stuck then. So it is recommended + to upgrade for chan_misdn users. ........ ................ + r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul + 2008) | 2 lines use correct function name... please compile with + --enable-dev-mode ................ + + * include/asterisk/stringfields.h, /, main/utils.c: Merged + revisions 132964 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul + 2008) | 10 lines Merged revisions 132872 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul + 2008) | 2 lines minor optimization for stringfields: when a field + is being set to a larger value than it currently contains and it + happens to be the most recent field allocated from the currentl + pool, it is possible to 'grow' it without having to waste the + space it is currently using (or potentially even allocate a new + pool) ........ ................ + +2008-07-23 08:18 +0000 [r132824] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c: Merged revisions 132823 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 | + oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines + Well, the content of a channel variable may be longer than the + size of a pointer... Thanks, eliel! Reported by: eliel Patches: + chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64) + (closes issue #13135) ........ + +2008-07-22 22:20 +0000 [r132797] Mark Michelson <mmichelson@digium.com> + + * /, channels/chan_sip.c: Merged revisions 132795 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul + 2008) | 11 lines Merged revisions 132777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ Allow + Spiraled INVITEs to work correctly within Asterisk. Prior to this + change, a spiraled INVITE would cause a 482 Loop Detected to be + sent to the caller. With this change, if a potential loop is + detected, the Request-URI is inspected to see if it has changed + from what was originally received. If pedantic mode is on, then + this inspection is fully RFC 3261 compliant. If pedantic mode is + not on, then a string comparison is used to test the equality of + the two R-URIs. This has been tested by using OpenSER to rewrite + the R-URI and send the INVITE back to Asterisk. (closes issue + #7403) Reported by: stephen_dredge Modified: + branches/1.4/channels/chan_sip.c ........ ................ + +2008-07-22 22:15 +0000 [r132793] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132791 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul + 2008) | 2 lines correct fix made in r132777... the code *did* + compile in dev-mode, as long as libpri was installed and enabled + ........ + +2008-07-22 21:59 +0000 [r132782] Olle Johansson <oej@edvina.net> + + * /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged + revisions 132703 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 + lines Merged revisions 132645 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 + lines The most common question on the #asterisk iRC channel and + on mailing lists seems to be in regards to an error message when + retransmit fails. This is frequently misunderstood as a failure + of Asterisk, not a failure of the network to reach the other + party. This document tries to assist the Asterisk user in sorting + out these issues by explaining the logic and pointing at some + possible causes. Hopefully, we will get other questions now :-) + ........ ................ + +2008-07-22 21:55 +0000 [r132780] Tilghman Lesher <tlesher@digium.com> + + * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged + revisions 132778 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008) + | 18 lines Merged revisions 132713 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 + ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500 + (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ + r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) + | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ + ................ ................ + +2008-07-22 21:54 +0000 [r132779] Mark Michelson <mmichelson@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132777 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul + 2008) | 3 lines Get chan_dahdi to compile in devmode ........ + +2008-07-22 21:23 +0000 [r132574-132729] Kevin P. Fleming <kpfleming@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132721 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r132721 | kpfleming | 2008-07-22 16:21:56 -0500 + (Tue, 22 Jul 2008) | 14 lines Merged revisions 132712 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul + 2008) | 6 lines ensure that if any alarms exist at channel + creation time, they are handled identically to if they occurred + later, so that later alarm clearing will work properly and 'make + sense' (closes issue #12160) Reported by: tzafrir ........ + ................ + + * /, configure, configure.ac, acinclude.m4: Merged revisions 132705 + via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ................ r132705 | kpfleming | 2008-07-22 15:54:07 -0500 + (Tue, 22 Jul 2008) | 10 lines Merged revisions 132704 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul + 2008) | 2 lines make AST_C_COMPILE_CHECK able to print a 'pretty' + description of what it is doing ........ ................ + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, + configure, include/asterisk/autoconfig.h.in, configure.ac: Merged + revisions 132643 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul + 2008) | 10 lines Merged revisions 132641 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul + 2008) | 2 lines use renamed libpri API call for controlling this + feature (was improperly named before) ........ ................ + + * channels/chan_dahdi.c, /: Merged revisions 132573 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk + ................ r132573 | kpfleming | 2008-07-21 17:51:16 -0500 + (Mon, 21 Jul 2008) | 10 lines Merged revisions 132571 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 Jul + 2008) | 2 lines teach chan_dahdi how to find the D-channel on BRI + spans, and don't attempt to use channel 24 as a D-channel on + spans of unexpected sizes ........ ................ + +2008-07-21 21:13 +0000 [r132515] Brett Bryant <bbryant@digium.com> + + * configs/features.conf.sample, configs/gtalk.conf.sample, /, + configs/jingle.conf.sample, configs/manager.conf.sample: Merged + revisions 132514 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132514 | + bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines + Update configuration files to add missing options for jingle, + gtalk, manager.conf, and features.conf. (closes issue #13128) + Reported by: caio1982 Patches: missing_options1.diff uploaded by + caio1982 (license 22) ........ + +2008-07-21 21:02 +0000 [r132512-132513] Tilghman Lesher <tlesher@digium.com> + + * main/fskmodem.c (added), /, include/asterisk/fskmodem.h (added): + Merged revisions 132511 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 | + tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines + (Step 2 of 2) ........ + + * main/fskmodem.c (removed), include/asterisk/fskmodem_int.h + (added), build_tools/cflags.xml, main/fskmodem_float.c (added), + /, main/tdd.c, include/asterisk/fskmodem.h (removed), + main/fskmodem_int.c (added), main/callerid.c, + include/asterisk/fskmodem_float.h (added): Merged revisions + 132510 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 | + tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines + Optionally build integer-based routines for FSK tone decoding + (but default to the more accurate float-based routines). (Closes + issue #11679) (Step 1 of 2) ........ + +2008-07-21 20:55 +0000 [r132467-132509] Brett Bryant <bbryant@digium.com> + + * /, apps/app_sendtext.c: Merged revisions 132508 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132508 | + bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines + Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't + supported on a channel (yet _another_ useful patch by eliel). + (closes issue #13081) Reported by: eliel Patches: + app_sendtext.c.patch uploaded by eliel (license 64) Tested by: + eliel ........ + + * /, channels/chan_sip.c: Merged revisions 132468 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 | + bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines + Fix bug where ast_parse_arg would inadvertantly enable sip tcp + when parsing a tcpbindaddr if it was disabled. (closes issue + #13117) Reported by: pj ........ + + * /, channels/chan_iax2.c: Merged revisions 132466 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008) + | 3 lines Fix an issue in iax2 where a call that's been rejected + still kept an open channel on the side that attempted to make the + call (not the side of the call that rejected the call). Changes + were load tested and also approved by Russell. ........ + +2008-07-21 15:34 +0000 [r132426] Jeff Peeler <jpeeler@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132425 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008) + | 2 lines make buffers config option (chan_dahdi.conf) parsing + safer and added logging in case of failure ........ + +2008-07-21 14:48 +0000 [r132389-132391] Russell Bryant <russell@digium.com> + + * apps/app_jack.c, include/asterisk/libresample.h (removed), /, + build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, main/Makefile, main/libresample + (removed), configure.ac, codecs/codec_resample.c, makeopts.in: + Merged revisions 132390 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 | + russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines + Remove libresample from the Asterisk source tree. It is now + available in its own repository, and must be installed like any + other library for Asterisk to use. The two modules that require + it are codec_resample and app_jack. To install libresample: $ svn + co http://svn.digium.com/svn/libresample/trunk libresample $ cd + libresample $ ./configure $ make $ sudo make install This code is + currently in our own repository because the build system did not + include the appropriate targets for building a dynamic library or + for installing the library. ........ + + * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions + 132388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 | + russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines + Enable higher quality resampling, as it doesn't have a noticeable + performance impact on my machine .. ........ + +2008-07-19 16:47 +0000 [r132313] Kevin P. Fleming <kpfleming@digium.com> + + * /, LICENSE: Merged revisions 132312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul + 2008) | 10 lines Merged revisions 132311 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul + 2008) | 2 lines grant a license exception to allow distribution + of Asterisk binaries that use the UW IMAP Toolkit (which is + licensed under a non-GPL-compatible license) ........ + ................ + +2008-07-19 10:47 +0000 [r132278] Michiel van Baak <michiel@vanbaak.info> + + * res/res_config_sqlite.c, /: Merged revisions 132277 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008) + | 7 lines fix a couple of comments in sqlite resource driver. + (closes issue #13110) Reported by: gknispel_proformatique + Patches: res_config_sqlite_comments.patch uploaded by gknispel + (license 261) ........ + +2008-07-18 22:20 +0000 [r132245] Brett Bryant <bbryant@digium.com> + + * main/manager.c, /: Merged revisions 132242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 | + bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines + Fixes problem where manager users loaded from users.conf would be + removed early (before the routine to load the configuration was + finished) because a variable wasn't initialized. ........ + +2008-07-18 20:58 +0000 [r132114-132207] Tilghman Lesher <tlesher@digium.com> + + * /, main/say.c: Merged revisions 132113 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008) + | 14 lines Merged revisions 132112 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008) + | 6 lines Fix for Taiwanese number syntax (closes issue #12319) + Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch + uploaded by CharlesWang (license 444) ........ ................ + +2008-07-18 18:53 +0000 [r132111] Matthew Fredrickson <creslin@digium.com> + + * channels/chan_dahdi.c, /: Merged revisions 132108 via svnmerge + from https://origsvn.digium.com/svn/asterisk/trunk ........ + r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) | + 1 line Make sure we break the poll so that messages queued will + be sent on the SS7 when using CLI commands for blocking and + blocking of CICs and linksets. ........ + +2008-07-18 18:51 +0000 [r132110] Tilghman Lesher <tlesher@digium.com> + + * main/config.c, /: Merged revisions 132109 via svnmerge from + https://origsvn.digium.com/svn/asterisk/trunk ................ + r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008) + | 14 lines Merged revisions 132107 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008) + | 6 lines Textual clarification (closes issue #13106) Reported + by: flefoll Patches: config.c.br14.120173.patch-unknown-directive + uploaded by flefoll (license 244) ........ ................ + +2008-07-18 17:56 +0000 [r132051] Brett Bryant <bbryant@digium.com> + + * main/hashtab.c, /, cdr/cdr_radius.c: Merged revisions 132050 via + svnmerge from https://origsvn.digium.com/svn/asterisk/trunk + ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18 + Jul 2008) | 8 lines Fix ma |