[svn-commits] lmadsen: tag 1.6.0.7-rc2 r183113 - /tags/1.6.0.7-rc2/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Mar 19 11:02:36 CDT 2009
Author: lmadsen
Date: Thu Mar 19 11:02:32 2009
New Revision: 183113
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=183113
Log:
Importing files for 1.6.0.7-rc2 release
Added:
tags/1.6.0.7-rc2/.lastclean (with props)
tags/1.6.0.7-rc2/.version (with props)
tags/1.6.0.7-rc2/ChangeLog (with props)
Added: tags/1.6.0.7-rc2/.lastclean
URL: http://svn.digium.com/svn-view/asterisk/tags/1.6.0.7-rc2/.lastclean?view=auto&rev=183113
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--- tags/1.6.0.7-rc2/ChangeLog (added)
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+2009-03-19 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.6.0.7-rc2
+
+2009-03-19 15:40 +0000 [r183066-183109] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * Release 1.6.0.7-rc1
+
+2009-03-06 17:28 +0000 [r180535] David Vossel <dvossel at 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 at 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 at 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 at 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 at 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
[... 49941 lines stripped ...]
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