[svn-commits] lmadsen: tag 1.4.27-rc2 r222226 - /tags/1.4.27-rc2/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Oct 6 11:02:49 CDT 2009
Author: lmadsen
Date: Tue Oct 6 11:02:47 2009
New Revision: 222226
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=222226
Log:
Importing files for 1.4.27-rc2 release.
Added:
tags/1.4.27-rc2/.lastclean (with props)
tags/1.4.27-rc2/.version (with props)
tags/1.4.27-rc2/ChangeLog (with props)
Added: tags/1.4.27-rc2/.lastclean
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+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>
+
+ * channels/chan_sip.c: revert accidental commit
+
+ * channels/chan_sip.c, sounds/Makefile: Use proper hostname for
+ downloading sound files.
+
+2009-09-14 21:47 +0000 [r218401] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Fix handling of DAHDI_EVENT_REMOVED event
+ to prevent crash in do_monitor. After talking to rmudgett about
+ some of his recent iflist locking changes, it was determined that
+ the only place that would destroy a channel without being
+ explicitly to do so was in handle_init_event. The loop to walk
+ the interface list has been modified to wait to destroy the
+ channel until the dahdi_pvt of the channel to be destroyed is no
+ longer needed. (closes issue #15378) Reported by: samy
+
+2009-09-14 19:16 +0000 [r218331] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, sounds/Makefile: Don't say "Please try
+ again" if we don't give the user another chance to try again.
+ (issue #15055, SWP-129) Reported by: jthurman
+
+2009-09-14 14:53 +0000 [r218223] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_directed_pickup.c: Ensure we don't pickup ourselves when
+ doing pickup by exten. (closes issue #15100) Reported by:
+ lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan
+ (license 779)
+
+2009-09-10 23:52 +0000 [r217917-217989] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: Don't ring another channel, if there's not
+ enough time for a queue member to answer. (Fixes AST-228)
+
+ * contrib/scripts/iax-friends.sql, channels/chan_sip.c,
+ channels/chan_iax2.c: Backport realtime fix to 1.4
+
+2009-09-10 21:06 +0000 [r217806] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: IAX2 encryption regression The IAX2 Call
+ Token security patch inadvertently broke the use of encryption
+ due to the reorganization of code in the socket_process()
+ function. When encryption is used, an incoming full frame must
+ first be decrypted before the information elements can be parsed.
+ The security release mistakenly moved IE parsing before
+ decryption in order to process the new Call Token IE. To resolve
+ this, decryption of full frames is once again done before looking
+ into the frame. This involves searching for an existing callno,
+ checking the pvt to see if encryption is turned on, and
+ decrypting the packet before the internal fields of the full
+ frame are accessed. associated with AST-2009-006 (closes issue
+ #15834) Reported by: karesmakro Patches:
+ iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
+ Tested by: dvossel, karesmakro Review:
+ https://reviewboard.asterisk.org/r/355/
+
+2009-09-10 19:52 +0000 [r217668-217735] Olle Johansson <oej at edvina.net>
+
+ * utils/Makefile: Reinstate muted that was removed by mistake.
+ muted doesn't compile any more on os/x, so I have to disable it
+ in order to testcompile other code...
+
+ * utils/Makefile, channels/chan_sip.c: Remove harmful code that
+ causes endless loops. Remove code that causes loops in
+ registrations. We have agreed that the patch that this code was
+ part of was bad. I am ripping out the code that causes the issue.
+ putnopvut needs to check the rest of the patch, if it needs to be
+ changed as well. This solves the issue reported in #15540, but
+ needs more work before we close it (as described above).
+
+2009-09-08 20:01 +0000 [r217156] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_meetme.c: When MOH is playing on the channel,
+ announcements sent through the conference are not heard. (closes
+ issue #14588) Reported by: voipas Patches:
+ 20090716__issue14588__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: lmadsen, twisted, tilghman
+
+2009-09-04 13:56 +0000 [r216432-216435] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/utils.c, include/asterisk/lock.h: make asterisk compile
+ under devmode with DEBUG_THREADS enabled on OpenBSD
+
+ * channels/chan_sip.c: make chan_sip compile under devmode again
+
+2009-09-04 13:45 +0000 [r216430] Olle Johansson <oej at edvina.net>
+
+ * apps/app_playback.c, main/pbx.c, channels/chan_sip.c,
+ apps/app_disa.c, configs/sip.conf.sample: Make apps send PROGRESS
+ control frame for early media and fix too early media issue in
+ SIP The issue at hand is that some legacy (dying) PBX systems
+ send empty media frames on PRI links *before* any call progress.
+ The SIP channel receives these frames and by default signals 183
+ Session progress and starts sending media. This will cause phones
+ to play silence and ignore the later 180 ringing message. A bad
+ user experience. The fix is twofold: - We discovered that
+ asterisk apps that support early media ("noanswer") did not send
+ any PROGRESS frame to indicate early media. Fixed. - We introduce
+ a setting in chan_sip so that users can disable any relay of
+ media frames before the outbound channel actually indicates any
+ sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be
+ disabled for backward compatibility. In later versions of
+ Asterisk, this will be enabled. We don't assume that it will
+ change your Asterisk phone experience - only for the better. We
+ encourage third-party application developers to make sure that if
+ they have applications that wants to send early media, add a
+ PROGRESS control frame transmission to make sure that all channel
+ drivers actually will start sending early media. This has not
+ been the default in Asterisk previous to this patch, so if you
+ got inspiration from our code, you need to update accordingly.
+ Sorry for the trouble and thanks for your support. This code has
+ been running for a few months in a large scale installation (over
+ 250 servers with PRI and/or BRI links to old PBX systems). That's
+ no proof that this is an excellent patch, but, well, it's tested
+ :-)
+
+2009-09-04 13:16 +0000 [r216369] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/astobj2.c: Make sure 'start' is always initialized. This is
+ the same as rev 216222 in trunk but 1.4 is affected as well
+
+2009-09-04 10:48 +0000 [r216008-216263] Russell Bryant <russell at digium.com>
+
+ * doc/IAX2-security.txt (added), /: Merged revisions 216262 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009)
+ | 2 lines Add a plain text version of the IAX2 security document.
+ ........
+
+ * /, UPGRADE.txt: Merged revisions 216080 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009)
+ | 2 lines Add a note about IAX2 to UPGRADE.txt. ........
+
+ * /, doc/IAX2-security.pdf (added): Merged revisions 216005 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009)
+ | 2 lines Add IAX2 security document related to AST-2009-006.
+ ........
+
+2009-09-03 18:32 +0000 [r216000] David Vossel <dvossel at digium.com>
+
+ * channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample,
+ include/asterisk/acl.h, channels/iax2-parser.h,
+ include/asterisk/astobj2.h, channels/iax2.h, main/acl.c,
+ channels/chan_iax2.c: Merge code associated with AST-2009-006
+ (closes issue #12912) Reported by: rathaus Tested by: tilghman,
+ russell, dvossel, dbrooks
+
+2009-09-02 21:41 +0000 [r215682] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Re-send non-100 provisional responses to
+ prevent cancellation From section 13.3.1.1 of RFC 3261: If the
+ UAS desires an extended period of time to answer the INVITE, it
+ will need to ask for an "extension" in order to prevent proxies
+ from canceling the transaction. A proxy has the option of
+ canceling a transaction when there is a gap of 3 minutes between
+ responses in a transaction. To prevent cancellation, the UAS MUST
+ send a non-100 provisional response at every minute, to handle
+ the possibility of lost provisional responses. (closes issue
+ #11157) Reported by: rjain Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/315/
+
+2009-09-01 23:04 +0000 [r215270] Dwayne M. Hubbard <dwayne.hubbard at gmail.com>
+
+ * apps/app_softhangup.c: Use strrchr() so SoftHangup will correctly
+ truncate multi-hyphen channel names In general channel names are
+ in the form Foo/Bar-Z, but the channel name could have multiple
+ hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the
+ channel name at the last hyphen. (closes issue #15810) Reported
+ by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by
+ dhubbard (license 733)
+
+2009-08-31 16:16 +0000 [r214940] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_local.c: Also unlock the "other" channel, when
+ returning, due to glare. (closes issue #15787) Reported by:
+ tim_ringenbach Patches: chan_local.diff uploaded by tim
+ ringenbach (license 540) Tested by: tim_ringenbach
+
+2009-08-28 20:13 +0000 [r214357-214701] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c: Modify comment to be a bit more accurate. We have
+ kept this comment around long enough, that it's pretty clear that
+ we're keeping the code, because changing the code would require a
+ pretty fundamental architectural shift. We've also taken
+ criticism in some quarters, because it was believed that it was
+ referring to the code being nasty. No, the code isn't nasty, just
+ the operation itself is rather odd. Fixed for eternity (probably
+ not).
+
+ * autoconf/libcurl.m4 (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Use autoconf to
+ detect libcurl, as this enables cross-compilation checks,
+ something we didn't allow before. (closes issue #15714) Reported
+ by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded
+ by tilghman (license 14) Tested by: pprindeville
+
+ * autoconf/ast_ext_lib.m4, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: One more build
+ system change, to make the descriptions look better, if we have
+ better information.
+
+ * autoconf/ast_ext_lib.m4, configure,
+ include/asterisk/autoconfig.h.in: Make autoheader descriptions
+ render correctly in our autoconfig.h file. (Figured out while
+ working with issue #14906)
+
+2009-08-26 16:36 +0000 [r214194] David Vossel <dvossel at digium.com>
+
+ * main/channel.c: ast_write() ignores ast_audiohook_write() results
+ In ast_write(), if a channel has a list of audiohooks, those
+ lists are written to and the resulting frame is what ast_write()
+ should continue with. The problem was the returned audiohook
+ frame was not being handled at all, and the original frame passed
+ into it did not contain the mixed audio, so essentially audio was
+ being lost. One result of this was chan_spy's whisper mode no
+ longer worked. To complicate the issue, frames passed into
+ ast_write may either be a single frame, or a list of frames. So,
+ as the list of frames is processed in the audiohook_write, the
+ returned frames had to be added to a new list. (closes issue
+ #15660) Reported by: corruptor Tested by: dvossel
+
+2009-08-25 19:28 +0000 [r213899-214069] Tilghman Lesher <tlesher at digium.com>
+
+ * main/say.c: I should always compile before committing...
+
+ * main/say.c: Fix pronunciation of German dates. (closes issue
+ #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
+ by Benjamin Kluck (license 803)
+
+ * main/pbx.c: Improve error message by informing user exactly which
+ function is missing a parethesis. (closes issue #15242) Reported
+ by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
+ dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
+ loloski (license 68)
+
+ * Makefile: Use the default runlevels for Debian derivatives,
+ instead of making up our own. (closes issue #14730) Reported by:
+ pkempgen
+
+2009-08-21 20:23 +0000 [r213631] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: Ensure that T.38 INVITEs generated by
+ Asterisk properly result in T.38 being enabled. (closes issue
+ #15373) Reported by: dcolombo Patches: chan_sip.patch uploaded by
+ mbrancaleoni (license 342) Tested by: dcolombo, mbrancaleoni
+
+2009-08-21 16:52 +0000 [r213559] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk.h: Permit DEBUG_FD_LEAKS to be used with C++
+ source files. (closes issue #15698) Reported by: slavon Patches:
+ 20090817__issue15698.diff.txt uploaded by tilghman (license 14)
+ Tested by: slavon, tilghman
+
+2009-08-21 16:03 +0000 [r213493] Jason Parker <jparker at digium.com>
+
+ * configs/queues.conf.sample: Clarify queues.conf comments to
+ specify that variables should be set in the dialplan. (closes
+ issue #15755) Reported by: trendboy
+
+2009-08-20 20:33 +0000 [r213339] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_features.c: Fix a crash by checking the proper pointer
+ for validity before deferencing it. (closes issue #15751)
+ Reported by: atis Patches: ast_bridge_call_peer_cdr.patch
+ uploaded by atis (license 242)
+
+2009-08-20 19:53 +0000 [r213283] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.exports (added): Make all the symbols for the
+ C-client callbacks global
+
+2009-08-19 21:18 +0000 [r213103] David Vossel <dvossel at digium.com>
+
+ * apps/app_mixmonitor.c: Fixes memory leak caused by incorrectly
+ freeing mixmonitor (closes issue #15699) Reported by: edantie
+ Patches: mixmonitor.patch uploaded by edantie (license 862)
+
+2009-08-18 20:26 +0000 [r212913] Kevin P. Fleming <kpfleming at digium.com>
+
+ * doc/musiconhold-opsound.txt (added), CREDITS, /, UPGRADE.txt,
+ sounds/sounds.xml, build_tools/prep_tarball,
+ doc/musiconhold-fpm.txt (removed), doc/00README.1st,
+ sounds/Makefile: Convert this branch to Opsound music-on-hold.
+ For more details:
+ http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
+
+2009-08-18 16:36 +0000 [r212763] Sean Bright <sean at malleable.com>
+
+ * main/manager.c: Delay the creation of temporary files until we
+ have a valid manager command to handle. Without this patch,
+ asterisk creates a temporary file before determining if the
+ specified command is valid. If invalid, we weren't properly
+ cleaning up the file. (closes issue #15730) Reported by: zmehmood
+ Patches: M15730.diff uploaded by junky (license 177) Tested by:
+ zmehmood
+
+2009-08-18 16:00 +0000 [r212727] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_lib.c: Removed some deadwood and added some
+ doxygen comments.
+
+2009-08-17 16:34 +0000 [r212498] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/misdn_config.c: Fix segfault when reloading chan_misdn.
+ If more ports were specified than configured in misdn.conf a
+ reload would crash asterisk. The problem was the unconfigured
+ port was using data from the previously configured port. When the
+ data for an unconfigured port was freed a crash would result from
+ the double free. (closes issue #12113) Reported by: agupta
+ Patches: bug12113.patch uploaded by jpeeler (license 325)
+
+2009-08-17 15:36 +0000 [r212430] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Fix uninitialized variable.
+
+2009-08-12 23:04 +0000 [r211953] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: This patch adds additional checking when
+ generating queue log TRANSFER events. The additional checks
+ prevent generation of false TRANSFER events in certain
+ situations. (closes issue #14536) Reported by: aragon Patches:
+ queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, mnicholson
+
+2009-08-12 18:46 +0000 [r211807] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Backport fix so that outbound CANCEL
+ requests have same branch as challenged INVITEs. There already
+ was code present to be sure that a CANCEL will contain the same
+ branch-id as the INVITE it is cancelling. However, for INVITES
+ which are challenged downstream, this mechanism did not work
+ properly. Now this is taken care of. This is a backport of a fix
+ already present in all 1.6.X branches and in trunk. It also fixes
+ ABE-1907.
+
+2009-08-10 19:48 +0000 [r211528-211583] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/CODING-GUIDELINES: Conversion specifiers, not format
+ specifiers
+
+ * main/indications.c, main/cli.c, pbx/pbx_loopback.c,
+ channels/chan_dahdi.c, res/res_smdi.c, pbx/pbx_spool.c,
+ channels/chan_skinny.c, pbx/pbx_ael.c, apps/app_dial.c,
+ main/pbx.c, apps/app_privacy.c, codecs/codec_speex.c,
+ funcs/func_math.c, channels/chan_agent.c, apps/app_morsecode.c,
+ apps/app_disa.c, channels/iax2-provision.c, funcs/func_cut.c,
+ pbx/dundi-parser.c, apps/app_talkdetect.c, channels/chan_misdn.c,
+ apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
+ pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
+ main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
+ doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c,
+ main/utils.c, apps/app_followme.c, utils/frame.c,
+ channels/misdn_config.c, main/cdr.c, main/channel.c,
+ channels/chan_phone.c, main/manager.c, apps/app_osplookup.c,
+ apps/app_setcallerid.c, res/res_agi.c, apps/app_rpt.c,
+ channels/chan_mgcp.c, apps/app_adsiprog.c, main/dnsmgr.c,
+ channels/chan_sip.c, apps/app_waitforsilence.c, agi/eagi-test.c,
+ main/acl.c, apps/app_queue.c, channels/chan_oss.c,
+ agi/eagi-sphinx-test.c, channels/chan_h323.c, pbx/pbx_dundi.c,
+ apps/app_sms.c, apps/app_verbose.c, apps/app_dahdibarge.c,
+ funcs/func_rand.c, apps/app_readfile.c, main/frame.c, /,
+ res/res_features.c, apps/app_record.c, funcs/func_strings.c,
+ apps/app_random.c, apps/app_alarmreceiver.c,
+ channels/chan_iax2.c: AST-2009-005
+
+2009-08-09 15:41 +0000 [r211274] Tilghman Lesher <tlesher at digium.com>
+
+ * main/astfd.c: Small oops. Clear the flags which have been
+ checked.
+
+2009-08-07 20:11 +0000 [r211112] Russell Bryant <russell at digium.com>
+
+ * apps/app_chanspy.c: Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936)
+
+2009-08-07 18:16 +0000 [r210913-211038] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: QUEUE_MEMBER_LIST _really_ wants the interface
+ name, not the membername. This is a partial revert of revision
+ 82590, which was an attempted cleanup, but in reality, it broke
+ QUEUE_MEMBER_LIST, which has always been intended as a method by
+ which component interfaces could be queried from the queue.
+ Membername isn't useful here, because that field cannot be used
+ to obtain further information about the member. See the
+ documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+ QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+ member argument for further justification. (closes issue #15664)
+ Reported by: rain Patches: app_queue-queue_member_list.diff
+ uploaded by rain (license 327)
+
+ * main/channel.c: Because channel information can be accessed
+ outside of the channel thread, we must lock the channel prior to
+ modifying it. (closes issue #15397) Reported by: caspy Patches:
+ 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy
+
+2009-08-05 19:18 +0000 [r210575] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Dialplan starts execution before the
+ channel setup is complete. * Issue 15655: For the case where
+ dialing is complete for an incoming call, dahdi_new() was asked
+ to start the PBX and then the code set more channel variables. If
+ the dialplan hungup before these channel variables got set,
+ asterisk would likely crash. * Fixed potential for overlap
+ incoming call to erroneously set channel variables as global
+ dialplan variables if the ast_channel structure failed to get
+ allocated. * Added missing set of CALLINGSUBADDR in the dialing
+ is complete case. (closes issue #15655) Reported by: alecdavis
+
+2009-08-05 18:46 +0000 [r210563] Leif Madsen <lmadsen at digium.com>
+
+ * doc/imapstorage.txt: Update imapstorage.txt documentation.
+ Updated the imapstorage.txt documentation to reflect that issues
+ with c-client versions older than 2007 seem to cause crashing
+ issues that are not seen with more recent versions. Documentation
+ has been updated to reflect this. (closes issue #14496) Reported
+ by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks
+
+2009-08-04 14:51 +0000 [r210237] Kevin P. Fleming <kpfleming at digium.com>
+
+ * Makefile: Eliminate spurious compiler warnings from system
+ headers on *BSD platforms. Ensure that system headers located in
+ /usr/local/include are actually treated as system headers by the
+ compiler, and not as local headers which are subject to warnings
+ from the -Wundef compiler option and others. (closes issue
+ #15606) Reported by: mvanbaak
+
+2009-08-03 16:15 +0000 [r210067] David Brooks <dbrooks at digium.com>
+
+ * channels/chan_dahdi.c: Fixes dialplan wildcard extension taking
+ precedence over call pickup code. Prior to this patch, a wildcard
+ extension in the dialplan (for example, _*.) would take
+ precedence over picking up a call in the channel's pickup group.
+ This patch simply moves the block of code handling pickup group
+ matching to above the extension matching code. (closes issue
+ #14735) Reported by: stevedavies Review:
+ https://reviewboard.asterisk.org/r/319/
+
+2009-08-03 16:11 +0000 [r210064-210066] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_oss.c, apps/app_playback.c, main/asterisk.exports,
+ configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac,
+ funcs/func_cut.c: Reverting index() fix, applying a different
+ methodology, based upon developer discussions. (related to issue
+ #15639)
+
+ * main/asterisk.exports, include/asterisk/compat.h: Helps if we
+ export the index() function. (Related to issue #15639)
+
+ * configure, include/asterisk/autoconfig.h.in, main/strcompat.c,
+ configure.ac: Apparently, some platforms don't have the index()
+ function. (closes issue #15639) Reported by: nmav
+
+2009-08-01 11:27 +0000 [r209838-209879] Russell Bryant <russell at digium.com>
+
+ * main/db1-ast/mpool/mpool.c: Resolve a valgrind warning about a
+ read from uninitialized memory. (issue #15396) Reported by:
+ aragon
+
[... 25477 lines stripped ...]
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