[svn-commits] lmadsen: tag 1.4.27-rc3 r228494 - /tags/1.4.27-rc3/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Nov 6 11:41:54 CST 2009
Author: lmadsen
Date: Fri Nov 6 11:41:50 2009
New Revision: 228494
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=228494
Log:
Importing files for 1.4.27-rc3 release.
Added:
tags/1.4.27-rc3/.lastclean (with props)
tags/1.4.27-rc3/.version (with props)
tags/1.4.27-rc3/ChangeLog (with props)
Added: tags/1.4.27-rc3/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.4.27-rc3/.lastclean?view=auto&rev=228494
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--- tags/1.4.27-rc3/ChangeLog (added)
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+2009-11-06 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.27-rc3
+
+2009-11-06 17:07 +0000 [r228418] David Vossel <dvossel at digium.com>
+
+ * codecs/codec_ilbc.c: 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 16:41 +0000 [r228409] Joshua Colp <jcolp at digium.com>
+
+ * main/abstract_jb.c: 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 16:26 +0000 [r228378] Matthew Nicholson <mnicholson at digium.com>
+
+ * funcs/func_base64.c, main/utils.c: Properly handle '=' while
+ decoding base64 messages and null terminate strings returned from
+ BASE64_DECODE. (closes issue #15271) Reported by: chappell
+ Patches: base64_fix.patch uploaded by chappell (license 8) Tested
+ by: kobaz
+
+2009-11-06 15:41 +0000 [r228272-228338] David Vossel <dvossel at digium.com>
+
+ * main/astfd.c: fixes crash in astfd.c (closes issue #15981)
+ Reported by: slavon
+
+ * funcs/func_audiohookinherit.c: fixes memory leak in
+ func_audiohookinherit.c (closes issue 0015394) Reported by:
+ boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks
+ (license 790) Tested by: dbrooks, boroda
+
+2009-11-05 19:14 +0000 [r228079] Jason Parker <jparker at digium.com>
+
+ * channels/chan_vpb.cc: Fix crash on VPB exception when no hardware
+ is present. (closes issue #14970) Reported by: tzafrir Patches:
+ vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+ markwaters
+
+2009-11-05 18:59 +0000 [r228078] David Brooks <dbrooks at digium.com>
+
+ * channels/chan_misdn.c: 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
+
+2009-11-04 23:47 +0000 [r227944] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c: 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 20:52 +0000 [r227758-227827] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_dial.c: 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 Tested by: frawd, mnicholson,
+ file Review: https://reviewboard.asterisk.org/r/385/
+
+2009-11-04 19:25 +0000 [r227700-227735] Joshua Colp <jcolp at digium.com>
+
+ * static-http/prototype.js: 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: 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 17:55 +0000 [r227275] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: 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 15:36 +0000 [r227166] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: 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 10:48 +0000 [r227088-227090] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Fixing bug before someone reports it...
+
+ * channels/chan_sip.c: Adding IP address in Contact ACL log message
+ and removing redundant message (based on kpfleming's feedback)
+
+ * channels/chan_sip.c: Use proper response code when violating
+ Contact ACL's. Review: https://reviewboard.asterisk.org/r/415/
+ Thanks kpfleming for a quick review. (EDVX-003)
+
+2009-11-02 20:52 +0000 [r226972] 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/
+
+2009-11-02 18:08 +0000 [r226889] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: 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 17:14 +0000 [r226811] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: 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
+
+2009-11-02 15:31 +0000 [r226688-226736] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: fixes crash on iterator_destroy on
+ uninitialized iterator (closes issue #16162) Reported by: krn
+
+ * channels/chan_iax2.c: changes calltoken debug messages from
+ LOG_NOTICE to LOG_DEBUG like they are supposed to be (closes
+ issue #16144) Reported by: aragon
+
+2009-10-29 18:11 +0000 [r226531] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_local.c, doc/localchannel.txt: 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 20:06 +0000 [r226377-226382] Leif Madsen <lmadsen at digium.com>
+
+ * configs/sip.conf.sample: 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/channelvariables.txt: 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 18:02 +0000 [r226138-226304] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/linkedlists.h: Fix documentation (pointed out by
+ TheDavidFactor on #-dev)
+
+ * main/manager.c: 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 22:13 +0000 [r225957] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * configure, include/asterisk/autoconfig.h.in, 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
+
+2009-10-23 14:00 +0000 [r225581] Kevin P. Fleming <kpfleming at digium.com>
+
+ * Makefile: 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 21:51 +0000 [r225484] Leif Madsen <lmadsen at digium.com>
+
+ * doc/valgrind.txt, contrib/valgrind.supp (added): 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
+
+2009-10-21 20:58 +0000 [r225243] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: 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/
+
+2009-10-21 16:44 +0000 [r225169-225171] Russell Bryant <russell at digium.com>
+
+ * main/translate.c: Revert 225169, as this doesn't account for the
+ possibility of a list of frames.
+
+ * main/translate.c: Isolate the frame returned from
+ ast_translate().
+
+2009-10-21 16:02 +0000 [r225103-225105] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c, apps/app_meetme.c, include/asterisk/channel.h: Fix
+ documentation for ast_softhangup() and correct the misuse
+ thereof. (closes issue #16103) Reported by: majorbloodnok
+
+ * apps/app_voicemail.c: Suffix is not needed for a match
+
+2009-10-21 14:37 +0000 [r225032] David Vossel <dvossel at digium.com>
+
+ * configs/iax.conf.sample, channels/chan_sip.c,
+ configs/sip.conf.sample, channels/chan_iax2.c: 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 02:59 +0000 [r224931] 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: 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 22:07 +0000 [r224855] Tilghman Lesher <tlesher at digium.com>
+
+ * main/audiohook.c: 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 17:46 +0000 [r224773] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: Add support for relaying early media in the
+ features attended transfer option. (closes issue #14828) Reported
+ by: licedey
+
+2009-10-19 23:44 +0000 [r224670] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/rtp.c: 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 19:47 +0000 [r224565] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: Do not attempt early media bridging (ie: direct
+ RTP setup) if options are enabled that should prevent it. (closes
+ issue #14763) Reported by: cupotka
+
+2009-10-17 01:32 +0000 [r224330] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: 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 20:25 +0000 [r224260] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: 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.)
+
+2009-10-13 20:58 +0000 [r223955] Jean Galarneau <jgalarneau at digium.com>
+
+ * channels/chan_dahdi.c: Fix PRI timer T309 operation
+
+2009-10-12 23:12 +0000 [r223804] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_dial.c: 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 15:30 +0000 [r223692] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: 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. (issue #16025)
+ Reported by: jamicque
+
+2009-10-11 18:34 +0000 [r223485-223550] Russell Bryant <russell at digium.com>
+
+ * apps/app_queue.c: Remove a duplicate ao2_iterator_destroy().
+
+ * main/autoservice.c: Remove some unnecessary code.
+
+ * main/autoservice.c: 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.
+
+2009-10-09 18:20 +0000 [r223225] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c: Signal timeouts by returning AST_CONTROL_RINGING
+ when originating calls. (closes issue #15104) Reported by:
+ nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
+ (license 96) Tested by: nblasgen, mnicholson
+
+2009-10-09 18:17 +0000 [r223213] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_dial.c: Fix potential memory leak in app_dial.c
+
+2009-10-09 17:52 +0000 [r223142-223205] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: fixes sip registration using authuser in
+ user.conf (closes issue #14954) Reported by: tornblad Tested by:
+ mmichelson, tornblad, dvossel
+
+ * channels/chan_sip.c: 'auth=' did not parse md5 secret correctly
+ (closes issue https://issues.asterisk.org/view.php?id=15949)
+ Reported by: ebroad Patches: authparsefix.patch uploaded by
+ ebroad (license 878) 15949_trunk.diff uploaded by dvossel
+ (license 671) Tested by: ebroad
+
+2009-10-08 19:45 +0000 [r222878] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/file.h, main/frame.c, main/file.c,
+ include/asterisk/frame.h: Make filestream frame handling safer by
+ isolating frames before returning them. This patch is related to
+ a number of issues on the bug tracker that show crashes related
+ to freeing frames that came from a filestream. A number of fixes
+ have been made over time while trying to figure out these
+ problems, but there re still people seeing the crash. (Note that
+ some of these bug reports include information about other
+ problems. I am specifically addressing the filestream frame crash
+ here.) I'm still not clear on what the exact problem is. However,
+ what is _very_ clear is that we have seen quite a few problems
+ over time related to unexpected behavior when we try to use
+ embedded frames as an optimization. In some cases, this
+ optimization doesn't really provide much due to improvements made
+ in other areas. In this case, the patch modifies filestream
+ handling such that the embedded frame will not be returned.
+ ast_frisolate() is used to ensure that we end up with a
+ completely mallocd frame. In reality, though, we will not
+ actually have to malloc every time. For filestreams, the frame
+ will almost always be allocated and freed in the same thread.
+ That means that the thread local frame cache will be used. So,
+ going this route doesn't hurt. With this patch in place, some
+ people have reported success in not seeing the crash anymore.
+ (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon
+ Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell
+ (license 2) Tested by: aragon, russell (closes issue #15817)
+ Reported by: zerohalo Tested by: zerohalo (closes issue #15845)
+ Reported by: marhbere Review:
+ https://reviewboard.asterisk.org/r/386/
+
+2009-10-08 19:45 +0000 [r222877] David Vossel <dvossel at digium.com>
+
+ * main/netsock.c, include/asterisk/netsock.h: fixes an
+ ast_netsock_list memory leak. ABE-1998 Review:
+ https://reviewboard.asterisk.org/r/395/
+
+2009-10-08 16:33 +0000 [r222691-222797] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn_config.c: Fix memory leak if chan_misdn config
+ parameter is repeated. Memory leak when the same config option is
+ set more than once in an misdn.conf section. Why must this be
+ considered? Templates! Defining a template with default port
+ options and later adding to or overriding some of them. Patches:
+ memleak-misdn.patch JIRA ABE-1998
+
+ * channels/chan_misdn.c: chan_misdn.c:process_ast_dsp() memory leak
+ misdn.conf: astdtmf must be set to "yes". With "no", buffer loss
+ does not occur. The translated frame "f2" when passing through
+ ast_dsp_process() is not freed whenever it is not used further in
+ process_ast_dsp(). Then in the end it is never ever freed.
+ Patches: translate.patch JIRA ABE-1993
+
+2009-10-07 17:41 +0000 [r222542] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: crash on transfer handle_invite_replaces()
+ attempts to uplock a pvt's owner channel without first verifing
+ that it exists. (issue #16027)
+
+2009-10-06 23:51 +0000 [r222393-222462] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Add missing unlock(s) in dahdi_read (two
+ cases in trunk) (closes issue #15683) Reported by: alecdavis
+
+ * channels/chan_dahdi.c: Fix potential crash when entire span
+ request is received. The variable index used in this scenario for
+ accessing the dahdi_pvts was wrong and was most likely copied
+ from the several other places it is used correctly. (closes issue
+ #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch
+ uploaded by tsearle (license 373)
+
+2009-10-06 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.27-rc2
+
+2009-10-06 01:16 +0000 [r222152] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h,
+ res/res_musiconhold.c, apps/app_queue.c, channels/chan_iax2.c:
+ Fix ao2_iterator API to hold references to containers being
+ iterated. See Mantis issue for details of what prompted this
+ change. Additional notes: This patch changes the ao2_iterator API
+ in two ways: F_AO2I_DONTLOCK has become an enum instead of a
+ macro, with a name that fits our naming policy; also, it is now
+ necessary to call ao2_iterator_destroy() on any iterator that has
+ been created. Currently this only releases the reference to the
+ container being iterated, but in the future this could also
+ release other resources used by the iterator, if the iterator
+ implementation changes to use additional resources. (closes issue
+ #15987) Reported by: kpfleming Review:
+ https://reviewboard.asterisk.org/r/383/
+
+2009-10-02 17:32 +0000 [r222026] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Removes unnecessary unlock, clarifies a
+ memcpy.
+
+2009-10-02 16:58 +0000 [r221776-221970] Tilghman Lesher <tlesher at digium.com>
+
+ * main/astobj2.c: Ensure the result of the hash function is
+ positive. Negative array offsets suck.
+
+ * main/asterisk.c, main/rtp.c, main/say.c: Fix a bunch of
+ off-by-one errors
+
+2009-10-01 23:18 +0000 [r221769] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h:
+ Occasionally losing use of B channels in chan_misdn. I have not
+ been able to reproduce the problem of losing channels. However, I
+ have seen in the code a reentrancy problem that might give these
+ symptoms. The reentrancy patch does several things: 1) Guards B
+ channel and B channel structure allocation. 2) Makes the B
+ channel structure find routines more precise in locating records.
+ 3) Never leave a B channel allocated if we received cause 44. The
+ last item may cause temporary outgoing call problems, but they
+ should clear when the line becomes idle. (closes issue #15490)
+ Reported by: slutec18 Patches:
+ issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
+ Reported by: FabienToune Patches:
+ issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
+ (license 664) Tested by: FabienToune, rmudgett, slutec18
+
+2009-10-01 15:24 +0000 [r221360-221588] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Use unsigned ints for portinuri flags.
+
+ * channels/chan_sip.c: Make portinuri a bitfield.
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Fix SRV lookup and
+ Request-URI generation in chan_sip. This patch adds a new field
+ "portinuri" to the sip dialog struct and the sip peer struct.
+ That field is used during RURI generation to determine if the
+ port should be included in the RURI. It is also used in some
+ places to determine if an SRV lookup should occur. (closes issue
+ #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson
+ Review: https://reviewboard.asterisk.org/r/369/
+
+2009-09-30 19:02 +0000 [r221303] Matthias Nick <mnick at digium.com>
+
+ * funcs/func_strings.c: changed the prototype definition of
+ csv_quote
+
+2009-09-30 16:55 +0000 [r221200] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c: Avoid a potential NULL dereference. (closes issue
+ #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
+ uploaded by tilghman (license 14) Tested by: kobaz
+
+2009-09-30 15:41 +0000 [r221153-221157] Matthias Nick <mnick at digium.com>
+
+ * configs/cdr_custom.conf.sample, funcs/func_strings.c: added a new
+ dialplan function 'CSV_QUOTE' and changed the
+ cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
+ Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
+ mnick
+
+ * funcs/func_strings.c: check bounds - prevents for buffer overflow
+
+2009-09-30 14:49 +0000 [r221086] Terry Wilson <twilson at digium.com>
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
+ configs/sip.conf.sample: Change the SSRC by default when our
+ media stream changes Be default, change SSRC when doing an audio
+ stream changes Asterisk doesn't honor marker bit when reinvited
+ to already-bridged RTP streams,resulting in far-end stack
+ discarding packets with "old" timestamps that areactually part of
+ a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever
+ there is a reinvite, unless the 'constantssrc' is set to true in
+ sip.conf. The original issue reported to Digium support detailed
+ the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
+ Application Server Call comes in fromITSP, Asterisk dials the app
+ server which sends a re-invite back toAsterisk--not to negotiate
+ to send media directly to the ITSP, but to indicatethat it's
+ changing the stream it's sending to Asterisk. The app
+ servergenerates a new SSRC, sequence numbers, timestamps, and
+ sets the marker bit on the new stream. Asterisk passes through
+ the teimstamp of the new stream, butdoes not reset the SSRC,
+ sequence numbers, or set the marker bit. When the timestamp on
+ the new stream is older than the timestamp on the originalstream,
+ the ITSP (which doesn't know there has been any change) discards
+ the newframes because it thinks they are too old. This patch
+ addresses this by changing the SSRC on a stream update unless
+ constantssrc=true is set in sip.conf. Review:
+ https://reviewboard.asterisk.org/r/374/
+
+2009-09-29 20:14 +0000 [r220907] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
+ spyee is masqueraded and chanspy_ds_chan_fixup() is called with
+ the channel locked. (closes issue #15965) Reported by: atis
+ Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
+ (license 96) Tested by: atis
+
+2009-09-29 17:59 +0000 [r220873] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Reduce CPU usage related to building a peer
+ merely for devicestates. This fixes a 100% CPU problem in the SIP
+ driver, found by profiling the driver while the problem was
+ occurring. (closes issue #14309) Reported by: pkempgen Patches:
+ 20090924__issue14309.diff.txt uploaded by tilghman (license 14)
+ Tested by: pkempgen, vrban
+
+2009-09-28 19:09 +0000 [r220717] Sean Bright <sean at malleable.com>
+
+ * Makefile.rules: When selecting DONT_OPTIMIZE in menuselect,
+ explicitly pass -O0 to the compiler so we override any default
+ optimization levels for a particular install.
+
+2009-09-24 19:39 +0000 [r220288] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_playback.c, main/pbx.c, apps/app_disa.c: Implicitly
+ sending a progress signal breaks some applications. Call
+ Progress() in your dialplan if you explicitly want progress to be
+ sent. (Reverts change 216430, closes issue #15957) Reported by:
+ Pavel Troller on the Asterisk-Dev mailing list
+ http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
+
+2009-09-24 18:18 +0000 [r220099-220213] Sean Bright <sean at malleable.com>
+
+ * Makefile: Resolve parallel build warnings. Reported by Klaus
+ Darilion on the asterisk-dev mailing list.
+
+ * Makefile, build_tools/mkpkgconfig: Remove the remaining bashisms
+ in the Makefile/mkpkgconfig
+
+2009-09-24 08:33 +0000 [r220027] Michiel van Baak <michiel at vanbaak.info>
+
+ * build_tools/mkpkgconfig: mkpkgconfig does not need bash so make
+ it use /bin/sh This fixes building on all systems that don't have
+ bash at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys
+ on #asterisk-dev
+
+2009-09-22 21:37 +0000 [r219816] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: When IMAP variables were changed during a
+ reload, Voicemail did not use the new values. This change
+ introduces a configuration version variable, which ensures that
+ connections with the old values are not reused but are allowed to
+ expire normally. (closes issue #15934) Reported by:
+ viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by
+ tilghman (license 14) Tested by: viniciusfontes
+
+2009-09-21 16:55 +0000 [r219720] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Reverting merge 219520. This change was not
+ necessary.
+
+2009-09-20 17:52 +0000 [r219653] Tilghman Lesher <tlesher at digium.com>
+
+ * main/file.c: Really stop the stream, when ast_closestream() is
+ called. (closes issue #15129) Reported by: bmh Patches:
+ 20090918__issue15129.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/372/
+
+2009-09-19 02:51 +0000 [r219586] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Make sure the iax_pvt exists before
+ dereferencing it. This fixes the latest crash posted on issue
+ 15609. (issue #15609)
+
+2009-09-18 23:19 +0000 [r219450-219519] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: iax2 frame double free The iax frame's
+ retrans sched id was written over right before iax2_frame_free
+ was called. In iax2_frame_free that retrans id is used to delete
+ the sched item. By writing over the retrans field before the
+ sched item could be deleted, it was possible for a retransmit to
+ occur on a freed frame.
+
+ * channels/chan_sip.c: via-header branches not updated correctly on
+ INVITE INVITE requests must always contain a new unique branch
+ id. When a new branch id is created for an INVITE, the dialog's
+ invite_branch variable must be updated so CANCEL requests use the
+ correct branch id. (closes issue #15262) Reported by: maniax
+ Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
+ (license 608) invite_new_branch_trunk.diff uploaded by dvossel
+ (license 671) Tested by: maniax, dvossel
+
+2009-09-17 22:20 +0000 [r219320] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Send a 100 Trying response when we detect a
+ spiral. This was problematic during spiral tests at SIPit...
+ along with some other things as well.
+
+2009-09-17 21:29 +0000 [r219303] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: INVITE w/Replaces deadlock fix This patch
+ cleans up the locking logic in chan_sip.c's
+ handle_invite_replaces() function as well as making use of
+ ast_do_masquerade() rather than forcing the masquerade on an
+ ast_read(). The code had several redundant unlocks that would
+ result in 'freed more times than we've locked!' errors. I cleaned
+ these up as well as moving all the unlock logic to the end of the
+ function. This patch should also resolve the issue people were
+ having with the replacecall channel never being unlocked with one
+ legged calls. (closes issue #15151) Reported by: irroot Patches:
+ invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
+ Tested by: irroot, dvossel Review:
+ https://reviewboard.asterisk.org/r/371/
+
+2009-09-17 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.27-rc1
+
+2009-09-17 14:58 +0000 [r219136] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c, include/asterisk/cdr.h,
+ include/asterisk/channel.h: Prevent a potential race condition
+ and crash when hanging up a channel by removing the channel from
+ the channel list before begining channel tear down. This fix may
+ potentially cause problems with CDR backends that access the
+ channel a CDR is associated with via the channel list. This fix
+ makes the channel unavabile at the time when the CDR backend is
+ invoked. This has been documented in include/asterisk/cdr.h.
+ (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/362/
+
+2009-09-16 23:21 +0000 [r219023] Tilghman Lesher <tlesher at digium.com>
+
+ * main/config.c, configs/extensions.conf.sample: Properly deal with
+ quotes in the arguments of '#exec' includes. (closes issue
+ #15583) Reported by: pkempgen Patches:
+ 20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+ 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+ 169) Tested by: pkempgen
+
+2009-09-16 18:00 +0000 [r218867] David Brooks <dbrooks at digium.com>
+
+ * main/pbx.c: Fixes CID pattern matching behavior to mirror that of
+ extension pattern matching. Pattern matching for extensions uses
+ a type of scoring system, giving values for specificity to each
+ character in the pattern. Unfortunately, this is done character
+ by character, in order. This does lead to some less specific
+ patterns being first in line for matching, but it will usually
+ get the job done. This patch merely brings CID matching to the
+ same level as extension matching. This patch does not attempt to
+ tackle the problem shared by extension matching. (closes issue
+ #14708) Reported by: klaus3000
+
+2009-09-16 13:33 +0000 [r218798] Russell Bryant <russell at digium.com>
+
+ * contrib/firmware/iax/iaxy.bin (removed), UPGRADE.txt: Remove the
+ IAXy firmware from Asterisk. The firmware can now be found on
+ downloads.digium.com, where the rest of our binary downloads
+ live. This was the last part of our Asterisk tarballs that was
+ considered non-free by Debian. :-) (closes issue #15838) Reported
+ by: paravoid
+
+2009-09-15 22:27 +0000 [r218730] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: If the user enters the same password as
+ before, don't signal an error when the change does nothing.
+ (closes issue #15492) Reported by: cbbs70a Patches:
+ 20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+
+2009-09-15 16:29 +0000 [r218623] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Fix small memory leak in handle_init_event
+ by always destroying the pthread attr before returning.
+
+2009-09-15 16:03 +0000 [r218578] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Send request contact header field with
+ response to registrer queries instead of the address of record.
+ (closes issue #14438) Reported by: ravindrad Patches:
+ regquerypatch uploaded by ravindrad (license 684) Tested by:
+ ravindrad
+
+2009-09-15 16:01 +0000 [r218577] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_followme.c: Ensure FollowMe sets language in channels it
+ creates. Also, not in the original bug report, but related fields
+ are accountcode and musicclass, and the inheritance of
+ datastores. (closes issue #15372) Reported by: Romik Patches:
+ 20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+ Tested by: cervajs
+
+2009-09-15 14:57 +0000 [r218497-218498] Kevin P. Fleming <kpfleming at digium.com>
+
[... 25943 lines stripped ...]
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