[svn-commits] lmadsen: tag 1.6.2.0-beta1 r183182 - /tags/1.6.2.0-beta1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Mar 19 11:31:19 CDT 2009
Author: lmadsen
Date: Thu Mar 19 11:31:13 2009
New Revision: 183182
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=183182
Log:
Importing files for 1.6.2.0-beta1 release
Added:
tags/1.6.2.0-beta1/.lastclean (with props)
tags/1.6.2.0-beta1/.version (with props)
tags/1.6.2.0-beta1/ChangeLog (with props)
Added: tags/1.6.2.0-beta1/.lastclean
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--- tags/1.6.2.0-beta1/ChangeLog (added)
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+2009-03-19 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.6.2.0-beta1
+
+2009-03-19 16:11 +0000 [r183122] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar
+ 2009) | 20 lines Merged revisions 183115 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
+ 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
+ would erroneously report the device as "in use." A user was
+ having an issue where if an outgoing SIP call was canceled, the
+ SIP device would remain in use if we had not received any
+ response to the initial INVITE we sent out. The SIP device would
+ remain in use until the autocongestion timer was exhausted. I
+ tracked down the cause of this to be the section of code I am
+ removing here. I asked several people what the purpose of this
+ code was meant to be, but no one could give me any sort of answer
+ as to why this was here. The person who was having this issue has
+ been using this patch for several months and it has stopped the
+ problems they have had. AST-196 ........ ................
+
+2009-03-19 15:45 +0000 [r183068-183111] Joshua Colp <jcolp 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 [r183031] 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 14:39 +0000 [r182947] 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 20:53 +0000 [r182725] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /,
+ channels/h323/ast_h323.cxx, configure,
+ autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h,
+ channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions
+ 182722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 |
+ jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
+ Allow H.323 Plus library to be used in addition to the OpenH323
+ library Chan_h323 can now be compiled against both the previously
+ supported versions of OpenH323 as well as the current H.323 Plus
+ (version 1.20.2). The configure script has been modified to look
+ in the default install location of h323 to hopefully help avoid
+ using the environment variables OPENH323DIR and PWLIBDIR. Also,
+ the CLI command "h323 show version" has been added which
+ indicates which version of h323 is in use. (closes issue #11261)
+ Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
+ uploaded by jthurman (license 614) ........
+
+2009-03-17 16:46 +0000 [r182592] Russell Bryant <russell 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 15:01 +0000 [r182528-182534] 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 05:54 +0000 [r182453] 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 ........ ................
+
+2009-03-17 02:02 +0000 [r182409] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 182408 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009)
+ | 8 lines OPENR2 uses an incorrect string value if the extension
+ delimiter is not present. * Fixed OPENR2 using an incorrect
+ string value if the extension delimiter is not present in the
+ Dial() function. This was fixed for SS7 and PRI in trunk
+ -r172400. * Made OPENR2 stripmsd behavior the same as the SS7,
+ PRI, and others. * Removed trailing whitespace that appeared with
+ OPENR2. ........
+
+2009-03-16 20:51 +0000 [r182360-182361] Russell Bryant <russell at digium.com>
+
+ * /: svnmerge init
+
+ * / (added): Create a branch for 1.6.2
+
+2009-03-16 20:35 +0000 [r182355] Russell Bryant <russell at digium.com>
+
+ * CREDITS, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ CHANGES, makeopts.in: Add MFC/R2 support for chan_dahdi. This
+ commit introduces official support for R2 signaling in
+ chan_dahdi. The modifications to chan_dahdi, and the supporting
+ library, LibOpenR2, were both written by Moises Silva. Many users
+ are using this code, or a variant of it, in Asterisk 1.2, 1.4 and
+ 1.6 in Brazil, México and Argentina. An unknown number of users
+ (but at least 1) are using it in each of the following countries:
+ Colombia, Nepal, Thailand, Venezuela, Perú, and probably others.
+ To use this code, LibOpenR2 must be installed from
+ http://www.libopenr2.org/. Information about configuration can be
+ found in configs/chan_dahdi.conf.sample. The code committed is
+ the most up to date version, which was being maintained in
+ svn/asterisk/team/moy/mfcr2/. I would also like to include a
+ Thank You to the many others that tested this code beyond those
+ listed in this commit message. These are the names that I could
+ find in the mantis issue. (closes issue #12509) Reported by: moy
+ Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested
+ by: moy, korihor, viniciusfontes, Skarmeth, loloski,
+ asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare,
+ ecarruda, rtorresduque, PTorres, ychen Review:
+ http://reviewboard.digium.com/r/40/
+
+2009-03-16 17:49 +0000 [r182282] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 182281 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16
+ Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32
+ byte random padding at the beginning of an encrypted IAX2 frame
+ turns out to not be all that random at all. This patch calls
+ ast_random to fill the padding buffer with random data. The
+ padding is randomized at the beginning of every encrypted call
+ and for every encrypted retransmit frame. Review:
+ http://reviewboard.digium.com/r/193/ ........
+
+2009-03-16 17:33 +0000 [r182211-182278] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_env.c: Fix an off-by-one error in the FILE() function,
+ and extend FILE()'s length parameter to work like variable
+ substitution. Previously, FILE() returned one less character than
+ specified, due to the terminating NULL. Both the offset and
+ length parameters now behave identically to the way variable
+ substitution offsets and lengths also work. (closes issue #14670)
+ Reported by: BMC
+
+ * channels/chan_local.c, /: Merged revisions 182208 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16
+ Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak
+ of a local pvt structure. (closes issue #14656) Reported by:
+ caspy Patches: 20090313__bug14656__2.diff.txt uploaded by
+ tilghman (license 14) Tested by: caspy ........
+
+2009-03-16 13:58 +0000 [r182171] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c: Fix a memory leak in the ast_answer /
+ __ast_answer API call. For a channel that is not yet answered
+ this API call will wait until a voice frame is received on the
+ channel before returning. It does this by waiting for frames on
+ the channel and reading them in. The frames read in were not
+ freed when they should have been.
+
+2009-03-13 21:26 +0000 [r182029-182121] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Change faulty comparison used when announcing
+ average hold minutes and seconds (closes issue #14227) Reported
+ by: caspy
+
+ * main/features.c: Remove ast_ prefix from functions which are not
+ public.
+
+ * /, main/features.c: Merged revisions 181990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar
+ 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and
+ peer when interpreting DTMF. Dynamic features defined in the
+ applicationmap section of features.conf allow one to specify
+ whether the caller, callee, or both have the ability to use the
+ feature. The documentation in the features.conf.sample file could
+ be interpreted to mean that one only needs to set the
+ DYNAMIC_FEATURES channel variable on the calling channel in order
+ to allow for the callee to be able to use the features which he
+ should have permission to use. However, the DYNAMIC_FEATURES
+ variable would only be read from the channel of the participant
+ that pressed the DTMF sequence to activate the feature. The
+ result of this was that the callee was unable to use dynamic
+ features unless the dialplan writer had taken measures to be sure
+ that the DYNAMIC_FEATURES variable was set on the callee's
+ channel. This commit changes the behavior of
+ ast_feature_interpret to concatenate the values of
+ DYNAMIC_FEATURES from both parties involved in the bridge. The
+ features themselves determine who has permission to use them, so
+ there is no reason to believe that one side of the bridge could
+ gain the ability to perform an action that they should not have
+ the ability to perform. Kevin Fleming pointed out on the
+ asterisk-users list that the typical way that this was worked
+ around in the past was by setting _DYNAMIC_FEATURES on the
+ calling channel so that the value would be inherited by the
+ called channel. While this works, the documentation alone is not
+ enough to figure out why this is necessary for the callee to be
+ able to use dynamic features. In this particular case, changing
+ the code to match the documentation is safe, easy, and will
+ generally make things easier for people for future installations.
+ This bug was originally reported on the asterisk-users list by
+ David Ruggles. (closes issue #14657) Reported by: mmichelson
+ Patches: 14657.patch uploaded by mmichelson (license 60) ........
+
+2009-03-13 17:25 +0000 [r182022] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix an issue with requesting a T38 reinvite
+ before the call is answered. The code responsible for sending the
+ T38 reinvite did not check if an INVITE was already being
+ handled. This caused things to get confused and the call to fail.
+ The code now defers sending the T38 reinvite until the current
+ INVITE is done being handled. (issue AST-191)
+
+2009-03-13 16:55 +0000 [r181985] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: improve a bit of suboptimal code
+
+2009-03-13 01:26 +0000 [r181899] Richard Mudgett <rmudgett at digium.com>
+
+ * /: Merged revisions 181898 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 Just
+ recording the v1.4 change in trunk since it originally came from
+ here. ........ r181898 | rmudgett | 2009-03-12 20:19:29 -0500
+ (Thu, 12 Mar 2009) | 4 lines Use the correct branch integrated
+ property when generating the version string. Copied the
+ make_version file from Asterisk trunk. ........
+
+2009-03-12 21:43 +0000 [r181769-181846] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Run the macro on the queue member's channel
+ when he answers, not the caller's channel.
+
+ * /, channels/chan_sip.c: Merged revisions 181768 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar
+ 2009) | 22 lines Properly send a 487 on an INVITE we have not
+ responded to if we receive a BYE. If we receive an INVITE from an
+ endpoint and then later receive a BYE from that same endpoint
+ before we have sent a final response for the INVITE, then we need
+ to respond to the INVITE with a 487. There was logic in the code
+ prior to this commit which seemed to exist solely to handle this
+ situation, but there was one condition in an if statement which
+ was incorrect. The only way we would send a 487 was if the
+ sip_pvt had no owner channel. This made no sense since we created
+ the owner channel when we received the INVITE, meaning that the
+ majority of the time we would never send the 487. The 487 being
+ sent should not rely on whether we have created a channel. Its
+ delivery should be dependent on the current state of the initial
+ INVITE transaction. With this commit, that logic is now correctly
+ in place. (closes issue #14149) Reported by: legranjl Patches:
+ 14149.patch uploaded by mmichelson (license 60) Tested by:
+ legranjl ........
+
+2009-03-12 17:32 +0000 [r181731] Tilghman Lesher <tlesher at digium.com>
+
+ * main/translate.c: Adjust translation table column widths based
+ upon the translation times. Previously, only 5 columns were
+ displayed, and if a translation time exceeded 99,999 useconds, it
+ would be displayed as 0, instead of its actual time. (closes
+ issue #14532) Reported by: pj Patches:
+ 20090311__bug14532.diff.txt uploaded by tilghman (license 14)
+ Tested by: pj
+
+2009-03-12 16:56 +0000 [r181612-181665] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 181664 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar
+ 2009) | 2 lines Fix incorrect usage of strncasecmp... I really
+ meant to use strcasecmp. ........
+
+ * /, res/res_musiconhold.c: Merged revisions 181659-181660 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8
+ lines Fix another scenario where depending on configuration the
+ stream would not get read. For custom commands we don't know
+ whether the audio is coming from a stream or not so we are going
+ to have to read the data despite no channels. (closes issue
+ #14416) Reported by: caspy ........ r181660 | file | 2009-03-12
+ 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in
+ previous commit. ........
+
+ * /, res/res_musiconhold.c: Merged revisions 181655 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar
+ 2009) | 10 lines Fix issue with streaming MOH failing if nobody
+ is listening. When a music class is setup to actually provide
+ music on hold from a stream we need to constantly read audio from
+ it since it will constantly be providing audio. This is now done
+ despite there being no channels listening to it. (closes issue
+ #14416) Reported by: caspy ........
+
+ * apps/app_dial.c: Fix crash when sleep and retries argument was
+ not given to RetryDial application. (closes issue #14647)
+ Reported by: sherpya
+
+2009-03-12 01:33 +0000 [r181542-181577] Richard Mudgett <rmudgett at digium.com>
+
+ * build_tools/make_version: Whitespace chages.
+
+ * build_tools/make_version: Use the correct branch integrated
+ property when generating the version string
+
+2009-03-11 23:14 +0000 [r181499] Michiel van Baak <michiel at vanbaak.info>
+
+ * configs/sip.conf.sample: Provide correct hint to debug SIP
+ trouble in the default config (closes issue #14646) Reported by:
+ strk
+
+2009-03-11 22:25 +0000 [r181465] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Make handling of the BRIDGE_PLAY_SOUND variable
+ thread-safe.
+
+2009-03-11 22:20 +0000 [r181444] Jason Parker <jparker at digium.com>
+
+ * /, configure, configure.ac: Merged revisions 181436 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar
+ 2009) | 4 lines Allow prefix to set localstatedir (when used and
+ different from the default). This is similar to the /etc change
+ that was made for the non-FreeBSD case. ........
+
+2009-03-11 22:14 +0000 [r181424-181428] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Make handling of the BRIDGEPVTCALLID variable
+ thread-safe.
+
+ * main/channel.c, /: Merged revisions 181423 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009)
+ | 9 lines Make code that updates BRIDGEPEER variable thread-safe.
+ It is not safe to read the name field of an ast_channel without
+ the channel locked. This patch fixes some places in channel.c
+ where this was being done, and lead to crashes related to
+ masquerades. (closes issue #14623) Reported by: guillecabeza
+ ........
+
+2009-03-11 17:34 +0000 [r181371] David Vossel <dvossel at digium.com>
+
+ * channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions
+ 181340 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009)
+ | 11 lines encrypted IAX2 during packet loss causes decryption to
+ fail on retransmitted frames If an iax channel is encrypted, and
+ a retransmit frame is sent, that packet's iseqno is updated while
+ it is encrypted. This causes the entire frame to be corrupted.
+ When the corrupted frame is sent, the other side decrypts it and
+ sends a VNAK back because the decrypted frame doesn't make any
+ sense. When we get the VNAK, we look through the sent queue and
+ send the same corrupted frame causing a loop. To fix this,
+ encrypted frames requiring retransmission are decrypted, updated,
+ then re-encrypted. Since key-rotation may change the key held by
+ the pvt struct, the keys used for encryption/decryption are held
+ within the iax_frame to guarantee they remain correct. (closes
+ issue #14607) Reported by: stevenla Tested by: dvossel Review:
+ http://reviewboard.digium.com/r/192/ ........
+
+2009-03-11 17:26 +0000 [r181345] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181328 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) |
+ 14 lines Fix issue where an attended transfer could not be
+ completed under a rare scenario. When completing an attended
+ transfer chan_sip does a check to make sure the extension in the
+ URI portion of the Refer-To header is a local valid extension. We
+ don't actually need to check this since we know for sure the
+ other channel is already up and talking to the extension. Some
+ devices do not put the extension in the Refer-To header either,
+ which can cause the extension check to fail. We now no longer do
+ this check if it is an attended transfer. (closes issue #14628)
+ Reported by: sverre Patches: 14628.diff uploaded by file (license
+ 11) ........
+
+2009-03-11 17:04 +0000 [r181301] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/astobj2.h: Turn off malloc debugging of astobj2,
+ since it apparently doesn't work too well during startup.
+
+2009-03-11 16:40 +0000 [r181296] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 181295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9
+ lines Fix a problem with inband DTMF detection on outgoing SIP
+ calls when dtmfmode=auto. When dtmfmode was set to auto the
+ inband DTMF detector was not setup on outgoing SIP calls. This
+ caused inband DTMF detection to fail. The inband DTMF detector is
+ now setup for both dtmfmode inband and auto. (closes issue
+ #13713) Reported by: makoto ........
+
+2009-03-11 16:19 +0000 [r181292] Russell Bryant <russell at digium.com>
+
+ * doc/google-soc2009-ideas.txt: Replace contents of this doc with a
+ pointer to its new home
+
+2009-03-11 14:28 +0000 [r181244] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fix segfault when dialing a typo'd queue If
+ trying to dial a non-existent queue, there would be a segfault
+ when attempting to access q->weight, even though q was NULL. This
+ problem was introduced during the queue-reset merge and thus only
+ affects trunk. (closes issue #14643) Reported by: alecdavis
+
+2009-03-11 13:44 +0000 [r181210] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_confbridge.c: Don't play the "you are about to be placed
+ into the conference" and "the leader has left the conference"
+ sounds if the quiet option is enabled. (reported by Vadim Lebedev
+ on the asterisk-dev list)
+
+2009-03-11 04:06 +0000 [r181135] Jeff Peeler <jpeeler at digium.com>
+
+ * utils/Makefile, include/asterisk/utils.h,
+ include/asterisk/astmm.h, channels/chan_sip.c,
+ channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c,
+ pbx/pbx_config.c: Fix malloc debug macros to work properly with
+ h323. The main problem here was that cstdlib was undefining free
+ thereby causing the proper debug macros to not be used.
+ ast_h323.cxx has been changed to call ast_free instead to avoid
+ the issue. A few other issues were addressed: - There were a few
+ instances of functions improperly passing ast_free instead of
+ ast_free_ptr. - Some clean up was done to avoid the debug macros
+ intentionally being redefined. (copied below from Kevin's commit,
+ appreciate the help) - disable astmm.h from doing anything when
+ STANDALONE is defined, which is used by the tools in the utils/
+ directory that use parts of Asterisk header files in hackish
+ ways; also ensure that utils/extconf.c and utils/conf2ael.c are
+ compiled with STANDALONE defined. (closes issue #13593) Reported
+ by: pj
+
+2009-03-11 02:25 +0000 [r181099] Russell Bryant <russell at digium.com>
+
+ * doc/google-soc2009-ideas.txt: tabs to spaces
+
+2009-03-11 00:49 +0000 [r181032-181033] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Add missing comment that quotes RFC 3891
+
+ * /, channels/chan_sip.c: Merged revisions 181029,181031 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar
+ 2009) | 9 lines Fix incorrect tag checking on transfers when
+ pedantic=yes is enabled. (closes issue #14611) Reported by:
+ klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt
+ uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
+ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar
+ 2009) | 3 lines Remove unused variables. ........
+
+2009-03-11 00:29 +0000 [r181027-181028] Tilghman Lesher <tlesher at digium.com>
+
+ * main/strings.c, main/hashtab.c, include/asterisk/astobj2.h,
+ main/heap.c, include/asterisk/strings.h,
+ include/asterisk/hashtab.h, main/astobj2.c,
+ include/asterisk/heap.h: Add MALLOC_DEBUG to various utility
+ APIs, so that memory leaks can be tracked back to their source.
+ (related to issue #14636)
+
+ * main/pbx.c: Spacing changes only
+
+2009-03-10 22:03 +0000 [r180944] Jason Parker <jparker at digium.com>
+
+ * /, configure, configure.ac, autoconf/ast_prog_sed.m4,
+ autoconf/ast_check_gnu_make.m4: Merged revisions 180941 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) |
+ 1 line Make things happier when using autoconf 2.62+ ........
+
+2009-03-10 22:03 +0000 [r180935-180942] Russell Bryant <russell at digium.com>
+
+ * doc/google-soc2009-ideas.txt: Add some notes on getting in
+ contact with the dev community
+
+ * doc/google-soc2009-ideas.txt: Remove difficulty and language
+ specifiers
+
+ * doc/google-soc2009-ideas.txt: Expand upon documentation of
+ manager event project
+
+2009-03-10 21:15 +0000 [r180898] Michiel van Baak <michiel at vanbaak.info>
+
+ * CHANGES: list the move of the astvarrundir from /var/run to
+ /var/run/asterisk (actually its $(localstatedir)/run/asterisk
+ Makes setups with asterisk as non-root easier to manage because
+ you can setup permissions on this dir instead of touching a file
+ and setting permissions on that. Files that come to mind are
+ asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell
+
+2009-03-10 19:36 +0000 [r180859-180862] Russell Bryant <russell at digium.com>
+
+ * doc/google-soc2009-ideas.txt: add more projects
+
+ * doc/google-soc2009-ideas.txt: add more project ideas
+
+2009-03-10 14:40 +0000 [r180800] Joshua Colp <jcolp at digium.com>
+
+ * main/manager.c: Reset the thread local string buffer when
+ handling the UserEvent action. (closes issue #14593) Reported by:
+ JimDickenson
+
+2009-03-09 22:00 +0000 [r180750] Russell Bryant <russell at digium.com>
+
+ * doc/google-soc2009-ideas.txt: Add current mentors list, and first
+ pass on a project list broken out of "PineMango" I will work on
+ adding projects that have been sent to be via email tomorrow.
+
+2009-03-09 20:58 +0000 [r180719] Jeff Peeler <jpeeler at digium.com>
+
+ * include/asterisk/rtp.h, include/asterisk/extconf.h,
+ main/devicestate.c, include/asterisk/tcptls.h, main/enum.c,
+ include/asterisk/callerid.h, include/asterisk/doxyref.h,
+ include/asterisk/event.h, include/asterisk/audiohook.h,
+ include/asterisk/dsp.h, include/asterisk/timing.h,
+ include/asterisk/udptl.h, include/asterisk/dlinkedlists.h,
+ include/asterisk/utils.h, include/asterisk/devicestate.h,
+ include/asterisk/taskprocessor.h, include/asterisk/enum.h,
+ include/asterisk/astobj2.h, include/asterisk/config.h,
+ include/asterisk/channel.h, include/asterisk/manager.h,
+ include/asterisk/heap.h, include/asterisk/logger.h,
+ include/asterisk/http.h, include/asterisk/res_odbc.h,
+ include/asterisk/app.h, main/tcptls.c,
+ include/asterisk/linkedlists.h, include/asterisk/sched.h,
+ include/asterisk/datastore.h, include/asterisk/lock.h,
+ include/asterisk/pbx.h, include/asterisk/dnsmgr.h: Add Doxygen
+ documentation for API changes from 1.6.0 to 1.6.1 Copied from my
+ review board description: This is a continuation of the API
+ changes documentation started for describing changes between
+ releases. Most of the API changes were pretty simple needing only
+ to be brought to attention via the new "Asterisk API Changes"
+ list. However, if you see anything that needs further explanation
+ feel free to supplement what is there. The current method of
+ documenting is to add (in the header file): \version <ver number>
+ <description of changes> and then to add the function to the
+ change list in doxyref.h on the AstAPIChanges page. I also made
+ sure all the functions that were newly added were tagged with
+ \since 1.6.1. I think this is a good habit to start both for the
+ historical aspect as well as for the future ability to easily add
+ a "New Asterisk API" page. Review:
+ http://reviewboard.digium.com/r/190/
+
+2009-03-09 14:14 +0000 [r180684] Russell Bryant <russell at digium.com>
+
+ * doc/google-soc2009-ideas.txt (added): Add skeleton for GSoC ideas
+ list
+
+2009-03-07 15:36 +0000 [r180641] Russell Bryant <russell at digium.com>
+
+ * contrib/asterisk-ng-doxygen: Make some minor updates to the
+ doxygen configuration - add bridges directory to be processed -
+ add some res/ subdirs - alphabetize subdirs - use consistent
+ indentation
+
+2009-03-06 18:25 +0000 [r180579] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 180567 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri,
+ 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when
+ IMAP storage is enabled. ........
+
+2009-03-06 17:26 +0000 [r180534] David Vossel <dvossel at digium.com>
+
+ * /, main/enum.c: Merged revisions 180532 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009)
+ | 9 lines Fix handling of backreferences for ENUM lookups enum.c
+ did not handle regex backtraces correctly. The '\1' in the regex
+ is a backreference that requires a pattern match to be inserted.
+ The way the code used to work is that it would find the
+ backreference and insert the entire input string minus the '+'.
+ This is incorrect. The regexec() function takes in a variable
+ called pmatch which is an array of structs containing the start
+ and end indexes for each backreference substring. The original
+ code actually passed the pmatch array pointer into regexec but
+ never did anything with it. Now when a backtrace is found, the
+ backtrace number is looked up in the pmatch array and the correct
+ substring is inserted. (closes issue #14576) Reported by:
+ chris-mac Review: http://reviewboard.digium.com/r/187/ ........
+
+2009-03-05 23:26 +0000 [r180383-180465] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 180464 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu,
+ 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when
+ identical mailbox names were defined in separate contexts. There
+ was a fix put in a while back so that an X-Asterisk-VM-Context
+ message header was added to stored IMAP voicemails. This would
+ allow for us to differentiate if the same mailbox name was used
+ in multiple contexts. The problem still left was that not all
+ places where messages were retrieved actually attempted to use
+ this header for information when retrieving messages. This commit
+ fixes that so that MWI and message retrieval from VoiceMailMain
+ work as expected. (closes issue #13853) Reported by: vicks1
+ Patches: 13853_v2.patch uploaded by mmichelson (license 60)
+ Tested by: lmadsen ........
+
+ * /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged
+ revisions 180380 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar
+ 2009) | 25 lines Fix broken mailbox parsing when searchcontexts
+ option is enabled. When using the searchcontexts option in
+ voicemail.conf, the code made the assumption that all mailbox
+ names defined were unique across all contexts. However, the code
+ did nothing to actually enforce this assumption, nor did it do
+ anything to alert a user that he may have created an ambiguity in
+ his voicemail.conf file by defining the same mailbox name in
[... 10671 lines stripped ...]
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