[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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@@ -1,0 +1,13373 @@
+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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