[svn-commits] lmadsen: tag 1.6.0.18-rc1 r228505 - /tags/1.6.0.18-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Nov 6 11:55:31 CST 2009
Author: lmadsen
Date: Fri Nov 6 11:55:26 2009
New Revision: 228505
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=228505
Log:
Importing files for 1.6.0.18-rc1 release.
Added:
tags/1.6.0.18-rc1/.lastclean (with props)
tags/1.6.0.18-rc1/.version (with props)
tags/1.6.0.18-rc1/ChangeLog (with props)
Added: tags/1.6.0.18-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.0.18-rc1/.lastclean?view=auto&rev=228505
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--- tags/1.6.0.18-rc1/ChangeLog (added)
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+2009-11-06 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.6.0.18-rc1
+
+2009-11-06 17:53 +0000 [r228479-228500] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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
[... 54146 lines stripped ...]
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