[asterisk-commits] russell: tag 1.4.14 r89343 - in /tags/1.4.14: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Nov 16 16:44:47 CST 2007
Author: russell
Date: Fri Nov 16 16:44:46 2007
New Revision: 89343
URL: http://svn.digium.com/view/asterisk?view=rev&rev=89343
Log:
importing files for 1.4.14 release
Added:
tags/1.4.14/.lastclean (with props)
tags/1.4.14/.version (with props)
tags/1.4.14/ChangeLog (with props)
Added: tags/1.4.14/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.14/.lastclean?view=auto&rev=89343
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--- tags/1.4.14/ChangeLog (added)
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+2007-11-16 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.14 released.
+
+2007-11-16 22:26 +0000 [r89339] Russell Bryant <russell at digium.com>
+
+ * main/loader.c, include/asterisk/module.h,
+ build_tools/make_buildopts_h: Temporarily revert revision 89325,
+ which added md5 magic for keeping track of what build options
+ were used. We agreed that we should remove this before making a
+ 1.4 release, and then we can put it back in. Then, we can take a
+ month or so to play around with it to get it how we want it.
+
+2007-11-16 16:47 +0000 [r89325] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/loader.c, include/asterisk/module.h,
+ build_tools/make_buildopts_h: To help combat problems where
+ people build external modules (asterisk-addons or others) and
+ then change the build options of the Asterisk build in a way that
+ makes the incompatible without warning, this commit introduces an
+ MD5 signature of the important build-time options and includes
+ that signature into modules when they are built. When the loader
+ loads one of these modules and notices the problem, it will emit
+ a warning to console and refuse to initialize the module, as
+ doing so could cause the system to be unstable or even crash. If
+ you upgrade to this version of Asterisk, you must rebuild *all*
+ of your modules that came from other sources before trying to run
+ this version. If you are using Digium's G.729 binary codec
+ module, you will need v33 or newer.
+
+2007-11-16 15:28 +0000 [r89323] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Make realtime queues accessible from the
+ QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
+ patched by atis, with small modifications from me)
+
+2007-11-15 18:37 +0000 [r89298-89302] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile: Start Asterisk in Debian at a more reasonable time
+ (since zaptel is at level 20)
+
+ * channels/misdn/isdn_lib.c: Fix an uninitialized memory read found
+ by valgrind
+
+ * channels/chan_iax2.c: Yet another memory corruption issue.
+ Reported by: atis Patch by: tilghman Fixes issue #10923
+
+2007-11-15 17:19 +0000 [r89296] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: Update the SLAStation application to account
+ for the case where the SLA thread has a call out to the station,
+ but the user has pressed a line button to answer the call instead
+ of picking up the handset. If they do, the phone sends out a new
+ INVITE. So, the SLAStation app must check to see if it is picking
+ up a ringing trunk, and ensure that the other stations stop
+ ringing. (reported internally, patched by me, tested by mogorman)
+
+2007-11-15 14:57 +0000 [r89286-89288] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c: Undoing previous commit since I realize it was
+ wrong
+
+ * main/manager.c: Adding a missing mutex unlock. (closes issue
+ 11256, reported and patched by ys)
+
+2007-11-15 11:26 +0000 [r89280-89281] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Don't send re-invites during pending INVITE
+ transactions. Patch by one47 - thanks! Closes issue #9305
+
+ * channels/chan_sip.c: Improve support for multipart messages. Code
+ by gasparz, changes by me (mostly formatting). Thanks, gasparz!
+ Closes issue #10947
+
+2007-11-14 23:23 +0000 [r89275] Tilghman Lesher <tlesher at digium.com>
+
+ * main/app.c: When a recording ends with '#', we are improperly
+ trimming an extra 200ms from the recording. Reported by: sim
+ Patch by: tilghman Closes issue #11247
+
+2007-11-14 01:15 +0000 [r89260] Joshua Colp <jcolp at digium.com>
+
+ * main/srv.c: Return the proper value when the srv_callback
+ function executes properly. (closes issue #11240) Reported by:
+ jtodd
+
+2007-11-13 21:07 +0000 [r89248-89254] Jason Parker <jparker at digium.com>
+
+ * channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer
+ systems which require a third arg to open() when using O_CREAT.
+ Issue 11238, reported by puzzled.
+
+ * res/res_features.c: Revert change from revision 67064. It is
+ documented behavior that if a parking extension already exists
+ while using PARKINGEXTEN, dialplan execution will continue. If
+ blind transferring to a Park with PARKINGEXTEN, you must keep
+ this in mind, and handle the failure yourself. Issue 11237,
+ reported by jon.
+
+2007-11-13 17:34 +0000 [r89246] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: If we set a value for qualify, we should
+ actually pay attention to it, instead of overriding the value
+
+2007-11-13 16:02 +0000 [r89241] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_mixmonitor.c: Reverting commit made in revision 89205
+ since it is unnecessary. Thanks to Kevin for pointing this out
+
+2007-11-13 13:51 +0000 [r89239] Tilghman Lesher <tlesher at digium.com>
+
+ * main/utils.c: Debugging is running into the 16-lock limit.
+ Increase to avoid. (This define is only effective when debugging
+ is turned on, so there's no effect for most installations.)
+
+2007-11-13 00:56 +0000 [r89205] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If
+ only 1 argument is given, then the args.options and
+ args.post_process strings are uninitialized and could contain
+ garbage. This change handles this situation properly by only
+ using arguments that we have parsed.
+
+2007-11-12 20:46 +0000 [r89194] Jason Parker <jparker at digium.com>
+
+ * main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev
+
+2007-11-12 20:16 +0000 [r89184-89191] Tilghman Lesher <tlesher at digium.com>
+
+ * main/config.c: If two config writes collide, file corruption
+ could result. Use a mkstemp() file, instead. Reported by:
+ paravoid Patch by: tilghman Closes issue #10781
+
+ * main/channel.c, channels/chan_sip.c: Fix two cases of memory
+ corruption caused by background threads. Reported by: atis Patch
+ by: tilghman Fixes issue #10923
+
+2007-11-12 11:26 +0000 [r89169-89173] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and
+ no number was dialed and overlapdial is set, we wait for the ISDN
+ timeout instead of starting our own timer. added a comment for
+ the misdn.conf.sample for the overlapdial config option.
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
+ channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
+ restart all interfaces Restart_Indicator, to automatically send a
+ RESTART after the L2 of a PTP Port comes up. Also fixed some
+ places where we have send a RELEASE without need for it.
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
+ state/event issue with overlapdial=yes when no extension matched.
+ removed the general sending of a RELEASE_COMPLETE when we receive
+ a RELEASE, this is done by mISDNuser/mISDN. This makes it
+ possible to use asterisk-1.4 with mISDN trunk, but requires users
+ of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6
+ (when using the NT mode at all)
+
+ * channels/misdn/isdn_lib.c: fixed the support for CW and therefore
+ for the reject_cause option.
+
+ * channels/misdn/isdn_lib.c, channels/misdn_config.c,
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
+ aded ntkeepcalls option, to avoid droÃpping calls when the L2
+ goes down on a PTP link. There are some pbx which do turn off the
+ L1 for a very short while and restart it immediately. normally
+ T310 should be started and after 10 seconds or so the calls
+ should be dropped, this is a simple fix wihtout this timer.
+
+2007-11-08 23:52 +0000 [r89125] Jason Parker <jparker at digium.com>
+
+ * main/say.c: Properly say the seconds here.. Issue 11203, fix
+ described by vma.
+
+2007-11-08 21:00 +0000 [r89119] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Rework of the commit I made yesterday to use
+ the already built-in ast_uri_decode function as opposed to my
+ home-rolled one. Also added comments. Thanks to oej for pointing
+ me in the right direction
+
+2007-11-08 18:45 +0000 [r89115] Jason Parker <jparker at digium.com>
+
+ * configs/res_odbc.conf.sample: Avoid warnings on load when using
+ sample configuration files. Issue 11195, patch by eliel.
+
+2007-11-08 16:47 +0000 [r89111] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: I made this same adjustment in trunk to fix
+ a bug, and it makes sense to do it in 1.4 as well. If an
+ imapfolder is specified in voicemail.conf, don't ever explicitly
+ connect to INBOX since it may not exist.
+
+2007-11-08 05:26 +0000 [r89105] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/srv.c: fix a glaring bug in the new SRV record handling that
+ would cause incorrect weight sorting
+
+2007-11-08 04:55 +0000 [r89103] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/valgrind.txt: Typo
+
+2007-11-08 02:26 +0000 [r89095-89101] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Do not add a sip: to the beginning of the To
+ URI unless needed. (closes issue #10756) Reported by: goestelecom
+
+ * channels/chan_sip.c: Improve the devicestate logic for multiple
+ devices. If any are available then the extension is considered
+ available. (closes issue #10164) Reported by: nic_bellamy
+ Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic
+ (license 299)
+
+ * channels/chan_sip.c: Add support for allowing one outgoing
+ transaction. This means if a response comes back out of order
+ chan_sip will still handle it. I dream of a chan_sip with real
+ transaction support. (closes issue #10946) Reported by: flefoll
+ (closes issue #10915) Reported by: ramonpeek (closes issue #9567)
+ Reported by: atca_pres
+
+ * channels/chan_sip.c: If callerid is configured in sip.conf use
+ that for checking the presence of an extension in the dialplan.
+ (closes issue #11185) Reported by: spditner
+
+2007-11-07 23:39 +0000 [r89093] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: The member refcount must be incremented, to
+ avoid using it after deallocation. A huge thanks go to lvl- for
+ patiently providing the necessary valgrind output that was
+ necessary to finding this problem of memory corruption. Reported
+ by: lvl- Patch by: tilghman Closes issue #11174
+
+2007-11-07 22:40 +0000 [r89090] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: This patch makes it possible for SIP phones
+ to dial extensions defined with '#' characters in extensions.conf
+ AND maintain their escaped characters when forming URI's (closes
+ issue #10681, reported by cahen, patched by me, code review by
+ file)
+
+2007-11-07 21:40 +0000 [r89088] Steve Murphy <murf at digium.com>
+
+ * cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to
+ 10578, I just ran 1.4 thru valgrind; some of the config leakage
+ I've already fixed, but it doesn't hurt to double check. I found
+ and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major,
+ tho.
+
+2007-11-07 15:56 +0000 [r89085] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c: Fixing a segfault in the manager "core show
+ channels concise" command. (closes issue #11183, reported by arnd
+ and patched by ys)
+
+2007-11-07 04:07 +0000 [r89079] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.ael.sample: Suppress AEL warnings on load.
+ Reported by: eliel Patch by: eliel Closes issue #11178
+
+2007-11-06 20:18 +0000 [r89053] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: Fix init_classes() so that classes that
+ actually do have files loaded aren't treated as empty, and
+ immediately destroyed ...
+
+2007-11-06 19:09 +0000 [r89046] Jason Parker <jparker at digium.com>
+
+ * codecs/codec_zap.c: Correctly set the total number of channels
+ from a zaptel transcoder board. SPD-49, patch by Matthew
+ Nicholson.
+
+2007-11-06 19:09 +0000 [r89045] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/lock.h: We went to the trouble of creating a
+ method of tracking failed trylocks, then never turned it on
+ (oops).
+
+2007-11-06 18:53 +0000 [r89042] Olle Johansson <oej at edvina.net>
+
+ * main/tdd.c: Bug fixes to tdd support in zaptel.
+
+2007-11-06 18:20 +0000 [r89037] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: If someone were to delete the files used
+ by an existing MOH class, and then issue a reload, further use of
+ that class could result in a crash due to dividing by zero. This
+ set of changes fixes up some places to prevent this from
+ happening. (closes issue #10948) Reported by: jcomellas Patches:
+ res_musiconhold_division_by_zero.patch uploaded by jcomellas
+ (license 282) Additional changes added by me.
+
+2007-11-06 17:52 +0000 [r89036] Steve Murphy <murf at digium.com>
+
+ * main/config.c: closes issue #8786 - where the [catname](!) and
+ [catname](othercat1,othercat2,...) notation gets dropped across a
+ ConfigUpdate (or any other thing that would cause a config file
+ to be written). While I was at it, I also cleaned up some of the
+ destroy routines to free up comments, which was not being done.
+ Made sure the new struct I introduced is also cleaned up properly
+ at destruction time. My code handles multiple template
+ inclusions. Many thanks to ssokol for his patch, which, while not
+ literally used in the final merge, served as a foundation for the
+ fix.
+
+2007-11-06 17:08 +0000 [r88994-89032] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Make it so that if a peer is determined to
+ be unreachable using qualify their devicestate will report back
+ unavailable. (closes issue #11006) Reported by: pj
+
+ * channels/chan_zap.c: Fix improbable but possible memory leaks in
+ chan_zap. (closes issue #11166) Reported by: eliel Patches:
+ chan_zap.c.patch uploaded by eliel (license 64)
+
+2007-11-06 13:50 +0000 [r88931] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/lock.h: Remove some checks to see if locks are
+ initialized from the non-DEBUG_THREADS versions of the lock
+ routines. These are incorrect for a number of reasons: - It
+ breaks the build on mac. - If there is a problem with locks not
+ getting initialized, then the proper fix is to find that place
+ and fix the code so that it does get initialized. - If additional
+ debug code is needed to help find the problem areas, then this
+ type of things should _only_ be put in the DEBUG_THREADS
+ wrappers.
+
+2007-11-06 02:52 +0000 [r88862] Kevin P. Fleming <kpfleming at digium.com>
+
+ * include/asterisk/srv.h: update comment to match the state of the
+ code
+
+2007-11-05 23:29 +0000 [r88826] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Reworked deadlock avoidance in __ast_read.
+ Restored audio to callback agents. (closes issue #11071, reported
+ by callguy, patched by me, tested by callguy and Ted Brown)
+
+2007-11-05 22:07 +0000 [r88709-88805] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c, include/asterisk/pbx.h: After seeing crashes related
+ to channel variables, I went looking around at the ways that
+ channel variables are handled. In general, they were not handled
+ in a thread-safe way. The channel _must_ be locked when reading
+ or writing from/to the channel variable list. What I have done to
+ improve this situation is to make pbx_builtin_setvar_helper() and
+ friends lock the channel when doing their thing. Asterisk API
+ calls almost all lock the channel for you as necessary, but this
+ family of functions did not. (closes issue #10923, reported by
+ atis) (closes issue #11159, reported by 850t)
+
+ * channels/chan_sip.c: When traversing the list of channel
+ variables here in transmit_invite(), the asterisk channel must be
+ locked, as this data may change at any time. (I have seen
+ numerous reports of crashes related to the handling of channel
+ variables. There are a couple of issues on the bug tracker
+ related to it, but it has also been noted on IRC and mailing
+ lists. So, I am finding and fixing some places where channel
+ variables are handled improperly.)
+
+ * channels/chan_sip.c: Fix up some indentation.
+
+ * main/srv.c, include/asterisk/srv.h: Merge changes from
+ asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
+ record support in Asterisk was broken. There was no guarantee on
+ what record Asterisk would choose to actually use. This set of
+ changes improves the situation by ensuring that Asterisk will
+ choose the highest priority record.
+
+ * main/channel.c: Merge the last bit of changes from
+ asterisk/team/russell/readq-1.4 The issue here is that the
+ channel frame readq handling got broken when the code was
+ converted to use the linked list macros. It caused corruption of
+ the list head and tail pointers. So, I fixed up the usage of the
+ linked list macros and in passing, simplified the code. I also
+ documented what the code is doing, as it was a bit difficult to
+ figure out at first. This bug showed itself with crashes showing
+ messed up head/tail pointers for the readq. However, there are a
+ couple of crashes that aren't quite as obvious, but I think may
+ be related. So, if your bug gets closed by this commit, but you
+ still have a problem, please reopen or create a new bug report.
+ (closes issue #10936) (closes issue #10595) (closes issue #10368)
+ (closes issue #11084) (closes issue #10040) (closes issue #10840)
+
+2007-11-05 18:47 +0000 [r88671] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: If a SIP channel is put on hold multiple
+ times do not keep incrementing the onHold value. (closes issue
+ #11085) Reported by: francesco_r Tested by: blitzrage (closes
+ issue #10474) Reported by: acennami
+
+2007-11-05 17:46 +0000 [r88624] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Fix up datastore handling in ast_do_masquerade().
+ The code is intended to move any channel datastores from the old
+ channel to the new one. However, it did not use the linked list
+ macros properly to accomplish the task. The existing code would
+ only work if there was only a single datastore on the old
+ channel.
+
+2007-11-05 17:19 +0000 [r88585] Jason Parker <jparker at digium.com>
+
+ * channels/chan_sip.c: Make sure we destroy the config structure on
+ configuration failure. Issue 11163, patch by eliel.
+
+2007-11-05 16:20 +0000 [r88539] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_odbc.c: Don't check used pooled connections for
+ connection status, as it will cause issues for prepared queries.
+ Reported by: Nick Gorham (via -dev list) Patch by: tilghman
+
+2007-11-04 22:38 +0000 [r88471] Luigi Rizzo <rizzo at icir.org>
+
+ * include/asterisk/stringfields.h, main/channel.c,
+ apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c:
+ Rename ast_string_field_free_pool to
+ ast_string_field_free_memory, and ast_string_field_free_all to
+ ast_string_field_reset_all to avoid misuse (due to too similar
+ names and an error in documentation). Fix two related memory
+ leaks in app_meetme. No need to merge to trunk, different fix
+ already applied there. Not applicable to 1.2
+
+2007-11-02 20:49 +0000 [r88328-88366] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Make subscribecontext behave as advertised.
+ It will now look for the presence of a hint in the given context
+ (be it subscribecontext or context). (closes issue #10702)
+ Reported by: slavon
+
+ * channels/chan_sip.c: If an INFO request within a dialog is
+ received with a content length of 0 simply send back a 200 OK. It
+ is valid to do this and the remote side is probably using it to
+ make sure the signalling is still alive. (closes issue #5747)
+ Reported by: chandi Patches: infofix-81430-1.patch uploaded by
+ IgorG (license 20)
+
+2007-11-02 16:51 +0000 [r88283] Jason Parker <jparker at digium.com>
+
+ * main/say.c: We need to make sure to specify a language to
+ ast_fileexists, otherwise it may fail for anything besides en
+ Issue 11147, fix discovered by both citats and myself
+ (independently), with input from Corydon76
+
+2007-11-02 13:03 +0000 [r88116-88210] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy
+ Patch by: ys Closes issue #11143
+
+ * doc/valgrind.txt (added): Add some notes on using valgrind
+
+2007-11-01 16:21 +0000 [r88078] Jason Parker <jparker at digium.com>
+
+ * channels/chan_zap.c: Make sure we set the poll fds to NULL after
+ free()ing it. Part of issue 11017, patch by tzafrir.
+
+2007-11-01 13:27 +0000 [r87970-88026] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_meetme.c: Fix up commit for my Zap channel with spies in
+ Meetme fix. (thanks Tony Mountifield!)
+
+ * apps/app_meetme.c: If a Zap channel contains a spy or a spy is
+ added take it out of the conference in kernel space and make it
+ go through Asterisk so the spy gets audio from both sides.
+ (closes issue #10060) Reported by: mparker
+
+2007-10-31 21:23 +0000 [r87906-87908] Jason Parker <jparker at digium.com>
+
+ * res/res_jabber.c: Make sure we free some allocated memory before
+ returning. Issue 11131, patch by eliel.
+
+ * channels/chan_gtalk.c: Don't try to allocate memory that we're
+ just going to re-allocate later anyways. Issue 11130, patch by
+ eliel.
+
+2007-10-31 18:03 +0000 [r87852] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile: Create samples for ALL of the available options in
+ asterisk.conf
+
+2007-10-31 17:49 +0000 [r87775-87849] Steve Murphy <murf at digium.com>
+
+ * pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan
+ save' cli command saves a file where the semicolon is not
+ escaped. Fixed this; User also wanted comments to be preserved
+ across dialplan save, but this is impossible at this point in
+ time, because comments are not stored in the dialplan. They are
+ 'compiled' out of extensions.conf. The only way to preserve those
+ comments is to use the config file reader/writer that the GUI
+ uses to allow online user edits. extensions.conf is first and
+ foremost, a config file, and is read in by the normal config-file
+ reading routines. Then, it is processed into a dialplan
+ (context/exten structs).
+
+ * pbx/pbx_ael.c: Included some verbage in the check_includes func,
+ to inform the user that included contexts that have no match in
+ the AEL, might be OK, as AEL cannot check in the extensions.conf
+ or the in-memory contexts, as they may not be there at the time
+ of the check.
+
+2007-10-30 23:02 +0000 [r87739] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD
+ Reported by: ys Fixed by: ys Closes issue #11116
+
+2007-10-30 21:19 +0000 [r87686] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Merge the changes from
+ team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a
+ race condition related to the handling of POKEing peers.
+ Essentially, a reference to a peer is held by the scheduler when
+ there are pending callbacks, but the reference count didn't
+ reflect it. So, it was possible for a peer to hit a reference
+ count of zero and have its destructor begin to be called at the
+ same time that the scheduler thread ran a POKE related callback.
+ If that happened, a crash would likely occur. (closes issue
+ #11082, closes issue #11094)
+
+2007-10-30 20:29 +0000 [r87650] Jason Parker <jparker at digium.com>
+
+ * channels/Makefile: Only try to clean out h323/ if the
+ h323/Makefile exists.
+
+2007-10-30 16:13 +0000 [r87571] Joshua Colp <jcolp at digium.com>
+
+ * res/res_features.c: Add two more checks before printing out a
+ warning message about bridging. If either channel has hungup of
+ course the bridge will have failed. (closes issue #10009)
+ Reported by: dimas
+
+2007-10-30 15:45 +0000 [r87567] Jason Parker <jparker at digium.com>
+
+ * main/editline/np/vis.c: Fix build of editline on Solaris. Issue
+ 11113, patch by snuffy.
+
+2007-10-30 15:10 +0000 [r87534] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_followme.c: Return 1.4 to a state where it builds.
+ Changing the arguments to a function and not changing where they
+ are used is bad, mmmk?
+
+2007-10-30 14:31 +0000 [r87514] BJ Weschke <bweschke at btwtech.com>
+
+ * apps/app_followme.c: Fix issue where the recorded name wasn't
+ getting removed correctly. (closes issue #11115) Reported by:
+ davevg Patches: followme-v3.diff
+
+2007-10-29 22:13 +0000 [r87460-87465] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/gsm: missed one directory
+
+ * codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs,
+ codecs/lpc10, main/db1-ast, main/editline, main,
+ codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile,
+ codecs/lpc10/Makefile, utils, codecs, agi,
+ main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr:
+ clean up (and ignore) assembler and preprocessor intermediate
+ files if any are created during the build
+
+ * Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is
+ already there (used when debugging preprocessor issues) because
+ the compiler will whine about each compile command
+
+2007-10-29 21:06 +0000 [r87427] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Removing a completely unnecessary quota
+ check from IMAP code.
+
+2007-10-29 20:22 +0000 [r87373-87396] Russell Bryant <russell at digium.com>
+
+ * main/utils.c, include/asterisk/lock.h: Add some more details to
+ the output of "core show locks". When a thread is waiting for a
+ lock, this will now show the details about who currently has it
+ locked. (inspired by issue #11100)
+
+ * main/astmm.c: Remove a lock that doesn't make any sense. The
+ regions lock needs to be held when traversing the list of
+ allocated chunks so that they can be printed out to the CLI.
+ (Thanks to eliel on #asterisk-dev for pointing this out!)
+
+2007-10-29 17:20 +0000 [r87342] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix issue where if both sides of the dialog
+ cancelled the dialog at the same time chan_sip could kepe
+ retransmitting a response for no reason. (closes issue #9566)
+ Reported by: atca_pres Patches: bug9566.patch uploaded by oej
+
+2007-10-29 17:13 +0000 [r87340] Jason Parker <jparker at digium.com>
+
+ * funcs/func_realtime.c, funcs/func_cut.c: Allow some function
+ modules to compile under dev mode. Issue 11104, patch by andrew.
+
+2007-10-29 14:23 +0000 [r87294] Joshua Colp <jcolp at digium.com>
+
+ * main/utils.c: Fix issue with ast_unescape_semicolon going into an
+ endless loop. (closes issue #10550) Reported by: ramonpeek
+ Patches: unescape-85177-1.patch uploaded by IgorG (license 20)
+
+2007-10-28 13:46 +0000 [r87262] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c,
+ funcs/func_cut.c: Add autoservice to several more functions which
+ might delay in their responses. Also, make sure that func_odbc
+ functions have a channel on which to set variables. Reported by
+ russell Fixed by tilghman Closes issue #11099
+
+2007-10-26 16:34 +0000 [r87168] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c,
+ pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c,
+ include/asterisk/ael_structs.h, pbx/ael/ael.tab.h,
+ utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16,
+ pbx/ael/ael.flex: closes issue #11086 where a user complains that
+ references to following contexts report a problem; The problem
+ was REALLy that he was referring to empty contexts, which were
+ being ignored. Reporter stated that empty contexts should be OK.
+ I checked it out against extensions.conf, and sure enough, empty
+ contexts ARE ok. So, I removed the restriction from AEL. This,
+ though, highlighted a problem with multiple contexts of the same
+ name. This should be OK, also. So, I added the extend keyword to
+ AEL, and it can preceed the 'context' keyword (mixed with
+ 'abstract', if nec.). This will turn off the warnings in AEL if
+ the same context name is used 2 or more times. Also, I now call
+ ast_context_find_or_create for contexts now, instead of just
+ ast_context_create; I did this because pbx_config does this. The
+ 'extend' keyword thus becomes a statement of intent. AEL can now
+ duplicate the behavior of pbx_config,
+
+2007-10-26 13:54 +0000 [r87120] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_curl.c: The addition of autoservice to func_curl
+ additionally made func_curl dependent on the existence of a
+ channel, with no real reason. This should make func_curl once
+ again work without a channel. Reported by jmls. Fixed by
+ tilghman. Closes issue #11090
+
+2007-10-25 23:03 +0000 [r87069] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/channel.c, include/asterisk/linkedlists.h: appending one
+ list to another should leave the first list empty, and not
+ require the user to do that
+
+2007-10-25 22:53 +0000 [r87067] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_cut.c: Backport alternate encoding of newline
+ delimiters from trunk to 1.4, as approved by Russell Reported by
+ blitzrage Closes issue #10903
+
+2007-10-24 20:56 +0000 [r86982] Jason Parker <jparker at digium.com>
+
+ * channels/chan_zap.c: Correctly respect hidecalleridname
+ configuration option. Simplify code slightly in the process.
+ Issue 11079, reported by ddv2005
+
+2007-10-24 04:14 +0000 [r86880-86936] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable
+ to specify app:spec in hint arguments
+
+ * funcs/func_logic.c: closes issue #11052 -- where nothing after
+ the ? will allow un-initialized variable values to corrupt and
+ crash asterisk on 64-bit platforms
+
+ * main/Makefile: this update to Makefile corrects how ast_expr2f.c
+ should be generated
+
+ * main/ast_expr2f.c: This should get rid of a really, really
+ irritating warning generated by some 64-bit platforms from libc,
+ where free(0) is frowned upon
+
+2007-10-22 21:36 +0000 [r86836] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/lock.h: If lock tracking is not enabled, then we
+ can not attempt to log any mutex failures. If so, we could end up
+ in infinite recursion. The only lock that is affected by this is
+ a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes
+ issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
+ (license 281)
+
+2007-10-22 17:38 +0000 [r86787] Tilghman Lesher <tlesher at digium.com>
+
+ * main/astmm.c: Minor FreeBSD build fix
+
+2007-10-22 16:35 +0000 [r86754-86756] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: After reading online I have confirmed that
+ Record-Route headers should be copied to 1xx responses as well.
+ (closes issue #10113) Reported by: makoto
+
+ * apps/app_controlplayback.c: Make sure res is a positive value
+ before performing the check to determine whether the user stopped
+ it or not. (closes issue #11023) Reported by: cfc
+
+2007-10-22 15:52 +0000 [r86726-86750] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Don't leak a frame in the case that an END frame
+ is received and the time since the BEGIN is less than that of the
+ defined minimum DTMF duration. (closes issue #11051) Reported by:
+ casper Patches: channel.c.86664.diff uploaded by casper (license
+ 55)
+
+ * include/asterisk/lock.h: Update the static mutex initializer to
+ include the initialization of the internal mutex used to protect
+ the lock debugging data. (closes issue #11044, patch suggested by
+ Ivan)
+
+2007-10-22 14:48 +0000 [r86694] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Account for the fact that sometimes headers
+ may be terminated with \r\n instead of just \n (closes issue
+ #11043, reported by yehavi)
+
+2007-10-22 14:27 +0000 [r86630-86663] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c: Move log message to before the frame it
+ references is freed. (closes issue #11050) Reported by: slavon
+ Patches: channel.c.86662.diff uploaded by casper (license 55)
+
+ * pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes
+ issue #11041) Reported by: jsmith Patches:
+ asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
+ (license 176)
+
+ * main/loader.c: Fixes for building under OpenSolaris. (closes
+ issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
+ uploaded by snuffy (license 35)
+
+2007-10-22 09:21 +0000 [r86598] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send
+ DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan
+ does not match after an overlap call. Also added out_cause=1
+
+2007-10-19 16:38 +0000 [r86469-86502] Joshua Colp <jcolp at digium.com>
+
+ * main/app.c: When returning a DTMF digit from
+ ast_control_streamfile cast it as a char so that 0 does not
+ overlap with the success return code. (closes issue #11023)
+ Reported by: cfc
+
+ * channels/chan_sip.c: Fix two issues with domains and transfers.
+ If a port was given in the hostname it was treated as part of the
+ hostname. If domains were configured but external domains were
+ not enabled all transfers would be considered remote. (closes
+ issue #11027) Reported by: ramonpeek Patches: 11027-1.diff
+ uploaded by ramonpeek (license 266)
+
+ * channels/chan_sip.c: Set port number in received as information
+ for registrations as well. (closes issue #11028) Reported by:
+ brad-x
+
+2007-10-19 01:45 +0000 [r86438] TransNexus OSP Development <support at transnexus.com>
+
+ * apps/app_osplookup.c: Fixed OSP module did not report
+ source/devinfo IP in correct format.
+
+2007-10-18 22:01 +0000 [r86405-86406] Jason Parker <jparker at digium.com>
+
+ * Makefile: Correct documentation. I removed the wrong line..
+
+ * Makefile: Add documentation for options in asterisk.conf Issue
+ 11029, patch by eserra
+
+2007-10-18 21:16 +0000 [r86330-86372] Russell Bryant <russell at digium.com>
+
+ * configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous
+ commit.
+
+ * configs/iax.conf.sample, channels/chan_iax2.c: Add support for
+ setting the maximum trunk size for IAX2 trunking
+
+ * main/channel.c, include/asterisk/channel.h: The channel needs to
+ stay locked while running timer callbacks, as they access and
+ modify channel data that may change elsewhere. I went through
+ every timer callback in the source tree to make sure that none of
+ them did any additional locking that could introduce deadlocks,
+ and all is well. (closes issue #10765) Reported by: Ivan Patches:
+ ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license
+ 229)
+
+2007-10-18 17:38 +0000 [r86328] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: If a non-existent file is specified to be
+ played either as a periodic announcement or as a hold/position
+ announcement, the caller would be kicked out of the queue. No
+ longer does this happen.
+
+2007-10-18 15:45 +0000 [r86237-86296] Russell Bryant <russell at digium.com>
+
+ * codecs/codec_zap.c: Execute the RELEASE operation on transcoder
+ channels in the destroy callback. (patch from jsloan)
+
+ * main/utils.c: Revert a change that I made for issue #10979 which,
+ as has been pointed out to me in issue #11018, doesn't really
+ make sense. There is no reason to have the base64 decode function
+ force a '\0' terminated buffer, when the result is almost always
+ binary, anyway. In fact, this caused some breakage, as some code
+ in res_crypto passed in a buffer exactly the right size to get
+ its binary result, which got stomped on by this patch. (closes
+ issue #11018, reported by dimas)
+
+2007-10-17 21:39 +0000 [r86202] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Changing the strategy field of the call_queue
+ struct to be signed instead of unsigned, since the code attempts
+ to set the strategy to -1 if you specify a bogus strategy. While
+ this isn't a huge issue in 1.4, it could be a problem for someone
+ who, say, tries to use the roundrobin strategy in trunk (despite
+ all the deprecation warnings in 1.4).
+
+2007-10-17 17:57 +0000 [r86149] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: If Asterisk is in the middle of shutting
+ down, respond to OPTIONS with 503 Unavailable. (closes issue
+ #10994) Reported by: eserra Patches: sip-options-503.patch
+ uploaded by eserra (license 45)
+
+2007-10-17 16:58 +0000 [r86117] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Whoops, forgot to remove the original
+ sip_scheddestroy. (closes issue #11010) Reported by: vadim
+
+2007-10-17 15:23 +0000 [r86066] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: When runuser/rungroup is specified, a remote
+ console could only be attained by root (Closes issue #9999)
+
+2007-10-17 15:06 +0000 [r86063] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Don't schedule dialog destruction if a
+ MESSAGE is received using an existing dialog. (closes issue
+ #11010) Reported by: vadim
+
+2007-10-16 23:35 +0000 [r86028-86032] Mark Michelson <mmichelson at digium.com>
+
+ * configs/queues.conf.sample: Since monitor-join is deprecated now,
+ remove the example from the sample queues.conf file
+
[... 12549 lines stripped ...]
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