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2009-01-06  Leif Madsen <lmadsen@digium.com>

	* Asterisk 1.2.31 released

2009-01-06 20:44 +0000 [r167259]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Security fix AST-2009-001.

2008-12-10  Tilghman Lesher <tlesher@digium.com>

	* Asterisk 1.2.30.4 released

2008-12-10 21:06 +0000 [r162868]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Fix for AST-2008-012

2008-12-05 20:50 +0000 [r161421]  Sean Bright <sean.bright@gmail.com>

	* include/asterisk/astobj2.h, astobj2.c: Fix build errors on
	  FreeBSD (uint -> unsigned int). (closes issue #14006) Reported
	  by: alphaque Patches: astobj2.h-patch uploaded by alphaque
	  (license 259) (Slightly modified by seanbright)

2008-12-01  Tilghman Lesher <tlesher@digium.com>

	* Asterisk 1.2.30.3 released

2008-11-25 21:37 +0000 [r159245]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Regression fix for last security fix. Set
	  the iseqno correctly. (closes issue #13918) Reported by:
	  ffloimair Patches: 20081119__bug13918.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: ffloimair

2008-08-09  Tilghman Lesher <tlesher@digium.com>

	* Asterisk 1.2.30.2 released

2008-08-09 15:24 +0000 [r136945]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/compat.h, include/asterisk/astobj2.h: Regression
	  fixes for Solaris

2008-07-25 15:00 +0000 [r133577]  Russell Bryant <russell@digium.com>

	* LICENSE: Fix the IAX2 URI for calling Digium

2008-07-23  Tilghman Lesher <tlesher@digium.com>

	* Asterisk 1.2.30.1 released

2008-07-24 03:46 +0000 [r133360]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: This part was not correctly patched for
	  AST-2008-010.

2008-07-22  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.30 released

2008-07-22 21:14 +0000 [r132711]  Tilghman Lesher <tlesher@digium.com>

	* configs/iax.conf.sample, channels/chan_iax2.c: Fixes for
	  AST-2008-010 and AST-2008-011

2008-06-03  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.29 released

2008-06-03 19:30 +0000 [r120109]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Copy the From header into a variable so that
	  pedantic SIP handling does not try to mess with a NULL pointer.
	  (AST-2008-008) (closes issue #12607) Reported by: hooi

2008-05-30 12:49 +0000 [r119008-119237]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: - Instead of only enforcing destination
	  call number checking on an ACK, check all full frames except for
	  PING and LAGRQ, which may be sent by older versions too quickly
	  to contain the destination call number. (As suggested by Tim
	  Panton on the asterisk-dev list) - Merge changes from
	  team/russell/iax2-frame-race, which prevents PING and LAGRQ from
	  being sent before the destination call number is known.

	* channels/chan_iax2.c: Merge changes from
	  team/russell/iax2-another-fix-to-the-fix As described in the
	  following post to the asterisk-dev mailing list, only enforce
	  destination call numbers when processing an ACK.
	  http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html

2008-05-21  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.28.1 released

2008-05-08 19:14 +0000 [r115564]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a race condition that bbryant just
	  found while doing some IAX2 testing. He was running Asterisk
	  trunk running IAX2 calls through a few Asterisk boxes, however,
	  the audio was extremely choppy. We looked at a packet trace and
	  saw a storm of INVAL and VNAK frames being sent from one box to
	  another. It turned out that what had happened was that one box
	  tried to send a CONTROL frame before the 3 way handshake had
	  completed. So, that frame did not include the destination call
	  number, because it didn't have it yet. Part of our recent work
	  for security issues included an additional check to ensure that
	  frames that are supposed to include the destination call number
	  have the correct one. This caused the frame to be rejected with
	  an INVAL. The frame would get retransmitted for forever, rejected
	  every time ... This race condition exists in all versions that
	  got the security changes, in theory. However, it is really only
	  likely that this would cause a problem in Asterisk trunk. There
	  was a control frame being sent (SRCUPDATE) at the _very_
	  beginning of the call, which does not exist in 1.2 or 1.4.
	  However, I am fixing all versions that could potentially be
	  affected by the introduced race condition. These changes are what
	  bbryant and I came up with to fix the issue. Instead of simply
	  dropping control frames that get sent before the handshake is
	  complete, the code attempts to wait a little while, since in most
	  cases, the handshake will complete very quickly. If it doesn't
	  complete after yielding for a little while, then the frame gets
	  dropped.

2008-05-07 16:22 +0000 [r115511]  Russell Bryant <russell@digium.com>

	* include/asterisk/dlinkedlists.h (removed), channels/chan_iax2.c:
	  Remove remnants of dlinkedlists. I didn't actually use them in
	  the final version of my IAX2 improvements.

2008-05-06 19:54 +0000 [r115421]  Jason Parker <jparker@digium.com>

	* contrib/scripts/get_ilbc_source.sh: read requires an argument on
	  some non-bash shells (closes issue #12593) Reported by: bkruse
	  Patches: getilbc.sh_12593_v1.diff uploaded by bkruse (license
	  132)

2008-05-05 17:53 +0000 [r115296]  Russell Bryant <russell@digium.com>

	* Makefile, include/asterisk/astobj2.h (added), astobj2.c (added),
	  include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
	  Merge changes from team/russell/iax2_find_callno_1.2 These
	  changes address a critical performance issue introduced in the
	  latest release. The fix for the latest security issue included a
	  change that made Asterisk randomly choose call numbers to make
	  them more difficult to guess by attackers. However, due to some
	  inefficient (this is by far, an understatement) code, when
	  Asterisk chose high call numbers, chan_iax2 became unusable after
	  just a small number of calls. On a small embedded platform, it
	  would not be able to handle a single call. On my Intel Core 2 Duo
	  @ 2.33 GHz, I couldn't run more than about 16 IAX2 channels.
	  Ouch. These changes address some performance issues of the
	  find_callno() function that have bothered me for a very long
	  time. On every incoming media frame, it iterated through every
	  possible call number trying to find a matching active call. This
	  involved a mutex lock and unlock for each call number checked.
	  So, if the random call number chosen was 20000, then every media
	  frame would cause 20000 locks and unlocks. Previously, this
	  problem was not as obvious since Asterisk always chose the lowest
	  call number it could. A second container for IAX2 pvt structs has
	  been added. It is an astobj2 hash table. When we know the remote
	  side's call number, the pvt goes into the hash table with a hash
	  value of the remote side's call number. Then, lookups for
	  incoming media frames are a very fast hash lookup instead of an
	  absolutely insane array traversal. In a quick test, I was able to
	  get more than 3600% more IAX2 channels on my machine with these
	  changes.

2008-04-29 12:52 +0000 [r114822]  Kevin P. Fleming <kpfleming@digium.com>

	* contrib/scripts/get_ilbc_source.sh: stop script from appending
	  source code if run multiple times

2008-04-22  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.28 released

2008-04-22 22:20 +0000 [r114561]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: When we receive a full frame that is
	  supposed to contain our call number, ensure that it has the
	  correct one. (closes issue #10078) (AST-2008-006)

2008-03-26 19:49 +0000 [r110869-111125]  Kevin P. Fleming <kpfleming@digium.com>

	* UPGRADE.txt: update UPGRADE notes to document usage of the script

	* contrib/scripts/get_ilbc_source.sh (added), codecs/ilbc: add a
	  script to make getting the iLBC source code simple for end users

	* codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/packing.h
	  (removed), codecs/ilbc/getCBvec.c (removed),
	  codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
	  (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
	  (removed), codecs/ilbc/getCBvec.h (removed),
	  codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
	  (removed), codecs/ilbc/FrameClassify.c (removed),
	  codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
	  codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
	  (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
	  (removed), codecs/ilbc/anaFilter.c (removed),
	  codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
	  (removed), codecs/ilbc/doCPLC.h (removed),
	  codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
	  codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
	  (removed), codecs/ilbc/createCB.h (removed),
	  codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
	  (removed), codecs/ilbc/iCBSearch.c (removed),
	  codecs/ilbc/filter.c (removed), codecs/ilbc/gainquant.c
	  (removed), codecs/ilbc/hpInput.c (removed),
	  codecs/ilbc/hpOutput.c (removed), codecs/ilbc/iCBSearch.h
	  (removed), codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h
	  (removed), codecs/ilbc/gainquant.h (removed),
	  codecs/ilbc/LPCencode.c (removed), codecs/ilbc/hpOutput.h
	  (removed), codecs/ilbc/StateSearchW.c (removed),
	  codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
	  (removed), codecs/ilbc/iCBConstruct.c (removed),
	  codecs/ilbc/syntFilter.c (removed), codecs/ilbc/iCBConstruct.h
	  (removed), codecs/ilbc/syntFilter.h (removed),
	  codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
	  (removed): due to licensing restrictions, we cannot distribute
	  the source code for iLBC encoding and decoding... so remove it,
	  and add instructions on how the user can obtain it themselves

2008-03-20 21:53 +0000 [r110335]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Fix some very broken
	  code that was introduced in 1.2.26 as a part of the security fix.
	  The dnsmgr is not appropriate here. The dnsmgr takes a pointer to
	  an address structure that a background thread continuously
	  updates. However, in these cases, a stack variable was passed.
	  That means that the dnsmgr thread would be continuously writing
	  to bogus memory.

2008-03-18  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.27 released

2008-03-18 16:27 +0000 [r109488]  Terry Wilson <twilson@digium.com>

	* include/asterisk/astobj.h: Fix character string being treated as
	  format string

2008-03-18 15:08 +0000 [r109391]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Do not return with a successful
	  authentication if the From header ends up empty. (AST-2008-003)

2008-01-22  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.26.2 released

2008-01-07 20:46 +0000 [r96931]  Russell Bryant <russell@digium.com>

	* configs/extensions.conf.sample: Change misery.digium.com to
	  pbx.digium.com

2007-12-23 01:30 +0000 [r94661]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Fix for fix for security fix (third time's
	  the charm?)

2007-12-20  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.26.1 released

2007-12-20 20:21 +0000 [r94214-94255]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix another potential seg fault ... (closes
	  issue #11606) Reported by: dimas

	* channels/chan_iax2.c: Fix a couple of places where it's possible
	  to dereference a NULL pointer.

2007-12-18  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.26 released

2007-12-18 18:44 +0000 [r93667-93675]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Fixing AST-2007-027
	  (Closes issue #11119)

2007-11-29  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.25 released

2007-11-29 21:10 +0000 [r90170]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c: Properly escape src and dst fields (Fixes
	  AST-2007-026)

2007-09-13 18:10 +0000 [r82334]  Kevin P. Fleming <kpfleming@digium.com>

	* LICENSE: clarify the OpenSSL and OpenH323 license exceptions

2007-08-07  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.24 released

2007-08-07 17:44 +0000 [r78370]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Revert patch committed for issue #9660. It
	  broke E&M trunks. (closes issue #10360) (closes issue #10364)

2007-08-02 17:56 +0000 [r77942]  Steve Murphy <murf@digium.com>

	* fskmodem.c: This patch hopefully solves 10141; The user is
	  running with it, and it doesn't appear to harm asterisk's
	  operation, and may prevent a crash. I'll store it in 1.2, as we
	  have shut down support on 1.2, but since I developed the patch
	  before support finished, and it might affect 1.4 and trunk, I'm
	  going ahead with it.

2007-07-31 19:19 +0000 [r77842]  Steve Murphy <murf@digium.com>

	* contrib/scripts/ast_grab_core: This probably isn't super-general,
	  but it's a first stab at using kill -11 to generate a core file
	  instead of gcore.

2007-07-30 18:40 +0000 [r77782]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_agi.c: Revert change in revision 71656, even though it
	  fixed a bug, because many people were depending upon the (broken)
	  behavior.

2007-07-30 14:50 +0000 [r77767]  Joshua Colp <jcolp@digium.com>

	* apps/app_macro.c: (closes issue #10334) Reported by: ramonpeek
	  Pass through the return value from macro_exec through the MacroIf
	  application.

2007-07-25 00:07 +0000 [r76978]  Steve Murphy <murf@digium.com>

	* channels/chan_zap.c: this fixes bug 10293, where the error
	  message because defaultzone or loadzone was not defined was
	  confusing

2007-07-24 22:11 +0000 [r76934]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk/lock.h: Oops, res contains the error code, not
	  errno. I was wondering why a mutex was reporting "No such file or
	  directory"...

2007-07-24  Jason Parker <jparker@digium.com>

	* Asterisk 1.2.23 released

2007-07-24 16:32 +0000 [r76802]  Jason Parker <jparker@digium.com>

	* channels/chan_iax2.c: Don't create the Asterisk channel until we
	  are starting the PBX on it. (ASA-2007-018)

2007-07-23 18:28 +0000 [r76560-76653]  Joshua Colp <jcolp@digium.com>

	* channels/chan_agent.c: (closes issue #5866) Reported by: tyler Do
	  not force channel format changes when a generator is present. The
	  generator may have changed the formats itself and changing them
	  back would cause issues.

	* channels/chan_sip.c: (closes issue #10236) Reported by: homesick
	  Patches: rpid_1.4_75840.patch uploaded by homesick (license 91)
	  Accept Remote Party ID on guest calls.

2007-07-22 21:39 +0000 [r76409]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk/app.h: We should not use C++ reserved words in
	  API headers (closes issue #10266)

2007-07-21 02:01 +0000 [r76226]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Backport a fix for a memory leak that was
	  fixed in trunk in reivision 76221 by rizzo. The memory used for
	  the localaddr list was not freed during a configuration reload.

2007-07-20 17:16 +0000 [r76080]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #10247) Reported by:
	  fkasumovic Patches: chan_sip.patch uploaded by fkasumovic
	  (license #101) Drop any peer realm authentication entries when
	  reloading so multiple entries do not get added to the peer.

2007-07-19 15:49 +0000 [r75757-75927]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: When processing full frames, take sequence
	  number wraparound into account when deciding whether or not we
	  need to request retransmissions by sending a VNAK. This code
	  could cause VNAKs to be sent erroneously in some cases, and to
	  not be sent in other cases when it should have been. (closes
	  issue #10237, reported and patched by mihai)

	* channels/chan_iax2.c: When traversing the queue of frames for
	  possible retransmission after receiving a VNAK, handle sequence
	  number wraparound so that all frames that should be retransmitted
	  actually do get retransmitted. (issue #10227, reported and
	  patched by mihai)

2007-07-18 20:31 +0000 [r75748]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Store prior to copy (closes issue #10193)

2007-07-18 17:48 +0000 [r75657]  Dwayne M. Hubbard <dhubbard@digium.com>

	* apps/app_queue.c: removed the word 'pissed' from ast_log(...)
	  function call for BE-90

2007-07-17  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.22 released

2007-07-17 20:57 +0000 [r75440-75449]  Russell Bryant <russell@digium.com>

	* channels/chan_skinny.c: Properly check for the length in the
	  skinny packet to prevent an invalid memcpy. (ASA-2007-016)

	* channels/iax2-parser.h, channels/chan_iax2.c,
	  channels/iax2-parser.c: Ensure that when encoding the contents of
	  an ast_frame into an iax_frame, that the size of the destination
	  buffer is known in the iax_frame so that code won't write past
	  the end of the allocated buffer when sending outgoing frames.
	  (ASA-2007-014)

	* channels/chan_iax2.c: After parsing information elements in IAX
	  frames, set the data length to zero, so that code later on does
	  not think it has data to copy. (ASA-2007-015)

2007-07-16 20:46 +0000 [r75251-75304]  Kevin P. Fleming <kpfleming@digium.com>

	* dns.c: provide proper copyright/license attribution for this
	  structure that was copied from a BSD-licensed header file long,
	  long ago...

	* Makefile: install the LICENSE file along with the music files

	* sounds/fpm-world-mix.mp3 (removed), sounds/moh/fpm-calm-river.mp3
	  (added), Makefile, sounds/moh (added),
	  sounds/moh/fpm-world-mix.mp3 (added), sounds/moh/LICENSE (added),
	  sounds/fpm-sunshine.mp3 (removed), sounds/moh/fpm-sunshine.mp3
	  (added), sounds/fpm-calm-river.mp3 (removed): move FreePlayMusic
	  files into a subdirectory, and include a license statement for
	  them

2007-07-13 20:35 +0000 [r75107]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Fix a couple potential minor memory leaks.
	  load_moh_classes() could return without destroying the loaded
	  configuration.

2007-07-13 20:10 +0000 [r75066]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Fixed an issue where chanspy flags were
	  uninitialized if no options were passed. What triggered this
	  investigation was an IRC chat where some people's quiet flags
	  were set while others' weren't even though none of them had
	  specified the q option.

2007-07-13 20:07 +0000 [r75052-75059]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Ensure that adding a user to the list of
	  users of a specific music on hold class is not done at the same
	  time as any of the other operations on this list to prevent list
	  corruption. Using the global moh_data lock for this is not ideal,
	  but it is what is used to protect these lists everywhere else in
	  the module, and I am only changing what is necessary to fix the
	  bug.

	* channels/chan_zap.c: (closes issue #9660) Reported by: mmacvicar
	  Patches submitted by: bbryant, russell Tested by: mmacvicar,
	  marco, arcivanov, jmhunter, explidous When using a TDM400P (and
	  probably other analog cards) there was a chance that you could
	  hang up and pick the phone back up where it has been long enough
	  to be not considered a flash hook, but too soon such that the
	  device reports that it is busy and the person on the phone will
	  only hear silence. This patch makes chan_zap more tolerant of
	  this and gives the device a couple of seconds to succeed so the
	  person on the phone happily gets their dialtone.

2007-07-12 15:51 +0000 [r74814]  Joshua Colp <jcolp@digium.com>

	* res/res_musiconhold.c: Only print out a warning for situations
	  where it is actually helpful. (issue #10187 reported by denke)

2007-07-11 22:53 +0000 [r74766]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: The function make_trunk() can fail and
	  return -1 instead of a valid new call number. Fix the uses of
	  this function to handle this instead of treating it as the new
	  call number. This would cause a deadlock and memory corruption.
	  (possible cause of issue #9614 and others, patch by me)

2007-07-11 21:12 +0000 [r74719]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: The cli command "agent logoff Agent/x
	  soft" did not work...at all. Now it does. (closes issue #10178,
	  reported and patched by makoto, with slight modification for 1.4
	  and trunk by me)

2007-07-11 18:33 +0000 [r74656]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c: Make sure that the ESCAPE immediately
	  follows the condition that uses LIKE. This fixes realtime
	  extensions with ODBC. (closes issue #10175, reported by stuarth,
	  patch by me)

2007-07-11 17:15 +0000 [r74587]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, channels/Makefile: Use some Makefile magic
	  to determine if linux/compiler.h is present. (issue #10174
	  reported by francesco_r)

2007-07-10 19:57 +0000 [r74373-74427]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Fix an issue where it was possible to have a
	  service level of over 100% Between the time recalc_holdtime and
	  update_queue was called, it was possible that the call could have
	  been hungup. Move both additions to the same place, so this won't
	  happen. Issue 10158, initial patch by makoto, modified by me.

	* channels/chan_agent.c: Fix an issue with wrapuptime not working
	  when using AgentLogin. Issue 10169, patch by makoto, with a minor
	  mod by me to not re-break issue 9618

	* dns.c: Use res_ndestroy on systems that have it. Otherwise, use
	  res_nclose. This prevents a memleak on NetBSD - and possibly
	  others. Issue 10133, patch by me, reported and tested by scw

2007-07-10  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.21.1 released

2007-07-10 15:37 +0000 [r74316]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix a small typo in description in of
	  Voicemail() application. Issue 10170, patch by casper.

2007-07-10 15:30 +0000 [r74313]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c: Only use ESCAPE when LIKE is used. (issue
	  #10075, this part reported by jmls on IRC, patch by me)

2007-07-10 14:48 +0000 [r74264]  Joshua Colp <jcolp@digium.com>

	* app.c: Ensure the group information category exists before trying
	  to do a string comparison with it. (issue #10171 reported by
	  mlegas)

2007-07-09  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.21 released

2007-07-09 21:00 +0000 [r74165]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: When the specified class isn't found,
	  properly fall back to the channel's music class or the default.
	  (issue #10123, reported by blitzrage, patches from juggie, qwell,
	  and me)

2007-07-09 20:18 +0000 [r74158]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Several chan_zap options were not working on
	  reload because they were arbitrarily disallowed when reloading
	  some/most PRI options (such as signalling) was disallowed.
	  Options such as polarityonanswerdelay and answeronpolarityswitch
	  can safely be changed on a reload. This corrects that behavior.
	  Issue 9186, patch by tzafrir.

2007-07-06 23:01 +0000 [r73678-73768]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: If a sip_pvt struct has already registered
	  an extension state callback, remove the old one before adding a
	  new one. If this isn't done, Asterisk will crash. (issue #10120)

	* res/res_config_odbc.c: (closes issue #10075) Reported by: apsaras
	  Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
	  with MSSQL 2005 by explicitly stating that '\' is being used as
	  an escape character.

	* channels/chan_sip.c: (closes issue #10125) Reported by: makoto
	  Patches submitted by: makoto This fixes a crash in chan_sip that
	  happens when the bindaddr setting is not valid on Asterisk
	  startup, gets fixed, and then a reload gets issued.

2007-07-06 15:26 +0000 [r73674]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Fixed a bug wherein agents get stuck busy.
	  (issue 9618, reported by jiddings, patched by moi) closes issue
	  #9618

2007-07-05 22:11 +0000 [r73547]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: we shouldn't allow G.723.1 endpoints to use
	  VAD, just like we don't support it for G.729

2007-07-05 19:15 +0000 [r73315-73466]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Copy language information to the dialog
	  structure when calling a peer for situations where a PBX may be
	  started on the dialed channel. (issue #10121 reported by
	  clegall_proformatique)

	* apps/app_chanspy.c, channel.c: Tweak spy locking. (issue #9951
	  reported by welles)

	* channels/chan_local.c: Actually check to make sure a PBX was
	  started on one of the Local channels instead of blindly assuming
	  it was. (issue #10112 reported by makoto)

	* apps/app_queue.c: Reset ServicelevelPerf variable back to 0 if we
	  are unable to calculate it each time... otherwise we will get
	  previous values. (issue #10117 reported by noriyuki)

2007-07-04 14:50 +0000 [r73207-73252]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: bchannel configurations like
	  echocancel and volume control, need to be setuped on inbound
	  calls too.

	* channels/chan_misdn.c: bad bug in overlapdial case, we called
	  start_pbx multiple times, because the state wasn't changed..

2007-07-03 12:34 +0000 [r73052]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_dial.c: RetryDial should accept a 0 argument, but it
	  does not, because atoi does not distinguish between 0 and error
	  (closes issue #10106)

2007-07-03 08:04 +0000 [r73004]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed issue, that misdn_l2l1_check could
	  only be called from mISDN Source channels.. #9449

2007-07-02 17:58 +0000 [r72924]  Jason Parker <jparker@digium.com>

	* say.c: Fix an issue with playing "oclock" multiple times in
	  French with 24 hour time format. Issue 10101

2007-07-01 23:51 +0000 [r72805]  Russell Bryant <russell@digium.com>

	* pbx/pbx_spool.c: When appending lines to call files to keep track
	  of retries, write a leading newline just in case the original
	  call file did not have a newline at the end. This fix is in
	  response to a problem I saw reported on the asterisk-users
	  mailing list.

2007-06-29  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.20 released

2007-06-29 16:30 +0000 [r72629]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Backport changes that make chan_iax2 not
	  start the PBX on an incoming channel until the three-way call
	  setup is completed. These changes are already in 1.4 and trunk.

2007-06-29 13:08 +0000 [r72585]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: check if the
	  bchannel stack id is already used, if so don't use it a second
	  time. Also added a release_chan lock, so that the same chan_list
	  object cannot be freed twice. chan_misdn does not crash anymore
	  on heavy load with these changes.

2007-06-27 23:24 +0000 [r72378]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c: Update documentation to clarify variable
	  usage with MixMonitor. (issue #9494 reported by netoguy)

2007-06-27 23:22 +0000 [r72333-72373]  Brett Bryant <bbryant@digium.com>

	* asterisk.c: Reinstating patch. This actually fixes the problem,
	  however I was running a development branch without it and
	  mistakenly thought it wasn't fixed. Fixes issue #10010, and
	  #9654: 100% CPU usage caused by an asterisk console losing it's
	  controlling terminal.

	* asterisk.c: Reverted changes for earlier revisions 72259 to
	  72261. Issue #9654, #10010

2007-06-27 22:43 +0000 [r72327]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Fix issue where queue log events might be
	  missing. (issue #7765 reported by mtryfoss)

2007-06-27 21:06 +0000 [r72267]  Russell Bryant <russell@digium.com>

	* pbx/pbx_config.c: Fix a minor issue with parsing the priority
	  number. You could have as much whitespace as you want around a
	  numeric priority, but you couldn't have any whitespace around a
	  special priority like "n" or "hint". (issue #10039, reported by
	  mitheloc, fixed by me)

2007-06-27 20:43 +0000 [r72259]  Brett Bryant <bbryant@digium.com>

	* asterisk.c: Fixes 100% load when controlling terminal disappears.
	  Issue #9654, #10010

2007-06-27 20:23 +0000 [r72256]  Joshua Colp <jcolp@digium.com>

	* channel.c: I may possibly get shot for doing this... but... defer
	  CDR processing until after the channel has been dealt with. This
	  should eliminate all of the issues with channels going funky
	  (SIP/PRI) when you are posting CDRs to a database that is either
	  slow or unavailable and do not want to enable batching.

2007-06-27 18:40 +0000 [r72184]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix another problem in voicemail with
	  missing symbols. Issue 10074, patch by kryptolus, extended to
	  include #if 0'd blocks (just in case)

2007-06-27 13:22 +0000 [r72040-72099]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
	  simplified generation for dummy bchannels, also we mark them as
	  dummies, so they are not used later as real-bchannels, optimized
	  the RESTART mechanisms, we block a channel now on cause:44, and
	  send out a RESTART automatically, then on reception of
	  RESTART_ACKNOWLEDGE we unblock the channel again.

	* channels/misdn/isdn_lib.h, channels/misdn/isdn_lib.c: simplified
	  channel finding and locking a lot. removed unnecessary #ifdefed
	  areas.

	* channels/misdn/isdn_lib.c: isdn_lib.c didn't compile

	* channels/misdn/isdn_lib.c: for inbound TE calls, we setup the
	  bchannel when we get the CONNECT_ACKNOWLEDGE, to make sure mISDN
	  has everything ready. removed some #if 0 areas which weren't used
	  anymore.

2007-06-26 17:49 +0000 [r71847]  Jason Parker <jparker@digium.com>

	* Makefile: Don't try to install an init script that doesn't exist.
	  Reported to me on #asterisk on Freenode IRC.

2007-06-26 12:25 +0000 [r71656-71750]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Issue 10062 - Trying to move a message
	  without selecting one first results in memory corruption

	* res/res_agi.c: Issue 10035 - handle_exec returns a result
	  inconsistent with all of the other AGI commands

2007-06-25 01:02 +0000 [r71414]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Ignore other URIs after the first in a 300
	  Multiple Choice response. (issue #10041 reported by homesick)

2007-06-24 20:04 +0000 [r71358]  Russell Bryant <russell@digium.com>

	* asterisk.c: Revert the patch from issue 9654 due to an unexpected
	  side effect

2007-06-24 17:32 +0000 [r71288]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* manager.c, db.c: Issue 10043 - There is a legitimate need to be
	  able to set variables to the empty string.

2007-06-22 16:02 +0000 [r71124]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Send an unhold indication when going off
	  hold. (issue #10036 reported by speedy)

2007-06-22 14:52 +0000 [r71065]  Jason Parker <jparker@digium.com>

	* file.c, res/res_agi.c: Fix a few silly usages of ast_playstream()
	  - it only ever returns 0... Issue 10035

2007-06-22 14:39 +0000 [r71064]  Brett Bryant <bbryant@digium.com>

	* asterisk.c: Fixed infinite loop when controlling terminal was
	  lost and return value of input function wasn't checked for
	  errors. This would cause 100% cpu to be taken up. (closes issue
	  #9654, issue #10010) Reported by: mnicholson, and eserra Idea for
	  the patch from mnicholson, patched by me

2007-06-21 22:29 +0000 [r70948]  Steve Murphy <murf@digium.com>

	* cdr.c: This little fix is in response to bug 10016, but may not
	  cure it. The code is wrong, clearly. In a situation where you set
	  the CDR's amaflags, and then ForkCDR, and then set the new CDR's
	  amaflags to some other value, you will see that all CDRs have had
	  their amaflags changed. This is not good. So I fixed it.

2007-06-21 21:37 +0000 [r70898]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Don't explode if the gain option is
	  specified without a value. (issue #9274 reported by mfarver)

2007-06-21 19:13 +0000 [r70804]  Steve Murphy <murf@digium.com>

	* cdr/cdr_custom.c: it was pointed out that the cdr_custom config
	  load could get a lock, and under certain circumstances, would
	  never release it. I also noted that the situation where more than
	  one mapping spec was warned about, but did not ignore further
	  mappings as it had promised. I think I have fixed both
	  situations.

2007-06-21 13:11 +0000 [r70672]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: we activate the
	  bchannels in TE mode on incoming calls only when we want to
	  connect the call.

2007-06-20 22:20 +0000 [r70551]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't overwrite the configured username
	  setting upon a REGISTER. (issue #8565 reported by jsmith)

2007-06-20 19:25 +0000 [r70444]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_dial.c: Issue 9997 - Timelimit times out the wrong
	  channel

2007-06-20 18:45 +0000 [r70396]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Fix a problem where an established call
	  would not be properly disconnected when a PRI disconnect is
	  received depending on which cause code was received. (issue
	  #9588, original patch by softins, updated patch from jtexter3,
	  and some additional feedback from mhardeman)

2007-06-20 15:42 +0000 [r70311-70342]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: forgot one place ..

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/isdn_lib.c: fixed a bug that was introduced by
	  copy and paste in the last commit ..bchannels weren't cleaned
	  properly.

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/isdn_lib.c: on receiption of cause:44 we mark the
	  channel as in use and inform the user about the situation, we
	  need to test the RESTART stuff then. Also shuffled the
	  empty_chan_in_stack function after the bchannel cleaning
	  functions, to avoid race conditions.

2007-06-19 18:07 +0000 [r70053]  Steve Murphy <murf@digium.com>

	* channel.c: This fixes 9246, where channel variables are not
	  available in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
	  consolidate the channel variables during a masquerade, and then
	  copy the merged variables back onto the clone, so the zombie has
	  the same vars that the 'original' has.

2007-06-19 17:00 +0000 [r69992]  Joshua Colp <jcolp@digium.com>

	* rtp.c: Handle the CC field in the RTP header. (issue #9384
	  reported by DoodleHu)

2007-06-19 16:45 +0000 [r69990]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Backport fix for crashes related to
	  subscriptions from 1.4 ... Fix a crash that could occur when
	  handing device state changes. When the state of a device changes,
	  the device state thread tells the extension state handling code
	  that it changed. Then, the extension state code calls the
	  callback in chan_sip so that it can update subscriptions to that
	  extension. A pointer to a sip_pvt structure is passed to this
	  function as the call which needs a NOTIFY sent. However, there
	  was no locking done to ensure that the pvt struct didn't
	  disappear during this process. (issue #9946, reported by
	  tdonahue, patch by me, patch updated to trunk to use the sip_pvt
	  lock wrappers by eliel)

2007-06-19 16:21 +0000 [r69894-69986]  Joshua Colp <jcolp@digium.com>

	* channel.c: Update BRIDGEPEER variable if set to the new channel
	  name when a masquerade happens. (issue #9699 reported by dimas)

	* apps/app_meetme.c: Perform an extra hangup check just in case.
	  (issue #9589 reported by bcnit)

2007-06-19 13:23 +0000 [r69887]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: when we send out a SETUP, but get no
	  response, we should cleanup everything after reception of a
	  hangup.

2007-06-19 12:57 +0000 [r69765-69846]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Add parked call extension AFTER the parking
	  slot has been announced, otherwise two threads will try to handle
	  the same channel and it will go kaboom. (issue #9191 reported by
	  japple)

	* channels/chan_sip.c: Set the peer name on the dialog to the one
	  configured in sip.conf and NOT the username to be used for
	  authentication attempts. (issue #9967 reported by achauvin)

2007-06-18 17:45 +0000 [r69743]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/scripts/safe_asterisk: Issue 9998 - Remove SIG prefix,
	  since it's not supported by ksh

2007-06-15  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.19 released

2007-06-14 23:21 +0000 [r69469]  Jason Parker <jparker@digium.com>

	* config.c: Fix an issue where the line number in an unterminated
	  comment block error message would show the wrong line number.
	  "Reported" to me on #asterisk (somebody posted an error message,
	  and I happened to catch it)

2007-06-14 20:56 +0000 [r69347]  Russell Bryant <russell@digium.com>

	* channel.c: Backport rev 69010 from the 1.4 branch ... In
	  ast_channel_make_compatible(), just return if the channels' read
	  and write formats already match up. There are code paths that
	  call this function on a pair of channels multiple times. This
	  made calls fail that were using g729 in some cases. The reason is
	  that codec_g729a will unregister itself from the list of
	  available translators will all licenses are in use. So, the first
	  time the function got called, the right translation path was
	  allocated. However, the second time it got called, the code would
	  not find a translation path to/from g729 and make the call fail,
	  even if the channel actually already had a g729 translation path
	  allocated. (SPD-32)

2007-06-14 15:15 +0000 [r69258]  Jason Parker <jparker@digium.com>

	* funcs/func_groupcount.c: Change a quite broken while loop to a
	  for loop, so "continue;" works as expected instead of eating 99%
	  CPU... Issue 9966, patch by me.

2007-06-13 18:12 +0000 [r69127]  Joshua Colp <jcolp@digium.com>

	* app.c: Return group counting to previous behavior where you could
	  only have one group per category. (issue #9711 reported by
	  irroot)

2007-06-13 09:55 +0000 [r69053]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_msg_parser.c: restart indicator 0x80 is
	  correct, at least that's what libpri does.

2007-06-12 14:18 +0000 [r68921]  Joshua Colp <jcolp@digium.com>

	* rtp.c: Bring RTP back to Asterisk at the end of a native bridge
	  no matter what.

2007-06-12 08:35 +0000 [r68732-68887]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: if the bridged partner is mISDN too we
	  should not send dtmf tones, they are transmitted inband always

	* channels/chan_misdn.c: if we have already some digits, we just
	  stop the tones.

	* channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
	  check for NULL Pointer when calling misdn_new. Asterisk does not
	  allow us to create channels anymore when stop gracefully is used
	  :). also modified the restart_indicator to 0

2007-06-11 14:29 +0000 [r68682]  Joshua Colp <jcolp@digium.com>

	* channel.c: Improve deadlock handling of the channel list. (issue
	  #8376 reported by one47)

2007-06-11 09:18 +0000 [r68631]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
	  channels/misdn/isdn_lib.c: fixed problem that the dummybc chanels
	  had no lock, checking for the lock now. Also fixed the channel
	  restart stuff, we can now specify and restart particular channels
	  too.

2007-06-08 22:22 +0000 [r68526]  Russell Bryant <russell@digium.com>

	* apps/app_dictate.c: Don't automatically hang up after running
	  Dictate so that callers can exit cleanly using '#' (closes issue
	  #9577, patch from Thomas Andrews)

2007-06-08 00:15 +0000 [r68368-68397]  Joshua Colp <jcolp@digium.com>

	* say.c: Don't call ast_waitstream_full when the control file
	  descriptor and audio file descriptor are not set, simply call
	  ast_waitstream! (issue #8530 reported by rickead2000)

	* dnsmgr.c: Do a DNS lookup immediately upon calling the dnsmgr
	  function, don't wait until a refresh happens. (issue #9097
	  reported by plack)

2007-06-07 23:13 +0000 [r68351]  Russell Bryant <russell@digium.com>

	* say.c: Fix a problem where saying a character wouldn't properly
	  break out when the caller pressed '#' (issue #8113, reported by
	  patbaker82, patch from jamesgolovich (hey, long time no see!) and
	  patbaker82)

2007-06-07 20:02 +0000 [r68204]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Don't try to save voicemail greetings
	  unless the user presses '1' to accept/save. Issue 9904, patch by
	  me.

2007-06-07 14:19 +0000 [r67938-68070]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Allow the 'g' option to work if used with the
	  'S' option. (issue #9888 reported by gasparz)

	* channels/chan_sip.c: Only notify the devicestate system of a peer
	  state change when the peer is built from the config file. (issue
	  #9900 reported by arkadia)

2007-06-06 16:40 +0000 [r67715]  Russell Bryant <russell@digium.com>

	* channel.c: We have some bug reports showing crashes due to a
	  double free of a channel. Add a sanity check to
	  ast_channel_free() to make sure we don't go on trying to free a
	  channel that wasn't found in the channel list. (issue #8850, and
	  others...)

2007-06-06 13:28 +0000 [r67593-67649]  Joshua Colp <jcolp@digium.com>

	* rtp.c: Reinvite the RTP back to the Asterisk machine when the
	  timeout happens. (issue #9888 reported by gasparz)

	* devicestate.c: Revert channel name splitting fix for Zap. The
	  moral of the story is don't use - in your user/peer names. (issue
	  #9668 reported by stevedavies)

2007-06-05 15:42 +0000 [r67306-67307]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/chan_misdn_config.h: briding is a bool, fixed copy
	  and paste issue.

	* channels/chan_misdn.c: simplified the EVENT_SETUP handling in the
	  cb_events function a lot. Commented the different possibilities a
	  bit and made functions of shared code. When the dialed extension
	  does not exist in the extensions.conf we'll jump into the 'i'
	  extension if this does exist, else we disconnect the call with
	  the cause:1 = No Route to Destination.

2007-06-05 11:18 +0000 [r67239]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn_config.c, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h: Backport of the overlap_dial
	  functionality from asterisk-1.4's chan_misdn.

2007-06-05 10:05 +0000 [r67209]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn_config.c, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h: added possibility to
	  deactivate bridging per port

2007-06-04 23:41 +0000 [r67060-67161]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_math.c: According to MATH, 0+1181000386 = 1181000448.
	  Oops.

	* contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.mandrake.asterisk,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.mandrake.zaptel,
	  contrib/init.d/rc.slackware.asterisk: Add revision Id tags (by
	  request of tzafrir)

2007-05-31 16:12 +0000 [r66764]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: It is now possible for this path of
	  execution to have the frame pointer be NULL, therefore we need to
	  check for it before trying to access it. (issue #9836 reported by
	  barthpbx)

2007-05-31 15:58 +0000 [r66744]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_macro.c: Issue 9818 - Fix for issue 8329 breaks
	  pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime,
	  but only because we lack core API to do it.

2007-05-29 21:49 +0000 [r66537]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_strings.c: If the value of a variable passed to
	  FIELDQTY is blank, then FIELDQTY should return 0, not 1.

2007-05-29 07:53 +0000 [r66349]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #9802 - Change inuse counter on CANCEL

2007-05-25 13:46 +0000 [r66127]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* say.c: Issue 9791 - Fix pronunciation of seconds in Dutch

2007-05-24 14:40 +0000 [r65837]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Allow RFC2833 to be negotiated when an
	  INVITE comes in without SDP and is not matched to a user or peer.
	  (issue #9546 reported by mcrawford)

2007-05-24 09:19 +0000 [r65767]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: we should only activate the generator in
	  chan_misdn, when asterisk hask not yet taken the call
	  (WAITING4DIGS state). Alerting audio will be generated fomr
	  asterisk for example.

2007-05-23 20:46 +0000 [r65676-65682]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: ensure that variables are set on a newly
	  created channel before we start a PBX on it

	* channels/chan_iax2.c: if we are going to set variables on a newly
	  created channel, it should be done *before* we start the PBX on
	  it

2007-05-23 13:06 +0000 [r65389-65588]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Revert revision 62417 as someone reported
	  problems with it to Mark. This was related to issue #9588.

	* apps/app_queue.c: Fix a memory leak that I just noticed in the
	  device state handling in app_queue. On most device state changes,
	  it would leak roughly 8 to 64 bytes (the length of the name of
	  the device).

2007-05-22 07:46 +0000 [r65328]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: we stop the tones only when we're in the
	  pre-call phase, otherwise e.g. when in CONNECTED state we should
	  not stop tones when we receive an Information Message

2007-05-18 20:56 +0000 [r65172]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, include/asterisk/cdr.h, cdr.c: This update will
	  fix the situation that occurs as described by 9717, where when
	  several targets are specified for a dial, if any one them reports
	  FAIL, the whole call gets FAIL, even though others were ringing
	  OK. I rearranged the priorities, so that a new disposition, NULL,
	  is at the lowest level, and the disposition get init'd to NULL.
	  Then, next up is FAIL, and next up is BUSY, then NOANSWER, then
	  ANSWERED. All the related set routines will only do so if the
	  disposition value to be set to is greater than what's already
	  there. This gives the intended effect. So, if all the targets are
	  busy, you'd get BUSY for the call disposition. If all get BUSY,
	  but one, and that one rings is not answered, you get NOANSWER. If
	  by some freak of nature, the NULL value doesn't get overridden,
	  then the disp2str routine will report NOANSWER as before.

2007-05-18 18:10 +0000 [r65075-65122]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Not getting an ACK to a 200 OK in the
	  initial invite is critical to the call.

	* channels/chan_sip.c: Issue 9235 - part of the problem, maybe not
	  all. Please retry with this patch (and no other patch) if you
	  have problems with hanging SIP channels. Thank you. A special
	  Thank You to WeBRainstorm that gave me access to his system.

2007-05-18 11:23 +0000 [r64902-65007]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: fixed a
	  warning regarding Keypad encoding. encode the IE sending_complete
	  at the right position.

	* channels/chan_misdn.c: we *need* to send a PROCEEDING when
	  sending_complete is set, even if need_more_infos is requested.

2007-05-17 21:14 +0000 [r64819]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk/linkedlists.h: How is it that we never caught
	  that this is returning the opposite of our documentation, until
	  now?

2007-05-17 16:52 +0000 [r64758]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: If we have a negative current message, we
	  shouldn't go back even further... Issue 9727.

2007-05-16 10:55 +0000 [r64514-64603]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Fixing possible bug in auth of BYE

	* channels/chan_sip.c: Support SIP uri's starting with SIP: and
	  sip: (reported by Tony Mountfield on the mailing list. Thanks!)

	* channels/chan_sip.c: Issue #9726 - rlister - Better logging for
	  ACL denials While at it, also added better logging and handling
	  of peers that are not supposed to register. My patch, stole the
	  issue report from Russell. My apologies, Russell :-)

2007-05-16 08:23 +0000 [r64513]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: in the case immediate=yes, we directly
	  jump into the dialplan, where people can use PlayTones to
	  indicate a Dialtone, so we don't need to to that by ourself. also
	  we should not do a dialtone_indicate for incoming calls on a TE
	  port in overlapdialmode.

2007-05-14 18:34 +0000 [r64275]  Joshua Colp <jcolp@digium.com>

	* devicestate.c: Only perform stripping of - strings from the
	  channel name for Zap channels. Anywhere else we might remove a
	  legitimate part of a device name. (issue #9668 reported by
	  stevedavies)

2007-05-11 16:31 +0000 [r63903]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* Makefile, contrib/scripts/safe_asterisk: Issue 9121 - fixups for
	  safe_asterisk script

2007-05-10 23:14 +0000 [r63828]  Jason Parker <jparker@digium.com>

	* channels/chan_iax2.c: Fix an issue with trying to kill a thread
	  before it gets created. Issue 9709, patch by nic_bellamy.

2007-05-10 20:38 +0000 [r63748]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Do not allocate SIP pvt's for PEERs we can
	  not reach. This was seen as a lot of dialogs being created then
	  immediately destroyed at reload/restart of the SIP channel.

2007-05-09 17:20 +0000 [r63653]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we only create a DSP if it's
	  requested on SUB_REAL

2007-05-09 16:51 +0000 [r63610]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Properly handle hints that point to multiple
	  devices in chan_sip. Why chan_sip is even doing this I have no
	  idea but I would rather not go into a rant. (issue #9536 reported
	  by rlister)

2007-05-09 14:48 +0000 [r63565]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_directory.c: Replicate fix from 51158 (app_voicemail) to
	  app_directory (Issue 9224)

2007-05-09 11:26 +0000 [r63519]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: release_chan frees ch, so we should never
	  touch ch after release_chan, this may cause segfaults.

2007-05-08 22:19 +0000 [r63477]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_macro.c: Issue 9602 - segfault in app_macro

2007-05-08 15:07 +0000 [r63402]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: added
	  application misdn_check_l2l1 which tries to pull up the L1/L2 on
	  all ports that have the layers down in a group. It waits then for
	  a timeout. This helps for scenarios where multiple PMP BRIs are
	  grouped together, or where a provider has a faulty PTP
	  Implementation, that looses the L2 after a while.

2007-05-08 06:20 +0000 [r63359]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Issue 9527 - upon entering a folder, no
	  message is selected (curmsg == -1), so deleting causes memory
	  corruption (beyond bounds)

2007-05-07 21:39 +0000 [r63285]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/app.h, app.c, channel.c: Properly handle what
	  happens during a masquerade in relation to group counting. (issue
	  #9657 reported by ramonpeek)

2007-05-07 14:06 +0000 [r63195]  Olle Johansson <oej@edvina.net>

	* config.c: Releasing the whole in-memory configuration while
	  you're adding to it is not a good thing. Please review this
	  change. Caused by additions (+) to non-existing contexts.

2007-05-03 16:42 +0000 [r62987]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When a peer is seeded or built tell the
	  devicestate core to update it's status. This is easier then
	  having chan_sip load before pbx_config. (issue #9658 reported by
	  dlynes)

2007-05-03 15:39 +0000 [r62885-62945]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: when
	  we're in state WAITING4DIGS, we use the asterisk tone-generator
	  which prods us, so we can't just return -1 in misdn_write in this
	  case. Added a MISDN_KEYPAD channel variable, and fixed the
	  sending of keypad. this enables us to modify the call forward
	  parameters in the switch.

	* channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: fixed
	  the problem that misdn_write did not return -1 when called with 0
	  samples in a frame this resultet in a deadlock in some
	  circumstances, when the call ended because of a busy extension.
	  added encoding of keypad.

2007-05-03 00:23 +0000 [r62796-62841]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_config_odbc.c: doh... initializing the pointer variable
	  will work just a bit better

	* res/res_config_odbc.c: increase reliability and efficiency of
	  static Realtime config loading via ODBC: don't request fields we
	  aren't going to use don't request sorting on fields that are
	  pointless to sort on explicitly request the fields we want,
	  because we can't expect the database to always return them in the
	  order they were created (reported by blitzrage in person (!),
	  patch by me)

2007-05-02 20:10 +0000 [r62737]  Steve Murphy <murf@digium.com>

	* cdr.c, pbx.c: Some tweaks to satisfy CDR bug 8796, where being in
	  'h' extension louses up the dst field

2007-05-02 17:38 +0000 [r62691]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_iax2.c: Issue 9638 - if a text frame is sent with
	  no terminating NULL through a bridged IAX connection, the remote
	  end will receive garbage characters tacked onto the end.

2007-05-01 21:55 +0000 [r62496-62547]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Remove an unnecessary check that makes it so
	  if you hang up after doing an attended transfer before the target
	  extension answers the channel, the transfer is not successful.
	  (issue #9338, patch by svanlund)

	* configs/indications.conf.sample: Add indications.conf information
	  for the Philippines. (issue #9525, reported and patched by
	  loloski)

2007-04-30 15:57 +0000 [r62417]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: This patch fixes an issue where depending on
	  the cause code, when the network sends a PRI disconnect, the call
	  may not be properly hung up. (issue #9588, reported and patched
	  by softins)

2007-04-30 14:34 +0000 [r62368]  Joshua Colp <jcolp@digium.com>

	* asterisk.c: Update copyright notice. It's now the year 2007!

2007-04-27 16:16 +0000 [r62173]  Jason Parker <jparker@digium.com>

	* codecs/codec_zap.c: This transcoder message needn't be a NOTICE.
	  I've seen it cause confusion more than a few times.

2007-04-27 13:57 +0000 [r62126]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #7351 - SIP Cancel fails due to the
	  wrong contact uri. Reported by PPYY, failed to fix by OEJ final
	  fix by wojtekka - THANKS!!!! THis was a hard one to catch.

2007-04-26 16:30 +0000 [r61958-62037]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Revert previous fix for when the IAX2
	  channel goes funky (that's the technical term). This is causing
	  legit calls to be prematurely hung up. (issue #9600 reported by
	  justdave)

	* config.c: Don't count failed include attempts against the
	  configuration include level. (issue #9593 reported by mostyn)

2007-04-25 22:24 +0000 [r61913]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: handle a very bizarre race condition with
	  channels being redirected before a simple switch can be started
	  on them (issue #9286)

2007-04-25 21:55 +0000 [r61862-61866]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: If the callerid= option is specified, but
	  empty, clear any previous data.

	* channels/chan_iax2.c: Ensure that callerid settings are reset on
	  a reload.

2007-04-25 18:52 +0000 [r61804]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/app.h, funcs/func_groupcount.c, app.c,
	  apps/app_groupcount.c, channel.c: Merge rewritten group counting
	  support. No more storing data on the variable list of the
	  channels. That was bad, mmmk? (issue #7497 reported by sabbathbh)

2007-04-25 16:20 +0000 [r61798]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Fix a typo where cid_num got copied instead
	  of cid_ani. (issue #9587, reported and patched by xrg)

2007-04-24  Russell Bryant <russell@digium.com>

	* Asterisk 1.2.18 released

2007-04-24 21:33 +0000 [r61786]  Russell Bryant <russell@digium.com>

	* manager.c: Don't crash if a manager connection provides a
	  username that exists in manager.conf but does not have a
	  password, and also requests MD5 authentication. (ASA-2007-012)

2007-04-24 18:20 +0000 [r61776-61777]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_phone.c, channels/chan_zap.c,
	  channels/chan_modem.c: removed #if 0 block from chan_phone,
	  chan_zap, and chan_modem restart_monitor()

	* channels/chan_modem.c: removed pthread_join in restart_monitor()
	  to make it like chan_zap and other channel drivers

2007-04-24 16:05 +0000 [r61771]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Allow RFC2833 to be sent in the response SDP
	  when an INVITE comes in without SDP. (issue #9546 reported by
	  mcrawford)

2007-04-24 13:50 +0000 [r61770]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
	  channels/misdn/isdn_lib.c: added lock for sending messages to
	  avoid double sending. shuffled some empty_chans after the
	  cb_event calls, this avoids that a release_complete from a quite
	  different call releases a fresh created setup by accident.

2007-04-20 21:14 +0000 [r61692-61704]  Jason Parker <jparker@digium.com>

	* loader.c: Fix an issue that I noticed while looking over issue
	  9571. The reload timestamp was getting set after reloading the
	  built-in stuff, and before the modules.

	* apps/app_queue.c: If the '* to hangup' option is not enabled, we
	  don't need to disable * as a valid exit key. If it was enabled,
	  this statement would've never been checked in the first place.
	  Issue #9552

2007-04-19 07:56 +0000 [r61685]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Send NOTIFY to Contact: in SUBSCRIBE - as
	  reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp.

2007-04-19 02:30 +0000 [r61680]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_language.c, funcs/func_moh.c, funcs/func_groupcount.c,
	  funcs/func_timeout.c, funcs/func_cdr.c, funcs/func_callerid.c:
	  Bug 9557 - Specifying the GetVar AMI action without a Channel
	  parameter can cause Asterisk to crash. The reason this needs to
	  be fixed in the functions instead of in AMI is because Channel
	  can legitimately be NULL, such as when retrieving global
	  variables.

2007-04-16 14:08 +0000 [r61663]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't stop RTP on errors on INFO messages.
	  Disclaimer: This patch was needed for Edvina AstHoloApp and was
	  meant to be included in 1.2, but never made it in time so I felt
	  I could add it now. No, just joking, patching error found while
	  testing T.140 with Omnitor earlier this spring.

2007-04-13 19:15 +0000 [r61655]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Add OUTBOUND_GROUP_ONCE variable to app_dial.
	  This behaves the same as OUTBOUND_GROUP except it will get unset
	  after use so it won't get accidentally inherited. (issue #BE-140)

2007-04-11 16:01 +0000 [r61376-61476]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: If someone sets the "useragent" option in
	  sip.conf to be empty, then don't add the User-Agent header at
	  all. It is an optional header, anyway. Also, the bug report says
	  that some of Japan's SIP providers don't allow it for some weird
	  reason. (issue #9488, reported by makoto, fixed by me)

	* channels/chan_sip.c: Fix a bug with switching between
	  host=dynamic and using specific hosts for peers. The code would
	  only reset the peer's address when it is dynamic if it was a new
	  peer structure. Now, it will also reset the address if it was
	  already in the peer list, but before the reload, it was not
	  dynamic. (issue #9515, reported by caio1982, fixed by me)

	* channels/chan_sip.c: Remove the attempt at reporting
	  configuration errors in sip.conf. This can cause a bunch of
	  improper messages when using realtime. I give up. As oej tried to
	  convince me when I put this in, there is just no easy way to do
	  it. (inspired by a message on the -dev list)

2007-04-11 12:05 +0000 [r61357]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: some fixes for
	  PMP Hold/Retrieve, it should work now, when briding=no

2007-04-10 12:31 +0000 [r61170]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn_config.c: msns config parameter defaults to '*'

2007-04-09 19:38 +0000 [r61038]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Don't send ActionID before Response:
	  header. - Don't use a blank in an AMI header

2007-04-09 17:22 +0000 [r60935]  Jason Parker <jparker@digium.com>

	* apps/app_directory.c: Allow matching on names shorter than 3
	  chars. This also fixes the case where somebody wants to match on
	  less then 3 chars. Issue 9071

2007-04-09 02:49 +0000 [r60846-60849]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk.h, asterisk.c: Don't check for error when
	  lowering priority (according to the manpage, it should never
	  happen anyway). It might could happen, though, if another thread
	  messed with the priority, so safeguard against that (reported via
	  -dev list).

	* channels/chan_local.c: Bug 9505 - If the return value for
	  local_queue_frame is set, then p->lock is no longer valid.

2007-04-09 00:59 +0000 [r60797]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: When calling a device that then forwards us
	  elsewhere... we have to make our channels compatible if it is the
	  only channel being dialed. (issue #9445 reported by
	  marcelbarbulescu)

2007-04-08 14:00 +0000 [r60660-60711]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_macro.c: Gosub called within a Macro resets the
	  arguments improperly and causes general weirdness. (Issue 8329)

	* file.c: Bug 9486 - memory leak when opening a filestream

2007-04-06 17:03 +0000 [r60456]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: There should only be one code path for doing
	  DTMF conditionals on channels. This fixes it.

2007-04-06 14:41 +0000 [r60398]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: remove undocumented 'cardsmode' parameter and
	  stop searching for transcoders during reload()

2007-04-05 16:09 +0000 [r60267]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Just because we can't find the voicemail
	  configuration file, doesn't mean that the module failed to load.
	  The user could be using realtime. Issue #9473

2007-04-05 12:52 +0000 [r60213]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only unlock our pvt and net locks if we are
	  actually going to try to lock the owner again. (issue #9472
	  reported by zoa)

2007-04-04 17:38 +0000 [r59886-60134]  Russell Bryant <russell@digium.com>

	* manager.c: It is valid to redirect channels via the manager
	  interface that are not in the UP state. Instead of checking for
	  that to prevent to ensure a dead channel doesn't get redirected,
	  just use the ast_check_hangup() API call. (issue #9457, reported
	  by Callmewind, patch by me) (related to issue #8977)

	* channels/chan_sip.c: Fix the return value of
	  handle_common_options() so that it always properly indicates
	  whether it handled the option or not. (issue #9455, reported by
	  Netview, fixed by me)

	* channels/chan_sip.c: Add a missing "\r\n" in the body of the
	  NOTIFY that is sent to indicate the status of a transfer. (issue
	  #9388, reported by rarritt)

	* mkpkgconfig: Use the more generic check for "sed -r" support that
	  was already present in 1.4. (related to issue #9399)

	* mkpkgconfig: On Darwin, the -r argument to sed is not valid. It
	  has to be -E. (issue #9399, reported by jcovert)

	* channels/chan_sip.c: Don't attempt to report configuration errors
	  in build_user(). oej pointed out that for a "friend" entry, this
	  won't work, because all user options are valid for peers, but not
	  the other way around.

	* channels/chan_sip.c: Make chan_sip report when it encounters an
	  unknown option. (issue #9440, reported by nightcrawler)

	* app.c: When doing a built-in blind or attended transfer, restore
	  the ability to use '#' to terminate the number and immediately do
	  the transfer instead of having to dial the number and just wait
	  for the feature digit timeout. (issue #8366, xueliangliang)

2007-04-03 10:40 +0000 [r59788-59803]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn_config.c: ptp is the 5th bit, not the 4th.

	* channels/misdn_config.c, channels/misdn/chan_misdn_config.h: Use
	  the new sysfs way of mISDN 1.2 to check if a port is NT or not.

2007-04-02 18:55 +0000 [r59723]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Increase the maximum size for a string of
	  mailboxes to 1024. (issue #9270 reported by rtucker)

2007-04-02 12:08 +0000 [r59623-59639]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn_config.c, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h: added option which allows us
	  to accept incoming SETUP Messages without automatically sending
	  Proceeding or Setup Acknowledge, this is useful with some broken
	  switches and if you want to Release incoming calls without
	  previously having acknowledged them. The new option is
	  noautorespond_on_setup=yes|no default is no, so we don't break
	  the existing behaviour

	* channels/chan_misdn.c: don't be verbose if no need

	* channels/misdn/isdn_lib.c: we can now make 30 channels on a PRI
	  (before we forgot chan 31..)

2007-04-01 22:35 +0000 [r59608]  Russell Bryant <russell@digium.com>

	* netsock.c: Add the SO_REUSEADDR flag to sockets handled by
	  netsock. This is needed by the patch that went in for issue 7874.
	  chan_iax2 needs to be able to create socket that is lisetning on
	  INADDR_ANY, but also be able to bind sockets to specific
	  addresses. (Thanks to Stevenson on the asterisk-dev mailing list
	  for explaining why this flag was needed.)

2007-03-29 17:33 +0000 [r59360]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Keep a global array of variables indicating
	  whether certain conference rooms are in use. This ensures that
	  two people going into a new dynamic conference when the 'e'
	  option is set don't go into the same conference room. (issue
	  #8835 reported by eliel)

2007-03-29 17:14 +0000 [r59355-59357]  Russell Bryant <russell@digium.com>

	* rtp.c: If an error occurs when reading from an RTP socket, and
	  the error code does not indicate that we should try again, then
	  return NULL instead of a "null frame". This will prevent Asterisk
	  from trying over and over again, and eventually causing the
	  system to crash. (issue #8285, john)

	* channels/chan_iax2.c: Backport the change to chan_iax2 to return
	  NULL instead of a "null frame" from its read callback. See
	  revision 59341 to the 1.4 branch for more info.

2007-03-29 16:04 +0000 [r59299-59301]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* cdr/cdr_odbc.c: Issue 9415 - No point to getting a diagnostic
	  field if we aren't doing anything with the information. (Plus, it
	  tends to crash the Postgres ODBC driver.)

	* res/res_odbc.c: Change ENV section to use setenv, instead of
	  putenv (Alexandru Pirvulescu <sigxcpu@gmail.com>, reported via
	  -dev list)

2007-03-27 23:36 +0000 [r59280-59283]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Oops

	* apps/app_voicemail.c: Fix a few remaining bad mmap(2) return
	  values

2007-03-27 23:19 +0000 [r59258-59277]  Russell Bryant <russell@digium.com>

	* apps/app_directory.c: Fix the check of the return value from
	  mmap(). Thanks to Corydon for catching this one.

	* channels/chan_iax2.c: Fix the use of the "sourceaddress" option
	  when "bindaddr" is set to 0.0.0.0 instead of having each
	  interface explicitly listed. (issue #7874, patch by stevens)

2007-03-27 13:56 +0000 [r59252]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed #9355

2007-03-26 10:21 +0000 [r59199]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn_config.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, channels/misdn/Makefile,
	  channels/misdn/chan_misdn_config.h, channels/Makefile,
	  channels/misdn/isdn_lib.c: mISDN >= 1.2 provides a dsp pipeline
	  for i.e. echo cancellation modules, make chan_misdn use it.

2007-03-24 01:35 +0000 [r59194]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only try to handle a response if it has a
	  response code. (ASA-2007-011)

2007-03-23 16:07 +0000 [r59186-59187]  Steve Murphy <murf@digium.com>

	* apps/app_macro.c: Ugh. that was dumb. Fixed an error.

	* apps/app_macro.c: Added a few words in the Macro doc strings
	  about the behavior of macros with hangups (et al.), as per 9337

2007-03-21 18:03 +0000 [r59086]  Joshua Colp <jcolp@digium.com>

	* res/res_monitor.c: Indicate the filename changed when it is
	  changed. (issue #9311 reported by jsmith)

2007-03-20 09:23 +0000 [r59062-59063]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: modified a loglevel

	* channels/chan_misdn.c: avoid sending a disconnect when we already
	  received one.

2007-03-19  Jason Parker <jparker@digium.com>

	* Asterisk 1.2.17 released

2007-03-14 16:38 +0000 [r58896]  Russell Bryant <russell@digium.com>

	* SECURITY: Add a note to the security file that the Asterisk CLI
	  and log files may contain sensitive information, and that people
	  should keep this in mind.

2007-03-13 12:58 +0000 [r58849-58850]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed the crypt_keys stuff

	* channels/misdn_config.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
	  channels/misdn/isdn_lib.c: added method standard_dec for dialing
	  out on groups, to avoid conflicts, which caused issues with some
	  ISDN providers

2007-03-13 11:45 +0000 [r58847]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #9229 - No port in request URI on
	  register to non default SIP ports (neelakantan)

2007-03-12 16:49 +0000 [r58832]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/lock.h: We can't use the assembler version of
	  fetchadd_int under Intel Macs. (issue #9254 reported by darrell
	  budic)

2007-03-09 20:46 +0000 [r58579]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If we are unable to lookup the host in a c
	  line we have to abort, otherwise the previous data is gone and we
	  will (potentially) have no data when all is said and done.

2007-03-09 14:43 +0000 [r58558]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: we can free channel 31 as well, since
	  we can occupy it

2007-03-08 16:04 +0000 [r58388]  Joshua Colp <jcolp@digium.com>

	* dsp.c: Only print out debug message if the definition that makes
	  the variables shows up was actually defined. (issue #9233
	  reported by serginuez)

2007-03-07 18:17 +0000 [r58164-58242]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a problem where the Asterisk channel
	  name could be that of the wrong IAX2 user for a call. This is
	  because the first step of choosing this name is to look for an
	  IAX2 peer that happens to have the same IP/port number that this
	  call is coming from and assuming that is it. However, this is not
	  always correct. So, I have made it change this name after
	  authentication happens since at that point, we have an exact
	  match.

	* manager.c: Fix a misplaced block of code in the 1.2 version of
	  the patch to fix issue #8977

	* manager.c: If the channels acquired using the manager Redirect
	  action are not up, then don't attempt to do anything with them.
	  It could lead to weird behavior, including crashes. (issue #8977)

2007-03-06 22:52 +0000 [r58115]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: Fix for 9220: Eyebeam cannot renew
	  subscriptions for presence info. Reason: re-SUBSCRIBE requests
	  don't include Accept headers, which the rfc says are optional (to
	  put it tersely), (it uses MAY), and luckily, the sip_pvt struct
	  has the format info stored, so we simply leave it if the format
	  is set, and the accept header null.

2007-03-06 20:33 +0000 [r58052]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Change error message to proper message

2007-03-06 15:17 +0000 [r58008]  BJ Weschke <bweschke@btwtech.com>

	* channels/h323/Makefile: Cleanup the Makefile so that we only
	  attempt to include a file when we're building and not 'clean'ing
	  so 'make clean' completes successfully. chan_h323 maintainer:
	  please check to make sure I haven't broken your build target.
	  From: jsmith in #asterisk-dev

2007-03-05 23:18 +0000 [r57962]  Christian Richter <christian.richter@beronet.com>

	* channels/Makefile: subdirs like misdn and h323 should be cleaned
	  by the clean target from channels/Makefile as well

2007-03-05 17:49 +0000 [r57869]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Make create_dirpath use our standard for
	  return values. -1 is failure, 0 is success. (issue #9205 reported
	  by ballares)

2007-03-05 14:53 +0000 [r57825]  Steve Murphy <murf@digium.com>

	* pbx.c: Fixed a typo introduced via 9156

2007-03-04 10:39 +0000 [r57753]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed another
	  place where the out_cause was hardcoded to 16

2007-03-03 06:36 +0000 [r57648]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Memory leak of a list, if call recording
	  was abandoned

2007-03-02 Russell Bryant <russell@digium.com>

	* Asterisk 1.2.16 released

2007-03-02 18:32 +0000 [r57523]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: fixed typo

2007-03-02 17:02 +0000 [r57475]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If a SIP message comes in and goes to a
	  method handler that requires additional values that may not be
	  present then send back an error.

2007-03-02 16:39 +0000 [r57458]  Steve Murphy <murf@digium.com>

	* pbx.c: further refinement in wording of goto documentation, as
	  per 9156, goto not proceeding to next instruction

2007-03-01 22:19 +0000 [r57317]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: Don't even attempt to optimize things when
	  a proxy channel is involved. It will just explode in weird and
	  unexplaineable ways. (issue #9175 reported by
	  clegall_proformatique)

2007-02-28 19:12 +0000 [r57118]  Steve Murphy <murf@digium.com>

	* pbx.c: a small documentation update, to reflect reality in the
	  goto doc strings, as per 9156, Goto does not proceed to next prio
	  if jump fails

2007-02-28 18:55 +0000 [r57092]  Joshua Colp <jcolp@digium.com>

	* channels/chan_agent.c: Fix a few more issues with the agent
	  logoff CLI command. (issue #9123 reported by arbrandes)

2007-02-28 16:09 +0000 [r57034]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed
	  bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302,
	  #303, #304

2007-02-26 20:05 +0000 [r56850]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_config.c: Obey the clearglobalvars option in extensions
	  reload (or dialplan reload depending on your version). (issue
	  #9146 reported by ramonpeek)

2007-02-26 00:34 +0000 [r56729]  Russell Bryant <russell@digium.com>

	* utils.c: Ensure that lock.h is included in utils.c with
	  AST_API_MODULE defined so that the implementations will be
	  properly included when the AST_INLINE_API functions are not going
	  to be inlined. (issue #9124, festr)

2007-02-25 14:38 +0000 [r56684]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channel.c: Issue 9130 - If prev is the last item on the channel
	  list, then evaluating additional conditions (e.g. name prefix)
	  will cause a NULL dereference.

2007-02-23 23:20 +0000 [r56406-56504]  Russell Bryant <russell@digium.com>

	* asterisk.c: Fix up a couple more signal handlers to not do bad
	  things that could cause various undesirable results. The other
	  day, I made Asterisk deadlock by hitting Control-C because of a
	  bad signal handler. Now, signal handlers just set a flag and
	  write to an alert pipe for the flag to be handled. Then, there is
	  another thread that is monitoring for these flags. If being run
	  in console mode, it is just the main thread. If Asterisk is in
	  the background, a thread is created to do it.

	* channels/chan_iax2.c: Don't destroy mutexes before unregistering
	  all of the entry points from the core. Also, fix a potential
	  memory leak from not destroying the locks for all of the possible
	  call numbers (about 32k of them).

2007-02-22 23:19 +0000 [r56230-56279]  Joshua Colp <jcolp@digium.com>

	* channels/chan_agent.c: Always defer Agent logoff if any channels
	  are up until they hang up. (issue #9123 reported by arbrandes)

	* channels/chan_sip.c, channel.c: Only change the original or clone
	  channel if it's the channel behind the proxy channel, not if it's
	  just a regular bridged channel.

2007-02-22 00:53 +0000 [r56010]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: If we receive a frame that is not in any of
	  the negotiated formats, then drop it. (potentially issue #8781
	  and SPD-12)

2007-02-21 20:32 +0000 [r55956]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Change naughty warning message to provide
	  useful information. If a write now fails on a channel in meetme
	  it will tell you the channel name instead of spitting out the
	  wrong error message.

2007-02-21 14:03 +0000 [r55868]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/make_svn_branch_name: use new tag version script

2007-02-21 00:19 +0000 [r55750]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c, utils.c, include/asterisk/lock.h: Fix random
	  crashes when using the MeetMe application. This patch converts
	  list handling to use the linked list macros and most importantly,
	  implements reference counting on the ast_conference objects. The
	  reference counting was first backported from 1.4. However, that
	  code has some problems that caused the reference count to never
	  hit zero. Those problems are fixed in this patch and will be
	  resolved in 1.4 and trunk next, with a different patch. (issues
	  #7647, #9073, #9106, BE-115).

2007-02-20 22:39 +0000 [r55669]  Joshua Colp <jcolp@digium.com>

	* channels/chan_agent.c: Defer clearing callback information if
	  channels are up until they are hung up. This ensures the hangup
	  process goes smoothly and no channels get hung in limbo. (issue
	  #8088 reported by kebl0155)

2007-02-20 19:49 +0000 [r55588]  Russell Bryant <russell@digium.com>

	* apps/app_page.c: Convert a tab to spaces so that the
	  documentation is printed out properly aligned.

2007-02-19 18:09 +0000 [r55434]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: forcename and forcegreetings options should
	  check to see if the recording already exists

2007-02-18 12:32 +0000 [r55249-55277]  Olle Johansson <oej@edvina.net>

	* apps/app_record.c: Documentation update (#9053, jsmith)

	* channels/chan_sip.c: Issue #9020 - SIP message retransmission
	  time too short. Backporting fix implemented in 1.4, where we have
	  a minimum level for the T1 timer.

2007-02-17 03:53 +0000 [r55073-55153]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Answer the channel before recording privacy
	  information. (issue #8926 reported by lmamane)

	* channels/chan_sip.c: Allow chan_sip to handle attended transfers
	  from a SIP phone that is sitting behind chan_agent. Yes folks,
	  all it took was one line of code. (issue #8784 reported by
	  pzieba)

2007-02-16 22:48 +0000 [r55005]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c, configs/meetme.conf.sample: Revert the change
	  I did in revisions 54955, 54969, and 54970, in 1.2, 1.4, and
	  trunk. I decided that once a conference is created from
	  meetme.conf, it is acceptable behavior that the pin can not be
	  changed until the conference goes away. I also added a note in
	  meetme.conf to describe this behavior. We still have another
	  issue in 1.4 and trunk where some conferences with no users don't
	  go away. That is the real bug that needs to be addressed here.

2007-02-16 22:13 +0000 [r54999]  Joshua Colp <jcolp@digium.com>

	* channels/chan_agent.c: Do not send indications through
	  ast_indicate in chan_agent but instead go directly to the
	  technology. This way when indications are emulated they happen on
	  the Agent channel and do not screw up formats on the channels.
	  (issue #8439 reported by punkgode)

2007-02-16 20:56 +0000 [r54955]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: For conferences that are configured in
	  meetme.conf, check the configuration file every time someone
	  joins the conference instead of only when the conference is first
	  created. This is to ensure that changes to the pin numbers in the
	  config file are always honored. (issue #9073)

2007-02-16 11:38 +0000 [r54771]  Olle Johansson <oej@edvina.net>

	* res/res_agi.c: Issue #9069 - If we open with TH we should not
	  close with /TD. (seanbright)

2007-02-15 16:14 +0000 [r54622]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Use a separate variable to indicate execution
	  should continue instead of the return value. (issue #8842
	  reported by pluto70)

2007-02-14 18:40 +0000 [r54438]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c:

2007-02-14 16:25 +0000 [r54373]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: When handling glare on a PRI, move the
	  requested channel rather than hang up the old one. Fix for 8957
	  and 9011.

2007-02-13 18:35 +0000 [r54179]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Make sure that outbound calls are applied to
	  the peer. This fixes some issues with "hints not working", but
	  only in 1.2.

2007-02-08 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.15 released

2007-02-08 22:17 +0000 [r53658]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: ensure channelcount is cleared before we
	  enumerate transcoders, so 'reload' doesn't double the channel
	  count

2007-02-08 13:36 +0000 [r53529]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Issue 9003 - If fullname is empty, quote()
	  passes back "\""

2007-02-07 15:38 +0000 [r53357]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Fix a few potential memory leaks with
	  realtime users and peers. (issue #8999 reported by bsmithurst)

2007-02-07 15:30 +0000 [r53354]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_macro.c: Issue 7440 - Macro called from Macro from the h
	  extension exits prematurely

2007-02-06 06:58 +0000 [r53245]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* manager.c: Issue 8987 - Status could return two responses
	  (mnicholson)

2007-02-06 00:08 +0000 [r53224]  Jason Parker <jparker@digium.com>

	* configs/dnsmgr.conf.sample: Add a proper newline at the end of
	  this sample config file.

2007-02-03 20:39 +0000 [r53133-53134]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c, utils.c, include/asterisk/lock.h: Revert some
	  changes that accidentally got committed as a part of another fix.

	* apps/app_dial.c, apps/app_meetme.c, utils.c,
	  include/asterisk/lock.h: set the DIALSTATUS variable to contain
	  "INVALIDARGS" when the dial application exits early because of
	  invalid arguments instead of just leaving it empty. (issue #8975)

2007-02-02 16:58 +0000 [r53117]  Joshua Colp <jcolp@digium.com>

	* config.c: Pass the glob expanded filename to process_text_line so
	  that error messages contain the actual filename, not the original
	  include one. (issue #8959 reported by tzafrir)

2007-02-01 23:14 +0000 [r53107]  Jason Parker <jparker@digium.com>

	* apps/app_chanspy.c: Fix a small typo. Synopsis lines shouldn't
	  have a newline

2007-02-01 22:21 +0000 [r53095-53103]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Copy noncodeccapability over to the joint
	  variable so that telephone-event will get transmitted in the sent
	  INVITE.

	* channels/chan_sip.c: Don't negotiate RFC2833 when not configured
	  to do so. (issue #8799 reported by mdu113)

2007-02-01 21:12 +0000 [r53090]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Make sure we release call from call
	  counter before we destroy call (maybe #7744 and more) -
	  Backported by accident from 1.4

2007-02-01 21:03 +0000 [r53084]  Joshua Colp <jcolp@digium.com>

	* res/res_musiconhold.c: Return previous behavior of having MOH
	  pick up where it was left off. (issue #8672 reported by
	  sinistermidget)

2007-02-01 20:07 +0000 [r53069-53074]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_strings.c: Bug 8965

	* funcs/func_strings.c, pbx.c: No wonder FIELDQTY doesn't work with
	  functions... the documentation in pbx.c was wrong

2007-01-31 21:25 +0000 [r53045]  Russell Bryant <russell@digium.com>

	* channels/chan_mgcp.c, manager.c, channels/chan_zap.c,
	  pbx/pbx_spool.c, apps/app_meetme.c, apps/app_page.c,
	  channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_queue.c,
	  channels/chan_iax2.c, apps/app_rpt.c, cdr.c, pbx.c: Fix a bunch
	  of places where pthread_attr_init() was called, but
	  pthread_attr_destroy() was not.

2007-01-31 18:58 +0000 [r53044]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: update to match modified transcoder API

2007-01-31 17:41 +0000 [r53039]  Russell Bryant <russell@digium.com>

	* rtp.c: Use the proper format string to print unsigned values in
	  the rtp debug output. (issue #8954, wmis)

2007-01-31 17:28 +0000 [r53034]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/Makefile: allow codec_zap to build again, now that
	  transcoder support is in zaptel 1.2

2007-01-30 19:41 +0000 [r52857-52954]  Russell Bryant <russell@digium.com>

	* channel.c: Don't print a message indicating that we don't know
	  what to do with a proceeding control frame in
	  ast_request_and_dial(). We just need to ignore it. (reported by
	  JerJer on #asterisk-dev)

	* asterisk.c: The SIGHUP handler was implemented to allow admins to
	  send SIGHUP to a running Asterisk process to reload the
	  configuration. However, doing the actual reload in the signal
	  handler itself is a very bad thing to do, because the reload
	  process includes calling non-reentrant functions such as
	  malloc/calloc/etc. If Asterisk is running in the background, then
	  the reload will happen immediately. However, if running in
	  console mode, the reload doesn't work until something is typed at
	  the console. That sort of defeats the purpose, but I don't see an
	  easy way to get around it at this point.

	* codecs/Makefile: Comment out the parts in the Makefile that make
	  codec_zap get built. It will not yet build against zaptel 1.2, so
	  I am disabling it to prevent further bug reports until it gets
	  merged. (issue #8940)

2007-01-30 14:38 +0000 [r52843]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed some
	  possible segfaults. also fixed an very important bug which occurs
	  on high load (when calls are very fast generated)

2007-01-30 00:15 +0000 [r52762]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix the extraction of the timestamp from
	  video frames. It was using the mapping for a mini-frame instead
	  of a video-frame, which caused it to get invalid data. (issue
	  #8795, mihai)

2007-01-29 23:39 +0000 [r52716]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c: Now that filename is part of the structure
	  and since it comes before postprocess... we have to add it to our
	  postprocess line. (reported on asterisk-dev by Boris Bakchiev)

2007-01-29 16:48 +0000 [r52503]  Jason Parker <jparker@digium.com>

	* codecs/codec_zap.c: Use the correct zaptel header file location.
	  Currently, this will not build - transcoder support will be added
	  to zaptel later today.

2007-01-27 02:09 +0000 [r52360-52415]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Make COMPLETECALLER and COMPLETEAGENT output to
	  queue_log follow documentation. (issue #7677 reported by amilcar)

	* channels/chan_iax2.c: Make the last context entry read in the
	  dominant one. (issue #8918 reported by pj)

2007-01-25 19:15 +0000 [r52162-52264]  Joshua Colp <jcolp@digium.com>

	* jitterbuf.c: Allow dequeueing of frames with negative timestamp
	  by moving jitterbuffer frames check to jb_next. (issue #8546
	  reported by harmen)

	* apps/app_mixmonitor.c: Add another note about audio files being
	  played back to each bridged party. (issue #8718 reported by ppyy)

2007-01-25 00:39 +0000 [r52137]  Russell Bryant <russell@digium.com>

	* apps/app_groupcount.c: Fix a seg fault when running this
	  application with no arguments from AGI. (issue #8905, junky)

2007-01-24 17:43 +0000 [r52002]  Steve Murphy <murf@digium.com>

	* utils/check_expr.c, utils/Makefile: updated check_expr via 8322
	  (refactoring of expression checking impl); elfring contributed a
	  nice code reorg, I contributed some time to get it working again,
	  better messages

2007-01-24 10:48 +0000 [r51966]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed the busy problem (dialstatus was not
	  busy when we called a busy extension)

2007-01-24 00:57 +0000 [r51828-51843]  Russell Bryant <russell@digium.com>

	* channel.c: Fix an issue related to synchronization of recordings
	  when using Monitor(). The bug is a miscalculation of the amount
	  to seek the stream for writing to disk when the number of samples
	  coming in and out of a channel do not match up. (issue #8298,
	  #8887, report and patch by guillecabeza, patch files created and
	  testing done by whoiswes)

	* apps/app_while.c: Don't set a new value for the END_ variable on
	  the channel before using the old value. If you do, it will lead
	  to accessing a memory address that has been free()'d. (issue
	  #8895, arkadia)

2007-01-23 01:41 +0000 [r51512]  Joshua Colp <jcolp@digium.com>

	* res/res_musiconhold.c: Yield before reading from zaptel timing
	  source under Solaris so that other threads get a chance to do
	  things. (issue #7875 reported by bob)

2007-01-22 19:39 +0000 [r51410]  Russell Bryant <russell@digium.com>

	* codecs/Makefile, codecs/codec_zap.c (added): Merge codec_zap
	  support for the transcoder card. This is a standalone codec
	  module so it will not affect anything else.

2007-01-22 19:08 +0000 [r51359-51406]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c: Move filestream creation to Mixmonitor
	  loop. This will prevent a blank file from being created if no
	  frames ever pass through to be recorded. (issue #7589 reported by
	  steve_mcneil)

	* channels/chan_h323.c: Explicitly declare what codecs are
	  supported by default globally since using a bitmask for all may
	  include ones we don't need. (issue #8357 reported by
	  gknispel_proformatique)

2007-01-19 16:44 +0000 [r51300]  Russell Bryant <russell@digium.com>

	* asterisk.c: Fix a memory leak on command line tab completion. The
	  container for the matches was freed, but the individual matches
	  themselves were not. (issue #8851, arkadia)

2007-01-18 23:47 +0000 [r51271]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_zap.c: issue 7877: chan_zap module reload does not
	  use default/initialized values on subsequent loads. Reset
	  configuration variables to default values prior to parsing
	  configuration file.

2007-01-18 23:35 +0000 [r51269]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: support echo cancellers that can handle 64ms
	  or 128ms of echo cancellation

2007-01-18 21:11 +0000 [r51235-51255]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* stdtime/localtime.c: If a timezone is not specified, assume
	  localtime (instead of gmtime) (Issue #7748)

	* contrib/scripts/vmdb.sql: Document all the fields, including the
	  indication that "uniqueid" should not be renamed.

2007-01-17 21:17 +0000 [r51197]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Move the check for a failure of
	  ast_channel_alloc() to before locking the pvt structure again.
	  Otherwise, on a failure, this will cause a deadlock.

2007-01-17 20:52 +0000 [r51158-51194]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* utils.c: When ast_strip_quoted was called with a zero-length
	  string, it would treat a NULL as if it were the quoting character
	  (and would thus return the string in memory immediately following
	  the passed-in string).

	* doc/voicemail_odbc_postgresql.txt (added): Add documentation
	  walkthrough on getting Postgres to work with voicemail (from
	  Issue 8513)

	* apps/app_voicemail.c: Postgres driver doesn't like a NULL pointer
	  when retrieving the length (Bug 8513)

2007-01-16 17:36 +0000 [r51085-51145]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Return previous behavior. ParkedCalls will be
	  able to do DTMF based transfers again. trunk however will get an
	  option to allow this to be set on/off. (issue #8804 reported by
	  nortex)

	* channels/chan_zap.c: Add none as a valid callgroup/pickupgroup
	  option. I consider it a bug that it would inherit it all the way
	  down and not have any way to reset it to nothing - so that's why
	  it is in 1.2. (issue #8296 reported by gkloepfer)

2007-01-15 23:09 +0000 [r50987]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_groupcount.c: Check return value before dereferencing
	  (Bug 8822)

2007-01-15 20:44 +0000 [r50946]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c: Solves issue with forwarding voicemails
	  from folders other than inbox. patch by anthonyl.

2007-01-14 05:01 +0000 [r50781]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* db1-ast/recno/rec_utils.c, db1-ast/hash/hash_bigkey.c,
	  db1-ast/hash/hsearch.c, db1-ast/recno/rec_open.c,
	  db1-ast/recno/rec_delete.c, db1-ast/btree/bt_page.c,
	  db1-ast/hash/hash_buf.c, db1-ast/hash/hash_page.c,
	  db1-ast/recno/rec_close.c, db1-ast/recno/rec_search.c,
	  db1-ast/btree/bt_get.c, db1-ast/hash/hash.c,
	  db1-ast/recno/rec_put.c, db1-ast/include/ndbm.h, db1-ast/db/db.c,
	  db1-ast/btree/bt_debug.c, db1-ast/mpool/mpool.c,
	  db1-ast/btree/bt_seq.c, db1-ast/recno/rec_get.c,
	  db1-ast/btree/bt_split.c, db1-ast/hash/hash_func.c,
	  db1-ast/btree/bt_utils.c, db1-ast/btree/bt_open.c,
	  db1-ast/recno/rec_seq.c, db1-ast/btree/bt_delete.c,
	  db1-ast/btree/bt_overflow.c, db1-ast/hash/hash_log2.c,
	  db1-ast/btree/bt_search.c, db1-ast/btree/bt_conv.c,
	  db1-ast/btree/bt_close.c, db1-ast/btree/bt_put.c: Bug 8814 - db
	  should look for its header using a relative path, instead of the
	  system path (Fixes FreeWRT)

2007-01-12 14:34 +0000 [r50561]  Kevin P. Fleming <kpfleming@digium.com>

	* pbx.c: minor documentation clarification

2007-01-11 18:11 +0000 [r50517]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #8793 bad response for Unsupported
	  Extension (different fix).

2007-01-11 14:45 +0000 [r50495-50506]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: when we get L2 UP, the L1 is UP
	  definitely too, so we set the L1 state up as well.

	* channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c,
	  channels/misdn/isdn_lib.c: * more additions to make the RESTART
	  message work * added fix for misdn_call to allow SETUPs with
	  empty extensions, replaced the strtok_r functions with strsep for
	  that (inspired by Sandro Cappellazzo, thanks)

2007-01-10 09:51 +0000 [r50335]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/fac.c, channels/misdn/ie.c,
	  channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: more
	  fixes regarding warnings for gcc-4 and first additions for the
	  restart Information element, in the first step we initiate a
	  restart with a CLI command

2007-01-10 04:51 +0000 [r50295]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Add another return value to dial_exec_full that
	  indicates execution is going to continuing at a new
	  extension/context/priority and to just let it slide. (issue #8598
	  reported by jon)

2007-01-10 02:16 +0000 [r50227]  Russell Bryant <russell@digium.com>

	* Makefile: Make the number that represents the major version
	  number a single digit instead of 2. Using two digits makes it an
	  octal number when put into version.h, which breaks the
	  compilation of any out of tree module that checks the version for
	  any version after 1.2.7 (reported by Matteo Brancaleoni on the
	  asterisk-dev mailing list, who gave credit to vihai for pointing
	  it out)

2007-01-09 13:30 +0000 [r50150]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: The advent of realtime has enabled people
	  to use commas in the fullname field. This could cause an issue
	  with sending voicemails, when the field is unquoted. (Issue 8595)

2007-01-08 08:37 +0000 [r49922]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/misdn/ie.c,
	  channels/misdn/isdn_lib.c: make gcc 4 happy, remove some warnings

2007-01-08 05:10 +0000 [r49889]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Ensure we use the default refresh value of
	  60 if the remote server does not send one. (issue #8746 reported
	  by maethor)

2007-01-07 21:43 +0000 [r49833]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_dictate.c: If openstream fails, then we crash (Issue
	  8564)

2007-01-05 16:56 +0000 [r49635]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_iax2.c: ensure that threads which are supposed to
	  be detached (because we aren't going to wait on them) are created
	  properly

2007-01-04 17:45 +0000 [r49354-49447]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c: converted a lot of 256 to PATH_MAX and some
	  white space fixes.

	* apps/app_voicemail.c: good catch russell sorry i missed that. fix
	  magic number with proper sizeof

	* apps/app_voicemail.c: When using ODBC_STORAGE VoicemailMain
	  doesn't create the subdirectories for a mailbox such as the INBOX
	  directory. this patch solves that problem, was written by anthony
	  be-125

2007-01-03 08:24 +0000 [r49135-49303]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: * Added check
	  for bridging in misdn_call to avoid setting echocancellation when
	  2 mISDN channels are involved and when bridging is set. That lead
	  to a kernel panic before under different situations, because we
	  switched about 2 times between hardware bridging and
	  echocancelation * readded MISDN_URATE variable which got lost
	  before, this should make app_v110 work again * fixed typo

	* channels/misdn_config.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, configs/misdn.conf.sample,
	  channels/misdn/isdn_lib.c: added check for channel ranges in the
	  set/empty channel functions. set pmp_l1_check default to no.
	  added misdn restart pid cli command. added cleaning of channel
	  when we send a RELEASE_COMPLETE.

2006-12-29 00:32 +0000 [r49045]  Kevin P. Fleming <kpfleming@digium.com>

	* BUGS: location of the bug posting guidelines has changed

2006-12-27 15:43 +0000 [r48974]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 8596: Set CAN_BYE flag for 100 trying
	  too

2006-12-25 05:19 +0000 [r48939-48955]  Russell Bryant <russell@digium.com>

	* funcs/func_math.c: Fix an error introduced by copying and pasting
	  the handling of the >= operator for the MATH function. If a
	  single equal sign was used as an operator, the function would
	  treat it is as if it were the >= operator. Now, it properly
	  handles it as an invalid operator. (issue #8665, patch by
	  tempest1)

	* channels/chan_iax2.c: Check for the proper return value on an
	  error in a call to mmap(). This was reported by Andy Wang on the
	  asterisk-dev list. Thanks!

	* channels/chan_sip.c: Remove a couple of misplaced dots in log
	  messages. This was reported by Andrea Spadaccini on the
	  asterisk-dev mailing list.

2006-12-21 20:25 +0000 [r48782]  Joshua Colp <jcolp@digium.com>

	* redhat/asterisk.spec: Add new silence sound files to the spec for
	  Redhat. (issue #8652 reported by alvaro_palma_aste)

2006-12-19 21:10 +0000 [r48584]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Free localuser structure when we fail to dial
	  (issue #8612 reported by rizzo)

2006-12-19 13:08 +0000 [r48576]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: when we reject a channel, because it's
	  in use already, we shouldn't process the setup anymore. made the
	  channel allocation a bit easier and more understandable, removed
	  a few unused lines

2006-12-18 10:19 +0000 [r48552]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: when our PTP
	  Partner sends us a SETUP with a preselected channel we just
	  accept it, even when we're NT. added some checks for segfaults.

2006-12-15 10:51 +0000 [r48484]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #8592 - handle 504 as 503 - congestion

2006-12-14 13:03 +0000 [r48467]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: removed FIXUP
	  state. added check for channel allocation conflict when we create
	  a setup while the other site creates a setup on the same channel,
	  besides the check we resolve this conflict.

2006-12-14 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.14 released

2006-12-13 04:23 +0000 [r48434]  Steve Murphy <murf@digium.com>
		
	* channel.c: This small patch fixes bug 8541, where the L option to
	  the Dial app wasn't working right. A similar bug (8386) was filed
	  and fixed earlier, but an intervening bug fix to a DTMF problem
	  broke the L() code in a different way. Hopefully, everything is
	  happy now.
	
2006-12-12 05:11 +0000 [r48403]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/silence (added), sounds/silence/1.gsm (added),
	  sounds/silence/10.gsm (added), sounds/silence/2.gsm (added),
	  sounds/silence/3.gsm (added), sounds/silence/4.gsm (added),
	  sounds/silence/5.gsm (added), sounds/silence/6.gsm (added),
	  sounds/silence/7.gsm (added), sounds/silence/8.gsm (added),
	  sounds/silence/9.gsm (added): add silence files

2006-12-11 23:00 +0000 [r48394-48398]  Matt O'Gorman <mogorman@digium.com>

	* Makefile, apps/app_externalivr.c, sounds.txt: app_externalivr
	  needs a real silence file, and additional changes to add silence
	  files into core instead of extra patch provided by bug 8177 with
	  minor additions.

2006-12-11 00:33 +0000 [r48374]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_nbscat.c, apps/app_festival.c, apps/app_mp3.c,
	  res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
	  apps/app_ices.c, res/res_musiconhold.c: When doing a fork() and
	  exec(), two problems existed (Issue 8086): 1) Ignored signals
	  stayed ignored after the exec(). 2) Signals could possibly fire
	  between the fork() and exec(), causing Asterisk signal handlers
	  within the child to execute, which caused nasty race conditions.

2006-12-10 02:14 +0000 [r48371]  Steve Murphy <murf@digium.com>

	* channels/chan_zap.c: This version applies the patch suggested by
	  stevens in bug 7836 (make inbound channel RINGING state
	  consistent with other channels).

2006-12-09 15:45 +0000 [r48361]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Use locking when accessing the
	  registrations list. This list is not actually used very often, so
	  the likelihood of there being a problem is pretty small, but
	  still possible. For example, if the CLI command to list the
	  registrations was called at the same time that a reload was
	  occurring and the registrations list was getting destroyed and
	  rebuilt, a crash could occur.

2006-12-07 18:14 +0000 [r48356]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Ensure that the file position is not
	  incremented beyond the total number of files available for
	  playback. (issue #8539, ulogic)

2006-12-06 16:05 +0000 [r48322]  Russell Bryant <russell@digium.com>

	* configs/iax.conf.sample: Fix the name of the rtignoreregexpire
	  option in the sample configuration file. (issue #8526, arkadia)

2006-12-06 15:48 +0000 [r48321]  Christian Richter <christian.richter@beronet.com>

	* doc/README.misdn, channels/chan_misdn.c,
	  channels/misdn/isdn_msg_parser.c: added the export and import of
	  the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the
	  extension is already completely dialed or if there might come
	  additional digits by information elements. also added some docs
	  for that.

2006-12-06 15:42 +0000 [r48320]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #8528 - make sure we don't delete the
	  dialog too quickly after receiving a 487. Move 487 handling into
	  handle_response_invite where it really belongs and don't add an
	  ALREADYGONE flag to the dialog.

2006-12-06 14:35 +0000 [r48319]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: changed a few debugs to higher debug
	  levels

2006-12-06 12:14 +0000 [r48272-48315]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't add Contact header on BYE, CANCEL,
	  MESSAGE requests (Bye, Cancel backported from 1.4, MESSAGE bug
	  reported to me by Gunnar at Omnitor)

	* channels/chan_sip.c: Only set the ALREADYGONE flag once in
	  handle_response()

2006-12-05 01:26 +0000 [r48251]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: If the recording in the database is too
	  large, it will fail to retrieve with an mmap error. Not too sure
	  why this doesn't happen when we put it in the database, also, but
	  since that doesn't seem to be broken, I'm not going to fix it (at
	  least until someone reports it). Solution is to ask for the file
	  in smaller chunks. (Bug 8385)

2006-12-04 21:20 +0000 [r48236-48246]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Revert change from 8016 - this breaks other
	  stuff... Needs further review. Tip: When you've reported a bug
	  about something and somebody has put up a patch for it.. It's not
	  a good idea to open a completely new bug and say that something
	  is broken because of the patch in the other bug - PLEASE mention
	  something in the bug where the patch was actually created.

	* apps/app_voicemail.c: Fix an issue where a message isn't saved
	  correctly when using ODBC storage and reviewing a message. Issue
	  8016 - patch by sokhapkin.

2006-12-04 18:14 +0000 [r48233]  Joshua Colp <jcolp@digium.com>

	* channel.c: If the generic bridge tells us not to retry, and we
	  have a frame to spit out then break the bridge. Props to markit
	  in #asterisk-bugs for bringing this up.

2006-12-01 23:30 +0000 [r48192]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dial.c: if Dial() is going to send music-on-hold to the
	  calling party, it has to send PROGRESS first to ensure that the
	  reverse audio path has been setup first (BE-106)

2006-12-01 20:19 +0000 [r48183]  Jason Parker <jparker@digium.com>

	* configs/extensions.conf.sample: Fix a small typo - issue 8848,
	  reported by pabelanger

2006-11-30 20:47 +0000 [r48165]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 8319 - noriyuki - nonce-count updated
	  *after* use

2006-11-30 20:27 +0000 [r48142-48161]  Joshua Colp <jcolp@digium.com>

	* channel.c: Don't write AST_FRAME_NULL or AST_FRAME_IAX frames out
	  to the channel driver. (issue #8390 reported by hselasky)

	* channels/chan_iax2.c: Only print out debug message if bridged
	  channel is not NULL. (issue #8412 reported by jubilex)

	* res/res_features.c: Do not listen for DTMF on the bridge that
	  comes into existence when ParkedCall is executed. This means
	  native bridging can now occur for this. (issue #8406 reported by
	  kebl0155)

	* cdr.c: Print certain CDR messages out at the NOTICE level versus
	  WARNING since they can occur when used with the CDR applications
	  and are perfectly fine. (issue #8367 reported by dartvader)

	* res/res_features.c: Remember the pointer to the allocated block
	  of memory so that we can free it and not cause a memory leak.
	  (issue #8449 reported by arkadia)

	* configs/sip.conf.sample: Document 'port' for SIP peers, came up
	  because of the current mailing list thread. (issue #8450 reported
	  by blitzrage)

2006-11-30 09:05 +0000 [r48127]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Do proper test and don't leave dialogs
	  hanging...

2006-11-29 16:47 +0000 [r48053-48106]  Joshua Colp <jcolp@digium.com>

	* rtp.c: If the frame was duplicated before writing out then we
	  need to free it. (issue #8429 reported by edguy3)

	* channels/chan_phone.c: According to the research I have done we
	  never needed to include compiler.h in the first place so let's
	  not! (issue #8430 reported by edguy3)

	* apps/app_voicemail.c: Use the proper function to get the new
	  message count instead of always using the filesystem. (issue
	  #8421 reported by slimey)

2006-11-27 17:15 +0000 [r48045]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_musiconhold.c: Random MOH wasn't really random (bug 8381)

2006-11-27 15:30 +0000 [r48037]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_spool.c: Do not reference the freed outgoing structure in
	  the debug message. (issue #8425 reported by arkadia)

2006-11-24 14:33 +0000 [r47987]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Change some logging levels. Not having hints
	  is not an ERROR, but still should be reported.

2006-11-23 16:10 +0000 [r47968]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn_config.c, channels/chan_misdn.c,
	  channels/misdn/isdn_lib.c: fixed a litle bug regarding
	  HOLD/RETRIEVE. beatufied some logs, changed some loglevels.
	  changed the default value of block_on_alarm

2006-11-23 10:54 +0000 [r47958]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Remove unused variable (rizzo)

2006-11-22 02:19 +0000 [r47910]  Steve Murphy <murf@digium.com>

	* channel.c: This is the fix for bug 8386, wherein the time-limit
	  args to dial didn't work correctly

2006-11-20 19:59 +0000 [r47862]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Failing to trap -1 error from mmap causes
	  segfault (Issue 8385)

2006-11-20 19:50 +0000 [r47855-47859]  Joshua Colp <jcolp@digium.com>

	* frame.c: Don't forget to byte swap if we are exiting the smoother
	  feed early. (issue #8287 reported by arturs)

	* channels/chan_sip.c: Free history items at the end of use of the
	  temporary SIP pvt structure. (issue #8383 reported by benh)

2006-11-20 10:17 +0000 [r47842]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Just to be safe, disable all the scheduled
	  items after deleting a scheduler entry (rizzo)

2006-11-17 19:02 +0000 [r47802]  Kevin P. Fleming <kpfleming@digium.com>

	* channel.c: backport proper channel_find_locked behavior from 1.4
	  branch (noted by Steve Davies on asterisk-dev list)

2006-11-16 23:16 +0000 [r47780]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c, apps/app_cut.c, apps/app_directory.c,
	  apps/app_db.c: Fix a couple of typos in applications.. Initially
	  spotted by mrobinson.

2006-11-16 22:57 +0000 [r47776]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/README.cdr: update clearly wrong documentation regarding
	  cdr_custom

2006-11-16 20:29 +0000 [r47750-47761]  Joshua Colp <jcolp@digium.com>

	* cdr/Makefile: Look for the header file specifically in all cases,
	  not just the existence of the directory. (issue #8358 reported by
	  mrness)

	* channels/chan_local.c: Because of the way chan_local is written
	  we should be extra careful and make sure our callback functions
	  have a tech_pvt. (issue #8275 reported by mflorell)

2006-11-16 16:44 +0000 [r47743]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't fixup if we haven't got PVT.
	  Suggestion from Martin Vit on -dev mailing list inspired by
	  file's commit to chan_local. "This shouldn't happen" ;-)

2006-11-15 22:29 +0000 [r47711]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: Make sure that the pvt structure exists
	  before trying to do fixup on Local channels. (issue #7937
	  reported by mada123, fix by alamantia with mods by me)

2006-11-15 21:18 +0000 [r47705]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: CANCEL requests are never authenticated
	  (according to RFC 3261)

2006-11-15 20:30 +0000 [r47666-47696]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_voicemail.c: correct argument name typo that caused
	  global variable to be used instead of the one for the specified
	  voicemail user

	* config.c: when re-writing the config file, don't repeat the path
	  if it hasn't changed

	* config.c: when appending a list of variable to a category, ensure
	  the tail pointer points to the last variable in the list

	* config.c: clear the category's variable tail pointer as well when
	  variables are detached from it

	* config.c: ouch... don't use printf, use ast_log/ast_verbose

	* apps/app_voicemail.c, include/asterisk/config.h: ensure that
	  message duration is included in email notifications for forwarded
	  messages (BE-96, fix by me after corydon used his clue-bat on me)
	  ensure that duration in the message metadata is updated if
	  prepending is done during forwarding (related to BE-96) remove
	  prototype for API call that does not exist

2006-11-15 15:17 +0000 [r47648-47655]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Send error message if we fail to allocate
	  sip socket, possibly caused by too few file handles (wasn't
	  possible before, but with the new way of sending temp messages,
	  it is). Found this bug under heavy load testing with SIPP.

	* channels/chan_sip.c: Sending 200 OK and not getting ACK is
	  considered critical for the call.

2006-11-14 22:15 +0000 [r47631]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Update copyright information in the ADSI
	  logo blob.

2006-11-14 11:06 +0000 [r47596]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Avoid collissions between the peerpoke
	  system and the retransmits. Issue #8272. In some cases, changed
	  timers caused the retransmit system to destroy the dialog before
	  peerpoke_expire got a chance.

2006-11-13 21:26 +0000 [r47583]  Joshua Colp <jcolp@digium.com>

	* cdr/cdr_pgsql.c: Initialize global pointers for connection and
	  result to NULL. (issue #8356 reported by james)

2006-11-13 20:18 +0000 [r47580]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Having more than 255 old messages caused
	  corruption in the new/old count

2006-11-13 19:04 +0000 [r47571]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't send 487 if we've already sent 200 OK
	  on invite at time of receiving a BYE in the same transaction.
	  (SIPP testing)

2006-11-13 17:05 +0000 [r47549]  Joshua Colp <jcolp@digium.com>

	* apps/app_sms.c: When sending an SMS with a user data header
	  properly set the UDH flag in the first byte. (issue #8347
	  reported by hoffmeis)

2006-11-13 05:45 +0000 [r47522-47525]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: If the execute fails a second time, make sure
	  that we don't pass back a stale handle

	* channels/chan_zap.c: Don't play dialtone if the seizing the
	  channel fails (Bug 7754)

2006-11-12 06:09 +0000 [r47496]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Only do the check to determine whether the
	  channel calling this function is an IAX2 channel when getting the
	  IP address using the special argument, CURRENTCHANNEL. (issue
	  #8341, jcovert)

2006-11-10 20:46 +0000 [r47452-47470]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Clear dialog on loop (backport from 1.4 by
	  mistake)

	* channels/chan_sip.c: - Don't check for ignore in blocks that
	  isn't reached if ignore is on... - return properly after sending
	  reply in handle_request_invite

	* channels/chan_sip.c: Fix multipart/mixed SDP support (issue 8010,
	  alphaque)

2006-11-09 16:48 +0000 [r47379]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c: Don't include compiler.h on kernels 2.6.18
	  and higher as, well, it's apparently going to be removed. This
	  should make all you FC6 fans happy as your Asterisk will now
	  build without any mods.

2006-11-09 13:09 +0000 [r47359]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn_config.c, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h: Fixed segfault when no
	  misdn.conf exists, reported by Igor Neves, thanks.

2006-11-08 07:40 +0000 [r47307-47308]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Remove dialog properly at unload of module
	  (rizzo)

2006-11-07 18:22 +0000 [r47274]  Steve Murphy <murf@digium.com>

	* include/asterisk/channel.h, channel.c: This mod for bug_7506, to
	  make the manager code output the proper event

2006-11-07 13:02 +0000 [r47248]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't ever reply to an ACK. (Issue 8265)

2006-11-07 01:22 +0000 [r47238]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: If random order is enabled for files mode
	  music on hold, set a random initial position, instead of always
	  starting at the first file, and doing the random operation only
	  when switching to the next file. (bug reported by John Lange on
	  the asterisk-dev mailing list)

2006-11-02 17:47 +0000 [r46964]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: ignore files in a music on hold directory
	  that begin with '.' (issue #8249, cboie)

2006-11-02 15:15 +0000 [r46899]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't overwrite flags in the packet

2006-11-02 13:55 +0000 [r46876]  Russell Bryant <russell@digium.com>

	* callerid.c: Add a missing call to free before returning in an
	  error condition (issue #8268, mrness)

2006-11-01 21:20 +0000 [r46838]  Matt O'Gorman <mogorman@digium.com>

	* logger.c: fix for bug #8083 crash caused by double free on m->msg

2006-11-01 19:52 +0000 [r46803]  Steve Murphy <murf@digium.com>

	* res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
	  accept longer strings or mass confusion and a lot of lost time is
	  the result

2006-11-01 18:24 +0000 [r46776]  Russell Bryant <russell@digium.com>

	* res/res_monitor.c: soxmix and Asterisk expect different file
	  extensions for certain formats. This was already handled for the
	  wav49 format. However, it was not handled for ulaw and alaw. I
	  fixed this in such a way that using the alternate extensions for
	  ulaw and alaw will only happen if we know we're calling soxmix,
	  and not a custom script defined using the MONITOR_EXEC variable.
	  The wav49 processing was left alone so that external scripts will
	  see no behavior change. (issue #7550, reported by mnicholson,
	  proposed patch by junky, committed fix is a bit different)

2006-10-31 15:46 +0000 [r46662]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_curl.c: Move thread-unsafe initializer to the module
	  loading code; add the corresponding function to the module unload
	  to fix a memory leak.

2006-10-31 09:49 +0000 [r46585-46610]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: Another try to fix
	  ;rport NAT traversal support (issue #7473)

	* channels/chan_sip.c: If peer fails ACL check, fail the REGISTER
	  attempt

	* channels/chan_sip.c: On the other hand, we already copy the NAT
	  flags... Reverting.

	* channels/chan_sip.c: Issue 7473 - support ;rport on REGISTER
	  requests too.

2006-10-31 06:18 +0000 [r46557-46560]  Russell Bryant <russell@digium.com>

	* utils.c: When handling the case where the hostname is just an
	  IPV4 numeric address, be sure to set the address type. (issue
	  #8247, alexr)

	* res/res_agi.c: fix some copy/paste bugs in the checking of
	  arguments for the "control stream file" AGI command (issue #8255,
	  mnicholson)

2006-10-30 16:00 +0000 [r46402-46430]  Olle Johansson <oej@edvina.net>

	* rtp.c: Bind rtcp to proper IP address

	* channels/chan_sip.c: Issue #7869 - Stop sending 302 redirect when
	  not getting an answer...

	* channels/chan_sip.c: issue #7608: Notifications with wrong
	  content-type. Reported by jsiddall.

2006-10-27 17:36 +0000 [r46361]  Russell Bryant <russell@digium.com>

	* res/res_agi.c, asterisk.c, apps/app_externalivr.c,
	  res/res_musiconhold.c: We should always be using _exit() after a
	  fork() or vfork() instead of exit(). This is because exit() does
	  some extra cleanup which in some implementations of vfork(), for
	  example, can actually modify the state of the parent process,
	  causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)

2006-10-27 09:24 +0000 [r46350]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
	  fixed a bug which caused chan_misdn to try to allocate 2 times
	  the same channel on high load, which then caused instability of
	  mISDN. removed a useless function from isdn_lib.c

2006-10-26 20:06 +0000 [r46344]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #7240, by mistake only committed to
	  trunk (now 1.4), reported by edgreenberg in Issue #7966. Thanks
	  Ed!

2006-10-26 17:47 +0000 [r46332-46337]  Jason Parker <jparker@digium.com>

	* contrib/scripts/astgenkey.8: oops - somebody forgot to change
	  this - long ago, probably.

	* channels/chan_skinny.c: Remove a useless ast_mutex_unlock. Issue
	  #8186, patch by anthonyl (fix suggested by benh).

2006-10-25 19:28 +0000 [r46213-46258]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Working to resolve #7608 - adding debug
	  output

	* channels/chan_sip.c: Fix the attack shield for 1.2 too. REFER and
	  NOTIFY can create dialogs in the world of Asterisk.

2006-10-25 08:41 +0000 [r46176]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn_config.c, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
	  added nttimeout option to configure wether we disconnect calls on
	  NT timeouts or not during an overlapdial session

2006-10-23 00:25 +0000 [r45927]  Joshua Colp <jcolp@digium.com>

	* cdr/cdr_odbc.c: Don't leak memory mmmk?

2006-10-21 12:35 +0000 [r45808]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed issue, that if chan_misdn is loaded
	  and couldn't be initialized it would cause a segfault after
	  'reload'. Reported by Drew/Matt thx.

2006-10-19 17:16 +0000 [r45691]  Joshua Colp <jcolp@digium.com>

	* apps/app_externalivr.c: Respect language selection when seeing if
	  the file exists (issue #8178 reported by mnicholson)

2006-10-17 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.13 released

2006-10-17 20:37 +0000 [r45380]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't create a "real" pvt structure for
	  requests that shouldn't be able to create one. Instead use a
	  temporary pvt and fill it with enough information so we can send
	  a reply.

2006-10-17 17:50 +0000 [r45332]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix an integer signedness problem.

2006-10-17 17:22 +0000 [r45326]  Kevin P. Fleming <kpfleming@digium.com>

	* LICENSE: provide licensing language for IAXy firmware file

2006-10-17 15:50 +0000 [r45306]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: After some
	  research, we realized that the default behaviour since a long
	  time was doing the right thing, even though the change optimized
	  a bit and removed a lot of potential risks. Conclusion: No need
	  for a configuration option at all.

2006-10-16 19:59 +0000 [r45260-45265]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Use responses
	  rather then replies even though they mean the same thing.

	* channels/chan_sip.c, configs/sip.conf.sample: Add
	  'ignoreoodreplies' option which will not create a pvt structure
	  on a SIP response but instead basically drop it.

2006-10-14 00:16 +0000 [r45134]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: Made a small update to solve bug 8128; The
	  switch-case fallthru goto to a pattern extension needed to
	  resolved the wildcards to an appropriate digit for extension
	  matching to work

2006-10-13 22:57 +0000 [r45119]  Kevin P. Fleming <kpfleming@digium.com>

	* acl.c: don't drop the entire permit/deny list when an attempt is
	  made to add an invalid entry (BE-92)

2006-10-13 19:27 +0000 [r45090]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: avoiding warning, fixing potential bug
	  (backported from 1.2)

2006-10-13 17:01 +0000 [r45060]  Joshua Colp <jcolp@digium.com>

	* apps/app_chanspy.c: Turn on volume adjustment if it needs to be
	  on (issue #8136 reported by mnicholson)

2006-10-13 16:18 +0000 [r45048]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: when sending a call to a peer, use the
	  proper socket if we have multiple bindings (reported on
	  asterisk-dev)

2006-10-13 15:49 +0000 [r45030]  Joshua Colp <jcolp@digium.com>

	* dnsmgr.c: Pass the right value to usleep for sleeping, and always
	  add the background refresh item back into the scheduler if
	  enabled since it is deleted during reload. (issue #8142 reported
	  by p_lindheimer)

2006-10-13 13:11 +0000 [r44993-45020]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed some
	  echocandisable issues when bridged. this caused a kernel panic
	  sometimes..also some minor formatting fixes

	* channels/misdn/isdn_msg_parser.c: fixed issue, that the
	  hangupcause got a wrong isdn cause at RELEASE_COMPLETE

2006-10-12 18:31 +0000 [r44955]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/utils.h, channels/chan_sip.c, utils.c,
	  netsock.c: ensure that IAX2 and SIP sockets allow UDP
	  fragmentation when running on Linux (thanks to Brian Candler on
	  the asterisk-dev list for the tip)

2006-10-10 13:34 +0000 [r44785]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/isdn_lib.c: (re)added
	  support of dynamical enabling hdlc on bchannels

2006-10-09 14:36 +0000 [r44757]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #8101 - wrong parameter for screening
	  in remote-party-id

2006-10-06 16:52 +0000 [r44501-44580]  Joshua Colp <jcolp@digium.com>

	* file.c: Even more frames to treat as though the remote side
	  disappeared (issue #8097 reported by eldadran)

	* file.c: Treat busy control frames as hangup in the file streaming
	  core (issue #8097 reported by eldadran)

2006-10-05 10:02 +0000 [r44460]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed segfault which happens during
	  hold/transfer action

2006-10-05 01:27 +0000 [r44392-44432]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: fix Polycom presence notification again

	* channels/chan_sip.c: remove workaround for old Polycom firmware
	  SUBSCRIBE requests add workaround for new Polycom firmware
	  SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware)

2006-10-04 16:02 +0000 [r44343]  Steve Murphy <murf@digium.com>

	* apps/app_macro.c: For bug 7776, I have inserted a warning about
	  Macro nesting vs. stack limitations

2006-10-04 15:26 +0000 [r44334-44335]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: if INFORMATION Message come with keypad
	  instead of called party number, we just use the keypad as called
	  party number.

	* channels/misdn_config.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
	  configs/misdn.conf.sample, channels/misdn/isdn_lib.c: added the
	  option 'reject_cause' to make it possible to set the
	  RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is
	  automatically rejected because chan_misdn does not support that
	  kind of callwaiting. Therefore chan_misdn supports now 3 incoming
	  channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the
	  info if the requested channel is incoming or outgoing to make the
	  3. channel possible

2006-10-03 20:14 +0000 [r44296]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: fix a logic error in my previous fix to the
	  queue reload code

2006-10-02 20:07 +0000 [r44168-44213]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Change the fd on the I/O context in case it
	  changed during the reload, which is indeed possible. (issue #7943
	  reported by eclubb)

	* contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
	  instead of hardcoding the path for the error message (issue #7942
	  reported by eclubb)

	* io.c: Shrink when current_ioc is unused. It is set to -1 when
	  unused, not 0. (issue #7941 reported by eclubb)

2006-10-02 13:28 +0000 [r44149]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/isdn_lib.c: fixed the hold/retrieve/transfer
	  issues, removed a useless bc field, added setting of
	  frame.delivery fields, some minor code cleanups

2006-10-01 15:19 +0000 [r44110]  Russell Bryant <russell@digium.com>

	* configs/queues.conf.sample: Fix the name of the
	  "eventmemberstatus" option in the sample queues.conf (issue
	  #8065, adamg)

2006-09-29 13:44 +0000 [r43977]  Kevin P. Fleming <kpfleming@digium.com>

	* cli.c: proper fix for ast_group_t change

2006-09-28 18:00 +0000 [r43924]  Joshua Colp <jcolp@digium.com>

	* frame.c, include/asterisk/logger.h, channels/chan_misdn.c,
	  channels/chan_sip.c, channels/chan_skinny.c,
	  funcs/func_timeout.c, apps/app_festival.c, res/res_features.c,
	  apps/app_hasnewvoicemail.c, apps/app_alarmreceiver.c,
	  channels/iax2-provision.c, res/res_musiconhold.c,
	  res/res_monitor.c: Put in missing \ns on the end of ast_logs
	  (issue #7936 reported by wojtekka)

2006-09-28 17:31 +0000 [r43916]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: fix buggy (and overly complex) loop used during
	  reload of app_queue for static member list updating

2006-09-28 16:37 +0000 [r43897]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: app_queue is comparing the device names
	  incorrectly while checking their statuses. It's internal list of
	  interfaces includes the dial string, while the argument passed to
	  this function does not have the dial string (/n for a local
	  channel). This causes it to ignore the device state changes
	  because it thinks it belongs to none of its members. (#8040
	  reported and patch by tim_ringenbach)

2006-09-28 16:32 +0000 [r43895]  Kevin P. Fleming <kpfleming@digium.com>

	* cli.c: eliminate compiler warning introduced by recent changes

2006-09-28 16:13 +0000 [r43891]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Stop the stream after waitstream returns so
	  that our formats get restored. (issue #7370 reported by
	  kryptolus)

2006-09-28 15:18 +0000 [r43871]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: Fix race condion crash with get_member_status
	  (#7864 - tim_ringenbach reported and patched)

2006-09-27 20:20 +0000 [r43815]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Avoid inability to lock directory log
	  message by creating the directory ahead of time. (Issue 7631)

2006-09-27 19:35 +0000 [r43800]  Jason Parker <jparker@digium.com>

	* apps/app_playback.c, pbx.c: Playback() wasn't setting
	  PLAYBACKSTATUS under several circumstances. Playback() returns -1
	  on missing args - so should Background()

2006-09-27 16:54 +0000 [r43778]  Russell Bryant <russell@digium.com>

	* res/res_features.c, channel.c: Fix a problem that occurred if a
	  user entered a digit that matched a bridge feature that was
	  configured using multiple digits, and the digit that was pressed
	  timed out in the feature digit timeout period. For example, if
	  blind transfer is configured as '##', and a user presses just
	  '#'. In this situation, the call would lock up and no longer pass
	  any frames. (issue #7977 reported by festr, and issue #7982
	  reported by michaels and valuable input provided by mneuhauser
	  and kuj. Fixed by me, with testing help and peer review from
	  Joshua Colp). There are a couple of issues involved in this fix:
	  1) When ast_generic_bridge determines that there has been a
	  timeout, it returned AST_BRIDGE_RETRY. Then, when
	  ast_channel_bridge gets this result, it calls ast_generic_bridge
	  over again with the same timestamp for the next event. This
	  results in an endless loop of nothing until the call is
	  terminated. This is resolved by simply changing
	  ast_generic_bridge to return AST_BRIDGE_COMPLETE when it sees a
	  timeout. 2) I also changed ast_channel_bridge such that if in the
	  process of calculating the time until the next event, it knows a
	  timeout has already occured, to immediately return
	  AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
	  anyway. 3) In the process of testing the previous two changes, I
	  ran into a problem in res_features where ast_channel_bridge would
	  return because it determined that there was a timeout. However,
	  ast_bridge_call in res_features would then determine by its own
	  calculation that there was still 1 ms before the timeout really
	  occurs. It would then proceed, and since the bridge broke out and
	  did *not* return a frame, it interpreted this as the call was
	  over and hung up the channels. The reason for this was because
	  ast_bridge_call in res_features and ast_channel_bridge in
	  channel.c were using different times for their calculations.
	  channel.c uses the start_time on the bridge config, which is the
	  time that the feature digit was recieved. However, res_features
	  had another time, 'start', which was set right before calling
	  ast_channel_bridge. 'start' will always be slightly after
	  start_time in the bridge config, and sometimes enough to round up
	  to one ms. This is fixed by making ast_bridge_call use the same
	  time as ast_channel_bridge for the timeout calculation.

2006-09-27 12:51 +0000 [r43764]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/isdn_lib.c: fixed a bug which led to chan_list
	  zombies, when the call could not be properly established in
	  misdn_call. also removed the ACK_HDLC stuff which is not really
	  needed.

2006-09-26 20:49 +0000 [r43708]  Russell Bryant <russell@digium.com>

	* asterisk.c: Back in revision 4798, this message was changed from
	  using ast_cli() to directly calling write(). During this change,
	  checking if this was a remote console was removed. This caused
	  this message about using "exit" or "quit" to exit an Asterisk
	  console to come up in times where it did not make sense. This
	  change restores the check to see if this is a remote console
	  before printing the message. (fixes BE-4)

2006-09-26 20:38 +0000 [r43705-43706]  Joshua Colp <jcolp@digium.com>

	* .cleancount: I changed the channel structure... let's be sure
	  this gets updated!

	* channels/chan_sip.c, include/asterisk/channel.h: Use proper type
	  to represent the group variable (issue #8025 reported by makoto)

2006-09-26 20:23 +0000 [r43699]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: When parsing the sections of voicemail.conf
	  that contain mailbox definitions, don't introduce a length limit
	  on the definition by using a 256 byte temporary storage buffer.
	  Instead, make the temporary buffer just as big as it needs to be
	  to hold the entire mailbox definition. (fixes BE-68)

2006-09-25 21:14 +0000 [r43634]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue
	  7824): 1) delete=yes was ignored 2) maxmessages was ignored

2006-09-24 13:50 +0000 [r43552]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Check to see if the channel that is
	  activating the IAXPEER function is actually an IAX2 channel
	  before proceeding to process it to avoid crashing. (issue #8017,
	  reported by admott, fixed by myself)

2006-09-22 21:53 +0000 [r43509]  Joshua Colp <jcolp@digium.com>

	* apps/app_chanspy.c, channel.c: Yay another 'round of spy fixes!
	  This fixes a small logic flaw with the cleanup function and a
	  memory allocation issue. (issue #7960 reported by jojo & issue
	  #7999 reported by aster1) Special thanks to csum77 for letting me
	  into a box where this issue was happening.

2006-09-21 17:01 +0000 [r43409-43420]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_rpt.c: Whitespace change... really just an excuse to
	  test repotools

	* cdr/cdr_tds.c, cdr/Makefile: TDS 0.64 updates

2006-09-20 05:08 +0000 [r43314]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_misdn.c, channels/chan_sip.c,
	  channels/chan_skinny.c: make some more functions static

2006-09-19 16:21 +0000 [r43269]  Matt O'Gorman <mogorman@digium.com>

	* pbx/pbx_gtkconsole.c, apps/app_dial.c, channels/chan_sip.c,
	  apps/app_macro.c, asterisk.c, config.c, apps/app_queue.c, pbx.c:
	  fixes some verbose vs debug issues. patch from bug 2617

2006-09-19 12:28 +0000 [r43248]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: cid is passed to a destructive function;
	  thus a copy is needed (issue 7961)

2006-09-18 20:08 +0000 [r43220]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #7682 - don't add contacts to 4xx
	  responses. (Ugly fix, not proud at all)

2006-09-18 15:30 +0000 [r43163]  Joshua Colp <jcolp@digium.com>

	* apps/app_math.c: Add deprecation notice about app_math (issue
	  #7957 reported by k-egg)

2006-09-18 15:05 +0000 [r43160]  Steve Murphy <murf@digium.com>

	* configs/zapata.conf.sample: Clarified what "callwaiting" does in
	  zapata.conf.

2006-09-18 15:05 +0000 [r43159]  Joshua Colp <jcolp@digium.com>

	* configs/indications.conf.sample: Add number unobtainable tone for
	  New Zealand (issue #7969 reported by nic_bellamy)

2006-09-17 13:54 +0000 [r43072]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_directory.c: Directory used the wrong context for
	  delivery of 0- and *- keypresses (according to Directory's own
	  documentation) - Issue 7965

2006-09-16 07:57 +0000 [r43003-43019]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_iax2.c: When a realtime peer expires, reset the
	  ipaddress in the realtime database back to 0 (Issue 6656)

	* apps/app_meetme.c: When the marked user enters the conference, we
	  should no longer timeout

2006-09-14 22:16 +0000 [r42946]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_zap.c: Error message references wrong argument
	  (Issue 7951)

2006-09-13 19:51 +0000 [r42892]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Backport bugfix patch from 7918 to 1.2 -
	  msg_cfg destroyed before used

2006-09-11 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.12.1 released

2006-09-11 21:47 +0000 [r42697-42783]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_meetme.c, apps/app_page.c: When paging, only wait 5
	  seconds for the marked user to enter the conference. After that,
	  assume the paging already completed by the time the channel
	  entered the conference and drop back out. (Issue 7275)

	* configs/extensions.conf.sample, configs/alsa.conf.sample,
	  configs/zapata.conf.sample, configs/iax.conf.sample,
	  configs/osp.conf.sample, configs/dundi.conf.sample,
	  configs/enum.conf.sample, configs/vpb.conf.sample,
	  configs/cdr.conf.sample, configs/voicemail.conf.sample,
	  configs/phone.conf.sample, configs/misdn.conf.sample,
	  configs/sip.conf.sample, configs/skinny.conf.sample,
	  configs/features.conf.sample: Spelling/grammar fixes (Issue 7929)

	* configs/voicemail.conf.sample: Two grammar issues (bug 7927)

2006-09-09 20:24 +0000 [r42600]  Joshua Colp <jcolp@digium.com>

	* channel.c: Only truly consider the channel in the same format if
	  the format matches the raw format OR if a translation path
	  already exists to translate between them. (issue #7887 reported
	  by softins & issue #7803 reported by alvaro_palma_aste). Thanks
	  goes to stubert for giving me access to a box and showing me a
	  scenario where this occured.

2006-09-09 12:14 +0000 [r42535]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Reset proper flag - Don't delete SIP
	  dialog prematurely Strangely enough imported from svn trunk...
	  It's confusing here in Greenland. (Committing from 36.000 feet
	  above Greenland, on the way to asterisk@von
	  http://www.pulver.com/asterisk )

2006-09-08 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.12 released

2006-09-08 18:50 +0000 [r42452]  Joshua Colp <jcolp@digium.com>

	* channel.c: Swap spies during masquerading

2006-09-08 16:06 +0000 [r42421]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_authenticate.c: Jump logic was backwards: goto returns 0
	  if it succeeds, and we should jump if authentication fails. (Bug
	  #7907)

2006-09-08 04:37 +0000 [r42402]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: Use ast_best_codec to set the read/write
	  format

2006-09-07 23:12 +0000 [r42355]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_record.c: Format vulnerability fix - allowing the user
	  to specify a format is not a good idea (Bug 7811)

2006-09-07 16:30 +0000 [r42260]  Joshua Colp <jcolp@digium.com>

	* cdr.c: Let's use the same thing we use in other places to
	  calculate our time for ast_cond_timedwait (issue #7697 reported
	  by bn999)

2006-09-07 02:14 +0000 [r42150-42200]  Steve Murphy <murf@digium.com>

	* logger.c: This should fix the problem reported in 7564: logger
	  config file errors getting lost because logging isn't configured
	  yet. The problem was that the code that exists to handle this
	  case was not getting reached, because other tests were causing an
	  early return from ast_log().

	* Makefile: added hours,minutes,seconds .gsm files to the install
	  portion of the makefile, as per bug 7545

2006-09-06 20:02 +0000 [r42148]  Joshua Colp <jcolp@digium.com>

	* res/res_agi.c: Don't close the second file descriptor if it's the
	  same as the first one, as it will have already been closed
	  elsewhere and could cause massive panic. (issue #7699 reported by
	  bn999)

2006-09-06 18:16 +0000 [r42133]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_agent.c: Look ma! No more deadlocks! <sic> As
	  posted from #7458 and others similar to it in Mantis: p->app_lock
	  was a mutex really designed for use with agents not in callback
	  mode. That being the case, I've tried to code it so that when
	  callback mode is used, the app_lock mutex will not be
	  locked/unlocked at all. Please let me know how you make out - and
	  if you continue to deadlock now, please reproduce the deadlock
	  logging information and post to Mantis.

2006-09-06 17:10 +0000 [r42110]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed pipe consuming bug when using
	  chanIsAvail (#7878), also moved a debug log to the very begining
	  of misdn_hangup.

2006-09-06 15:55 +0000 [r42054-42086]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Make realtime regseconds work as people
	  expected (0 on registration expiration or release, and actual on
	  normal state) (issue #7684 reported by kshumard)

	* include/asterisk/chanspy.h, apps/app_chanspy.c,
	  apps/app_mixmonitor.c, channel.c: Merge in last round of spy
	  fixes. This should hopefully eliminate all the issues people have
	  been seeing by distinctly separating what each component
	  (core/spy) is responsible for. Core is responsible for adding a
	  spy to a channel, feeding frames to the spy, removing the spy
	  from a channel, and telling the spy to stop. Spy is responsible
	  for reading frames in, and cleaning up after itself.

2006-09-05 16:27 +0000 [r42014]  Jason Parker <jparker@digium.com>

	* configs/zapata.conf.sample: Small typo in zapata.conf.sample
	  Reported by ppyy in 7881

2006-09-04 15:46 +0000 [r41989]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't kill the pvt before we have sent ACK
	  on CANCEL

2006-09-03 17:38 +0000 [r41827-41882]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: Make sure the forwarded channel inherits
	  variables appropriately when we receive a call forward in the
	  queue. (#7867 - raarts reported and patched)

	* apps/app_queue.c: Don't keep trying the same member in certain
	  strategies when members of the queue are unavailable (#7278 -
	  diLLec reported and patched)

	* apps/app_chanspy.c: Let's NOT spy on Zap/psuedo channels,
	  mmmmmmmmk?

	* apps/app_queue.c: Setting a retry of 0 is generally not a good
	  idea and shouldn't be allowed. (#7574 - reported by regin)

2006-09-01 22:49 +0000 [r41768]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only wipe the redirected audio & video
	  IP/port if it's specified, and trigger a reinvite.

2006-09-01 17:35 +0000 [r41716]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c, include/asterisk/rtp.h, rtp.c: put in proper
	  fix for issue #7294 instead of the broken partial fix that was
	  committed, and thereby also fix issue #7438

2006-09-01 16:33 +0000 [r41690-41691]  Joshua Colp <jcolp@digium.com>

	* channel.c: Finish up the last commit (was worse then originally
	  reported)

	* channel.c: Don't treat an unexpected control subclass as voice
	  (issue #7858 reported by PCadach)

2006-08-30 19:01 +0000 [r41423]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #7572 - Hangup when receiving a buggy
	  487 response to an INVITE

2006-08-30 18:59 +0000 [r41411]  Russell Bryant <russell@digium.com>

	* channels/chan_mgcp.c, channels/chan_phone.c,
	  channels/chan_local.c, channels/chan_misdn.c,
	  channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_features.c, channels/chan_h323.c,
	  channels/chan_iax2.c: Restore original functionality of 1.2 in
	  places where ANI was not set, but was changed to be set. The
	  original change was done to ensure that the behavior of the
	  "callerid" option in each channel driver was consistent, but it
	  caused an unexpected behavior change of CDR records for users, so
	  this change is being reverted in 1.2. (issue #7695)

2006-08-30 17:58 +0000 [r41390]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/lock.h: Properly handle an ETIMEDOUT result from
	  pthread_cond_timedwait (issue #7318 reported by arkadia)

2006-08-30 14:31 +0000 [r41334]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 7822 - don't use SRV lookups if it's
	  disabled.

2006-08-29 13:33 +0000 [r41269]  Russell Bryant <russell@digium.com>

	* pbx/pbx_config.c: clean up last commit ... most notably, there is
	  no reason to do heap allocations here, and it also included a
	  potential memory leak

2006-08-29 05:49 +0000 [r41239-41262]  Steve Murphy <murf@digium.com>

	* pbx/pbx_config.c: Fixes for bug 7813, via patch submitted by
	  stevens.

	* doc/README.variables: Removed from the docs the mention of the !
	  and =~ operators, as these were knocked out of ast_expr2 because
	  they were new features. Let's hope I can keep them from getting
	  knocked out of the trunk, too!

	* apps/app_macro.c: According to a note added to 7731 by
	  mneuhauser, this will repair a break caused by the last fix
	  (7731).

2006-08-25 15:21 +0000 [r41066-41069]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Don't send proceeding twice (#7800)

2006-08-25 15:07 +0000 [r41065]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Text only - clarify the reason for entry
	  into authentication mode when the skipuser option is ignored

2006-08-24 19:41 +0000 [r40994]  Russell Bryant <russell@digium.com>

	* include/asterisk/linkedlists.h, channel.c, pbx.c: Fix a few
	  issues related to the handling of channel variables - in
	  pbx_builtin_serialize_variables(), the variable list traversal
	  would stop on a variables with empty name/values, which is not
	  appropriate - When removing the GROUP variables, use
	  AST_LIST_REMOVE_CURRENT instead of AST_LIST_REMOVE - During
	  masquerading, when copying the variables list from one channel to
	  the other, using AST_LIST_INSERT_TAIL is not valid for appending
	  a whole list. It leaves the tail pointer of the list invalid.
	  Introduce a new macro, AST_LIST_APPEND_LIST that appends a list
	  properly. (issue #7802, softins)

2006-08-24 17:13 +0000 [r40971-40979]  Joshua Colp <jcolp@digium.com>

	* configs/zapata.conf.sample: Minor documentation fix to add the
	  'dynamic' dialplan option from angler

2006-08-23 16:05 +0000 [r40901]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_agi.c: Revert last change - breaks retrieval of builtin
	  variables

2006-08-22 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.11 released

2006-08-22 02:59 +0000 [r40821]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_random.c: Bug 7779 - Using initstate(3) means that we
	  cannot unload this module once loaded.

2006-08-21 22:34 +0000 [r40798]  Matt O'Gorman <mogorman@digium.com>

	* asterisk.c: Move the load_modules call so that if a module needs
	  realtime support it will work, none do currently but a good move
	  none the less.

2006-08-20 22:09 +0000 [r40692]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* CREDITS: Reformat to match the contribution style of other
	  contributors

2006-08-20 04:49 +0000 [r40601]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Turn media level c= parsing on by default
	  (issue #7725 reported by psm)

2006-08-19 01:03 +0000 [r40446]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c, apps/app_directory.c: Fix a bug with
	  app_voicemail when trying to use app_directory to leave messages
	  to another user (options 3, 5, 2). If the context/extension
	  didn't exist in the dialplan (and why should it have to?), it
	  would fail, saying that it's an "invalid extension". Fix was
	  different in svn trunk. (issue BE-71)

2006-08-18 19:10 +0000 [r40310-40392]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/zapata.conf.sample: make a feeble attempt to avoid the
	  'how do I enable my hardware echo canceler' questions

	* channels/misdn_config.c (added), channels/chan_misdn_config.c
	  (removed): rename file per crichter's request

2006-08-17 21:57 +0000 [r40306]  Christian Richter <christian.richter@beronet.com>

	* doc/README.misdn, channels/misdn/mISDN.patch (removed),
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/fac.c (added), channels/misdn/Makefile,
	  channels/misdn/chan_misdn_config.h, channels/misdn/ie.c,
	  channels/misdn/fac.h (added), channels/misdn/portinfo.c
	  (removed), channels/misdn/isdn_lib_intern.h,
	  channels/chan_misdn_config.c, channels/misdn/isdn_msg_parser.c,
	  configs/misdn.conf.sample, channels/Makefile,
	  channels/misdn/isdn_lib.c: This rather small ;-) commit merges
	  the changes from my team branch 0.3.0 into t he 1.2 branch. These
	  changes include the new mISDN mqueue interface which makes it
	  possible to compile chan_misdn against the current cvs version of
	  mISDN/mISDNuser. These changes also contain various additions and
	  numerous bugfixes to chan_misdn . Each change is documented in
	  the commit logs in the team/crichter/0.3.0 branch.

2006-08-17 16:36 +0000 [r40227]  Russell Bryant <russell@digium.com>

	* channel.c: revert bogus change to attempt to fix bug 7506 which
	  actually causes half of the channels not to get "Newchannel"
	  events at all (issue #7745)

2006-08-17 16:22 +0000 [r40223-40225]  Joshua Colp <jcolp@digium.com>

	* funcs/func_cdr.c: Use the last CDR entry instead of the first CDR
	  entry for variable retrieving variables using the CDR dialplan
	  function. (issue #7689 reported by voipgate)

	* apps/app_macro.c: Make app_macro compile again

2006-08-17 16:07 +0000 [r40220]  Steve Murphy <murf@digium.com>

	* apps/app_macro.c: In app_macro, changed the previously changed
	  upper recursion depth limit to a variable, default of the
	  original val of 7. MACRO_RECURSION is a channel variable that
	  will override the limit, but until I can understand and fix why
	  this limit is neccessary, I am not advertising this variable in
	  the docs. This fix mirrors the changes made in r40200 in trunk.

2006-08-16 18:57 +0000 [r40057]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_mgcp.c: don't allow AUEP responses to overflow the
	  stack during a string copy (reported by Mu Security)

2006-08-15 22:49 +0000 [r39935]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: use pbx_builtin_getvar_helper() so that GET
	  VARIABLE can retrieve global variables (issue #7609)

2006-08-15 22:13 +0000 [r39931]  Steve Murphy <murf@digium.com>

	* apps/app_macro.c: This revision fixes bug 7731, the inability for
	  macros to be called more than one level deep in the 'h'
	  extension. It also pushes up the limit of recursion depth from 7
	  to 20.

2006-08-08 18:39 +0000 [r39379]  Kevin P. Fleming <kpfleming@digium.com>

	* CREDITS: add explicit listing of anthm's contributions (issue
	  #7683)

2006-08-08 17:04 +0000 [r39350]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Increase the buffer size for the callid
	  (issue #7675, reported by pssatcs)

2006-08-07 01:28 +0000 [r39081]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Fix a crash reported to me by hads on IRC.
	  This crash would occur with the use of the
	  "distinctiveringaftercid" option. Also, on this user's system,
	  the crash would only occur when built without optimizations. This
	  is because the bug is that the code would write past the end of
	  an array that was allocated on the stack, and the structure of
	  the stack is different with or without optimizations enabled.

2006-08-07 00:15 +0000 [r39056]  Joshua Colp <jcolp@digium.com>

	* channel.c: Reset our stream and vstream pointers back to NULL so
	  that any generator that uses them (file based MOH) will not try
	  to close them again. (issue #7668 reported by jmls)

2006-08-05 09:01 +0000 [r38903-38982]  Russell Bryant <russell@digium.com>

	* channel.c: Always generate a Newstate event in ast_setstate()
	  instead of making it a Newchannel event if the state was
	  AST_STATE_DOWN. The Newchannel event will always be generated in
	  ast_request(), so this just causes a duplicated Newchannel event
	  in some cases. (issue #7506, repoted by capouch, fixed by me)

	* apps/app_queue.c: remove duplicate queue log entry when the
	  caller exits on a timeout (issue #7616, ppyy)

	* channels/chan_sip.c: don't advertise that this function can set a
	  SIP header when it can only do reads

	* apps/app_dial.c: make sure the priv-callerintros directory exists
	  before trying to create a file there (issue #7659, patch by hads,
	  with some modifications by me)

	* channels/chan_mgcp.c, channels/chan_vpb.c, channels/chan_phone.c,
	  channels/chan_misdn.c, channels/chan_zap.c, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_modem.c, channels/chan_iax2.c: Fix an issue that
	  would cause a NewCallerID manager event to be generated before
	  the channel's NewChannel event. This was due to a somewhat recent
	  change that included using ast_set_callerid() where it wasn't
	  before. This function should not be used in the channel driver
	  "new" functions. (issue #7654, fixed by me) Also, fix a couple
	  minor bugs in usecount handling. chan_iax2 could have increased
	  the usecount but then returned an error. The place where chan_sip
	  increased the usecount did not call ast_update_usecount()

	* channel.c: suppress a compiler warning about the usage of a
	  potentially uninitialized variable

2006-08-03 19:54 +0000 [r38825]  Joshua Colp <jcolp@digium.com>

	* res/res_musiconhold.c: Treat the file as invalid if we have no
	  valid formats for it (issue #7643 reported by KNK)

2006-08-03 05:22 +0000 [r38761]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 7648 - Checking wrong count for
	  plurality on new messages for Dutch language

2006-08-02 19:29 +0000 [r38686-38731]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: fix brain-damage I introduced when trying to
	  fix the CANCEL/BYE sending mechanism for pending INVITES accept
	  unknown 1xx responses as 183 responses (as RFC3261 mandates we
	  should do)

	* res/res_features.c, channel.c: ensure that the 'feature digit
	  timeout' value is taken into account when deciding how long the
	  bridge should run (this fixes a problem report where a digit
	  press that did not invoke a feature is never passed across the
	  bridge)

2006-08-01 19:20 +0000 [r38654]  Joshua Colp <jcolp@digium.com>

	* res/res_musiconhold.c: Close the stream when file based MOH stop.
	  This won't get rid of their position in the file but it will
	  cause the translation path to be setup again. (issue #7634
	  reported by asimpson)

2006-07-31 21:14 +0000 [r38611]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: don't reissue hangup requests for SIP
	  channels that have expired their RTP timeouts (one time is
	  enough) don't rescan the SIP private structure list too fast, it
	  can cause channels to not be able to hang up (issue #7495, and
	  probably others) use ast_softhangup_nolock() since we already
	  hold the channel's lock

2006-07-31 17:09 +0000 [r38585]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Add missing code to bring transferee channel
	  out of MOH/autoservice under certain circumstance (issue #7611
	  reported by guillecabeza with minor mods by myself)

2006-07-31 04:06 +0000 [r38546-38547]  Russell Bryant <russell@digium.com>

	* frame.c: one more small tweak for thread-safety of TRACE_FRAMES

	* frame.c: Make the frame counting done with TRACE_FRAMES defined
	  thread-safe

2006-07-29 23:18 +0000 [r38501]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: How many attempts does it take to make a SIP
	  URI parser that works well? I'm up to 5 personally. On to the
	  good stuff - parse the domain first, user second, and get rid of
	  port & options/params last. (issue #7616 reported by andrew)

2006-07-28 18:49 +0000 [r38420]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make a copy of the request URI in
	  check_user_full instead of modifying the one on the structure,
	  and also strip params properly from the user portion of the SIP
	  URI so as to preserve the domain (issue #7552 reported by dan42)

2006-07-27 22:23 +0000 [r38347-38370]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_chanspy.c: use the enum that defines the option
	  arguments, so that the likelihood of mismatched option indexes is
	  reduced (which in this case was a bug, the volume argument was
	  not checked properly)

	* channel.c: do a better job avoiding translation path
	  teardown/setup when not needed

2006-07-27 04:25 +0000 [r38328]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix crash when using the "regexten" option
	  with MALLOC_DEBUG enabled. This was not reported in the bug
	  tracker but the same bug has been demonstrated in other places in
	  the code.

2006-07-27 02:43 +0000 [r38310]  Kevin P. Fleming <kpfleming@digium.com>

	* channel.c: don't do useless translation destroy/build when the
	  channel is already in the correct format

2006-07-27 01:58 +0000 [r38288]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: fix a crash when MALLOC_DEBUG is enabled and
	  the regexten is enabled. The crash would occur when the extension
	  got removed. (fixes issue #7484)

2006-07-26 15:26 +0000 [r38234]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Put default callerid into contact when the
	  one specified is either NULL or has a zero string length. (issue
	  #7590 reported by key2)

2006-07-25 19:43 +0000 [r38200]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: This resolves a deadlock that a tech support
	  customer was getting frequently when his users would answer call
	  waiting. If another thread is currently holding the zt_pvt lock
	  for the first channel, unlock both channels and let asterisk
	  retry the native bridge, just like what is done for the second
	  channel directly below these changes.

2006-07-24 17:05 +0000 [r38167]  Steve Murphy <murf@digium.com>

	* codecs/gsm/Makefile: This fixes a compile problem for s390 as
	  reported in bug 7253. Tested on both an s390 and non-s390
	  machine.

2006-07-19 17:10 +0000 [r37949]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: ensure that global 'maxauthreq' is reset to
	  zero during 'reload'

2006-07-18 00:41 +0000 [r37828-37856]  Russell Bryant <russell@digium.com>

	* frame.c: don't crash if the frame has no data, but has a src

	* frame.c: if asked to duplicate a frame that has no data, don't
	  set the frame's data pointer past the end of the allocatted
	  buffer for the new frame

2006-07-17 22:36 +0000 [r37765-37808]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* formats/format_h263.c: Backport buffer increase to 1.2

	* formats/format_h263.c: Overflow bad

2006-07-15 23:29 +0000 [r37691]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* enum.c: Bug 7513 - ensure that each time we do a query, the
	  results are returned in the same logical order, so that when we
	  iterate over the list, we get all results, not some results
	  repeated, due to insufficient sorting.

2006-07-14 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.10 released

2006-07-14 13:31 +0000 [r37612]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_sms.c: Bug 7526 - previous commit broke app_sms

2006-07-13 21:22 +0000 [r37571]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_voicemail.c: don't fail/abort if the message category
	  sound file cannot be played, just generate a warning message and
	  continue message playback

2006-07-13 18:44 +0000 [r37546]  Russell Bryant <russell@digium.com>

	* rtp.c: yeah, ummm... This frame pointer should not be static.
	  This situation only exists in 1.2 (pointed out by Constantine
	  Filin on the asterisk-dev mailing list)

2006-07-13 16:44 +0000 [r37531]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: report address of peer trying to subscribe
	  to unknown hint

2006-07-13 15:45 +0000 [r37458-37516]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* doc/README.enum: Bug 7532 - Typo in enum example

	* contrib/init.d/rc.mandrake.zaptel: Merge fixup for asterisk
	  startup script to zaptel startup script

2006-07-12 15:53 +0000 [r37441-37442]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_voicemail.c: fix a weird case where a lock file could be
	  left (but would happen almost never)

	* app.c: fix a case where ast_lock_path() could leave a
	  randomly-named lock file hanging around make ast_unlock_path
	  actually report when unlocking fails

2006-07-12 15:23 +0000 [r37439]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Add support to have maxauthreq as a global
	  option

2006-07-12 13:54 +0000 [r37417-37419]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, utils.c, res/res_agi.c, apps/app_zapras.c,
	  asterisk.c, channels/chan_modem.c, channels/chan_iax2.c: remove
	  some more bad examples of using printf

	* enum.c, pbx/pbx_config.c: get rid of some more printf's (although
	  most of these were ifdef-ed out)

2006-07-12 03:55 +0000 [r37402]  Matt O'Gorman <mogorman@digium.com>

	* app.c: GRRR no fprintf!

2006-07-11 19:00 +0000 [r37378]  Joshua Colp <jcolp@digium.com>

	* configs/iax.conf.sample, channels/chan_iax2.c: Add configuration
	  option for IAX2 users that will limit the amount of outstanding
	  AUTHREQs we are waiting for replies on.

2006-07-10 21:01 +0000 [r37361]  Kevin P. Fleming <kpfleming@digium.com>

	* channel.c: do masquerade-behind-proxy checking with better
	  control over locks

2006-07-07 23:57 +0000 [r37307]  Joshua Colp <jcolp@digium.com>

	* rtp.c: Change message regarding marker bit forcing when SSRC
	  changes to be shown only during debug so it doesn't overload high
	  capacity systems

2006-07-06 21:41 +0000 [r37224]  Matt O'Gorman <mogorman@digium.com>

	* channel.c: patch resolves issue with when to decide if its right
	  time to native bridge, feature redirect was not being checked.
	  patch from bug #7296

2006-07-06 20:38 +0000 [r37212]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_agent.c: Don't do weird things on a callback agent
	  that has attempted logoff while still on the phone.

2006-07-06 15:48 +0000 [r37173]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Instead of giving the scheduled item ID on a
	  peer expiration, give the time until they expire (issue #7455
	  reported by slavon)

2006-07-06 13:47 +0000 [r37143]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_db.c: Fix spelling/grammar (issue 7493)

2006-07-05 15:31 +0000 [r36998]  Joshua Colp <jcolp@digium.com>

	* channels/chan_oss.c: Spell extension correctly in documentation
	  for chan_oss dial (issue #7487 reported by flefoll)

2006-07-04 14:45 +0000 [r36838-36911]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Tell clients based on old SIP standard that
	  we only support MD5 digest authentication...

	* channels/chan_sip.c: issue #7470 - Need larger buffer for
	  record-route headers...

2006-07-03 05:12 +0000 [r36697-36751]  Russell Bryant <russell@digium.com>

	* asterisk.c: fix a race condition that caused asterisk to log a
	  *ton* of warnings on mac osx about poll returning an error
	  because the polled file descriptor was bad.

	* channels/chan_mgcp.c, channels/chan_phone.c,
	  channels/chan_local.c, channels/chan_misdn.c,
	  channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_agent.c, channels/chan_features.c,
	  channels/chan_h323.c, channels/chan_modem.c,
	  channels/chan_iax2.c: use ast_set_callerid to be more consistent
	  and to make sure that the "callerid" option in the conf files is
	  always handled the same way and sets ANI (issue #7285, gkloepfer)

	* dsp.c: fix the build with BUSYDETECT_TONEONLY defined (issue
	  #7414)

2006-06-30 14:05 +0000 [r36290-36377]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_directory.c: Bug 7349 - Directory did not work correctly
	  when USE_ODBC_STORAGE was defined.

	* Makefile: Bug 7388 - compatibility changes for Solaris

2006-06-29 07:19 +0000 [r36253-36254]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/queues.conf.sample: clarify documentation for
	  'persistentmembers' setting

	* configs/sip.conf.sample: add documentation for peer-specific
	  'outboundproxy' setting

2006-06-28 14:12 +0000 [r36187]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't delete scheduled item twice in
	  sip_destroy (already fixed in svn trunk)

2006-06-26 17:10 +0000 [r36078]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: ensure that two SIP channels that exist at
	  the same moment will not have the same channel names (issue
	  #7245, different fix)

2006-06-26 15:27 +0000 [r36043]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 6997 maybe, but anyway - don't
	  retransmit responses to NON-invite requests.

2006-06-25 15:10 +0000 [r35915]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Bug 7425 - Size of buffer is passed in by
	  len

2006-06-23 11:30 +0000 [r35669]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: We should lock the queue before we go making
	  changes to member interface statuses.

2006-06-21 19:25 +0000 [r35334]  Joshua Colp <jcolp@digium.com>

	* configs/indications.conf.sample: Add Venezuelan indications
	  (issue #7402 reported by palillo)

2006-06-20 15:05 +0000 [r35121]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* stdtime/private.h: Bug 7398 - Solaris puts its zoneinfo files in
	  a nonstandard place

2006-06-20 10:27 +0000 [r35058]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #6820 - Possible fix (already
	  implemented in trunk)

2006-06-19 20:27 +0000 [r34911]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Call reset_user_pw upon changing the
	  password using externpass (issue #7395 reported by Ryan Cumming)

2006-06-19 18:07 +0000 [r34875]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Issue 7357 - txt file left behind when
	  going to operator. Also, fix a possible file descriptor leak.

2006-06-18 21:03 +0000 [r34627-34655]  Russell Bryant <russell@digium.com>

	* pbx.c: don't set state to BUSY if the channel is already in the
	  UP state (issue #7376, backported from trunk)

	* configs/iax.conf.sample, channels/chan_iax2.c: don't store
	  multiple secrets delimited with semicolons for peers because this
	  is only valid for users. Instead, only keep the last specified
	  secret for a peer entry. Also, document how multiple secrets are
	  handled in the sample config. (Reported by PCadach on
	  #asterisk-bugs)

2006-06-16 03:37 +0000 [r34400]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Zero out a declared structure so as to not
	  crash if it contains invalid data (reported by Qwell on
	  #asterisk-dev)

2006-06-15 14:11 +0000 [r34306]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 7294 - patch by phsultan - Asterisk
	  sends Invite instead of BYE in some cases.

2006-06-15 13:30 +0000 [r34274]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: don't use prefixed structure names for internal
	  structures don't use a plural structure name for a singular
	  object

2006-06-15 12:40 +0000 [r34242]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: VoicemailMain exits on any key, when the
	  language is set to Italian, instead of properly handling the key
	  (issue 7353).

2006-06-14 22:22 +0000 [r33841-34160]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: coding style cleanups on queue interface
	  handling code that was committed for the last release

	* channels/chan_iax2.c: use existing dial string parser for strings
	  supplied to iax2_devicestate, because they can be complete dial
	  strings, not just device names

	* include/asterisk/plc.h, jitterbuf.c, plc.c, apps/app_dumpchan.c,
	  apps/app_chanspy.c: clarify file headers that mention disclaimer
	  usage

	* file.c: don't output 'no format found' when we _did_ find the
	  format but couldn't open the desired file for some other reason

	* apps/app_mixmonitor.c: memory allocation optimizations

2006-06-13 12:40 +0000 [r33753-33813]  Russell Bryant <russell@digium.com>

	* pbx.c: remove duplicate mutex_unlock

	* apps/app_voicemail.c: fix various places where the code returns
	  without unlocking vmlock or destroying loaded configuration

	* apps/app_festival.c: add a missing close of an open fd, destroy
	  of open config, and removal of the calling channel from the
	  localusers list

	* asterisk.c: revert a change that caused more problems than it
	  fixed and fix the real problem in this code. fds was declared as
	  an array of zero size which caused some weird problems, some of
	  which would only be seen when compiling without optimizations.
	  (fixes issues #7071, #7326, and #7305)

2006-06-12 21:34 +0000 [r33724]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/chanspy.h, apps/app_mixmonitor.c, channel.c:
	  Greatly simply the mixmonitor thread, and move channel reference
	  directly to spy structure so that the core can modify it.

2006-06-12 20:40 +0000 [r33693]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: fix a place where a frame would be free'd twice

2006-06-12 16:03 +0000 [r33638]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_local.c: only allow chan_local to masquerade the
	  outbound channel onto its owner, instead of the other way around
	  (this will ensure that group variables on the outbound channel are
	  preserved)

2006-06-12 15:27 +0000 [r33615]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_agi.c: Move set priority up, because at this point in the
	  code, stdout is no longer the console. If we're unable to set
	  priority, the error goes to Asterisk as if it were an AGI command
	  (issue 7335).

2006-06-11 21:21 +0000 [r33449-33548]  Russell Bryant <russell@digium.com>

	* pbx.c: fix another place where a frame does not get free'd

	* apps/app_meetme.c: fix up five little places where frames would
	  not be free'd and remove an unnecessary mutex_unlock where there
	  is no way for it to be locked at that time

	* apps/app_ices.c: fix a place that would leak a frame (all of
	  these fixes are in applications that call ast_read() on a channel
	  but have code paths in them that would not free the frame)

	* apps/app_festival.c: fix a couple places that would leak a frame

	* apps/app_alarmreceiver.c: fix two places that would cause a frame
	  to be leaked

	* apps/app_url.c: fix a case where an HTML frame would be leaked

	* apps/app_test.c: Free frames read from the channel when measuring
	  noise. This resulted in about 9 or 10 seconds of leaked frames in
	  both the TestClient and TestServer applications

	* apps/app_zapbarge.c, apps/app_zapscan.c: backport a couple of
	  frame leak fixes from the trunk (revisions 33446, 33447)

2006-06-09 18:52 +0000 [r33264-33300]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Allow the format outputted by meetme list to
	  be used for meetme commands (like kick) (issue #7322 reported by
	  darkskiez)

	* channels/chan_iax2.c: Remove an unneeded double lock (issue #7310
	  reported by arkadia)

	* apps/app_dial.c: Handle hangup during recording of screened name
	  (issue #7304 reported by kulldominique)

	* apps/app_meetme.c: Add missing newlines (issue #7323 reported by
	  darkskiez)

2006-06-09 15:53 +0000 [r33235]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Do not require a context on a domain=
	  setting

2006-06-08 16:57 +0000 [r33036]  Kevin P. Fleming <kpfleming@digium.com>

	* frame.c: handle out-of-memory conditions properly in
	  ast_frisolate() (reported by Slav Kenov on asterisk-dev mailing
	  list)

2006-06-07 17:53 +0000 [r32818]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: fix some broken code with
	  BRIDGE_OPTIMIZATION defined (issue #7292)

2006-06-06 16:55 +0000 [r32605]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 7287 - A too short voicemail with
	  ODBC_STORAGE will cause the first voicemail to be deleted
	  erroneously

2006-06-06 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.9.1 released

2006-06-06 16:02 +0000 [r32582]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* callerid.c: Bug 7268 - Callerid leaks memory on error

2006-06-06 15:48 +0000 [r32566]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: clean up yesterday's security fix to not
	  cause breakage when video mini frames are received

2006-06-03 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.9 released

2006-06-05 19:53 +0000 [r32373]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: ensure that the received number of bytes is
	  included in all IAX2 incoming frame analysis checks (fixes a
	  known vulnerability)

2006-06-04 03:43 +0000 [r31921]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: return bridge exit logic to what it was before
	  i broke it :-(

2006-06-03 17:02 +0000 [r31775]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: when using moh files mode, don't look for
	  a file past the number of files that have been loaded, or worse,
	  past the size of the files array

2006-06-01 21:46 +0000 [r31321-31555]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_musiconhold.c: remove pointless forcing of the channel
	  into SLINEAR mode; the write format will be set later based on
	  the file that is chosen to be played to the channel

	* include/asterisk/channel.h, channel.c: handle Zap transfers
	  behind chan_agent properly so the agent doesn't get stuck waiting
	  for the call to hang up

	* configs/sip.conf.sample: remove a sample entry that never should
	  have been added (code to support it was not merged)

2006-05-31 23:50 +0000 [r31194]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: if the connection to a FastAGI server fails
	  because of a timeout, log a more informative log message

2006-05-31 22:26 +0000 [r31161]  Kevin P. Fleming <kpfleming@digium.com>

	* rtp.c: silence a warning message that is not a warning

2006-05-31 20:26 +0000 [r31127]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: fix misplaced manager event (issue #6866,
	  flefoll)

2006-05-30 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.8 released

2006-05-30 14:55 +0000 [r30770]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: Fix infinite loop scenario and add some sanity
	  checking to prevent segfault on a NULL parameter coming in (which
	  probably shouldn't happen, but just to be safe...)

2006-05-26 17:09 +0000 [r30424-30546]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: A new way to try and deal with deadlocks that
	  occur in app_queue at present. Using this approach, we only
	  manipulate the main queue mutexes when we get a dev state change
	  on a device that is actually a member of a queue. Backported from
	  /trunk for the "bug fix".

2006-05-25 20:03 +0000 [r30373]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Don't play the enter sound twice when a person
	  joins a conference after the leader has joined it. (issue #6138
	  reported by shanermn)

2006-05-25 17:39 +0000 [r30293-30296]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/gsm/Makefile: don't try to use -march=s390 when building
	  on S/390 systems (reported via asterisk-users mailing list)

	* channels/chan_sip.c: allow SIPCHANINFO(peername) to work for
	  calls from users as well (issue #7215)

2006-05-25 15:27 +0000 [r30239]  Joshua Colp <jcolp@digium.com>

	* configs/extensions.conf.sample: Get rid of an incorrect SIP dial
	  string in the sample extensions.conf - I even tried variations...
	  no go (issue #7222 reported by arkadia)

2006-05-24 21:24 +0000 [r30069-30098]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: oops... make sure to stop processing a
	  request once we have sent an authentication challenge (issue
	  #7220)

	* channels/chan_sip.c: don't send CANCEL on a pending INVITE if we
	  haven't received a provisional response yet... mark it pending
	  until the first response is received (issue #7079)

2006-05-24 19:55 +0000 [r30037]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_meetme.c: app_meetme used the ast_max_exten instead of
	  path_max solves bug 6822

2006-05-24 19:44 +0000 [r30033-30035]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Merge branch for bug 6264 (Privacy option 2
	  returns dial-status ANSWER / option_priority_jumping not
	  respected) (reported by jkoopmann and branch by murf)

	* logger.c: Fix deadlock caused by a race condition in the logger
	  when reloading (issue #7195 reported and fixed by softins)

2006-05-24 16:59 +0000 [r29904-29973]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_agi.c: support video recording via AGI 'RECORD FILE'
	  command (issue #7068)

	* apps/app_queue.c: fix various bugs related to exiting from queue
	  via keypress and moh handling (issue #6776, different fix)

	* channels/chan_zap.c: respect 'usecallingpres' in zapata.conf even
	  if CLID has not been set for the channel (issue #7123)

	* channels/chan_sip.c, configs/sip.conf.sample: add an option to
	  allow the admin to 'hide' SIP user/peer names from systems trying
	  to 'fish' names

2006-05-23 21:44 +0000 [r29849]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: fix the sourceaddress option (issue #7213,
	  alphaque)

2006-05-23 18:16 +0000 [r29764]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: simplify/fix lock retry, and fix comment

2006-05-23 17:17 +0000 [r29733]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_sip.c: Sanity check code for an extended failure in
	  trying to obtain a channel lock that may have been obtained
	  elsewhere. Prevents the monitor thread of the SIP module from
	  going into an infinite loop, effectively, breaking SIP until you
	  restart Asterisk or the mutex is unlocked, whichever comes first.

2006-05-23 17:15 +0000 [r29732]  Kevin P. Fleming <kpfleming@digium.com>

	* dnsmgr.c, res/res_features.c, include/asterisk/linkedlists.h,
	  include/asterisk/lock.h, apps/app_sql_postgres.c, pbx.c: backport
	  some mutex initialization and linked list handling fixes from
	  trunk

2006-05-23 15:58 +0000 [r29696]  BJ Weschke <bweschke@btwtech.com>

	* res/res_features.c: Fix a potential leak and correct (hopefully)
	  a segfault under certain conditions. #6784 (vovan and perry
	  testing)

2006-05-22 21:27 +0000 [r29464-29555]  Joshua Colp <jcolp@digium.com>

	* apps/app_waitforsilence.c: Increase the silence threshold to 128
	  to "fix" it, so I'm told. (issue #6595 reported by davetroy fixed
	  by casper)

	* res/res_features.c: Use the correct language when playing the
	  transfer sound (issue #7109 reported by casper)

	* channels/chan_local.c: Preserve presentation bit when going
	  through chan_local (issue #7002 reported by acunningham)

2006-05-22 14:59 +0000 [r29394-29398]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_meetme.c: Bug 7194 - spelling fix

	* pbx.c: Bug 7196 - month range did not work

2006-05-21 15:16 +0000 [r29196]  BJ Weschke <bweschke@btwtech.com>

	* res/res_features.c: When an application that is executed via
	  applicationmap and exits non-zero, make sure that we pass through
	  the correct return value from the application to make sure a
	  segfault doesn't occur by a bridge trying to continue when it
	  should not. Also, when executing applications via applicationmap,
	  make sure that the application is executed against the channel
	  whose DTMF caused it to be fired off in the first place. (part
	  1/2 of #7090 - this is the only fix that will be applied to both
	  1.2 and /trunk) acunningham and blitzrage on testing...

2006-05-20 19:50 +0000 [r29052]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: fix the possibility of writing one byte past
	  the end of a buffer. (issue #7189, Mithraen)

2006-05-20 02:35 +0000 [r28968]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: don't allow queue member devices to ring longer
	  than the total queue timeout (issue #6423, reported and patched
	  by bcnit)

2006-05-20 02:31 +0000 [r28966]  Russell Bryant <russell@digium.com>

	* apps/app_sms.c: fix a case where code made assumptions about how
	  memory for variables is allocatted on the stack - this patch is
	  slightly different than the one that went in for the trunk

2006-05-20 00:55 +0000 [r28794-28896]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: don't try to predict where the compiler
	  will place things on the stack... put them in the right place
	  explicitly (issues #7029 and #7100, maybe others)

	* channels/chan_sip.c: use the specified 'subscribecontext' for a
	  peer rather than the context found via the target domain (domain
	  contexts are for calls, not for subscriptions) (issue #7122,
	  reported by raarts)

2006-05-19 19:18 +0000 [r28754-28790]  Russell Bryant <russell@digium.com>

	* utils/smsq.c: fix the build of smsq with -Werror. I learned
	  something new about format strings from this patch! (issue #7141,
	  Mithraen)

	* asterisk.c: This explicit poll is only needed on mac. In fact, it
	  breaks some systems such as some versions of Fedora, causing
	  'asterisk -rx' to never exit. This has been tested on systems
	  showing the asterisk -rx problem, as well as other unaffected
	  versions of linux, mac osx 10.4, and FreeBSD 6. (issue #7071)

2006-05-19 17:04 +0000 [r28627-28698]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c: Make the minidle option actually exist as
	  documented (issue #7159 reported by imran)

	* apps/app_voicemail.c: When forwarding messages use the context
	  that the active voicemail user was found in. (issue #7010)

	* enum.c: Backport of fix for issue #6654 that was fixed in trunk
	  but not here

	* apps/app_queue.c: Treat paused queue members as unreachable
	  (issue #7127 reported by peterh)

2006-05-18 20:43 +0000 [r28335-28384]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: fix up a few more places to find the SDP
	  properly (fallout from fix for #7124)

	* channels/chan_sip.c: handle incoming multipart/mixed message
	  bodies in SIP and find the SDP, if present (issue #7124 reported
	  and patched by eborgstrom, but very different fix)

	* enum.c: use unsigned counters for handling answer/IE lengths
	  while processing DNS results (issue #7174)

	* channels/chan_sip.c: support 'inactive' tag for SDP media streams
	  (simple fix, proper fix will appear in 1.4 release) (issue #7130)

2006-05-18 17:27 +0000 [r28257]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_hasnewvoicemail.c: Bug 7167 - HasNewVoicemail and
	  VMCOUNT() didn't work when USE_ODBC_STORAGE was defined

2006-05-18 16:31 +0000 [r28169-28212]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Return -1 on error in ODBC messagecount and
	  0 on success (issue #7133 reported by cfieldmtm)

	* apps/app_voicemail.c: Fix endless looping message by checking
	  value of res before doing retries stuff. (issue #7140 reported by
	  tanischen)

2006-05-18 12:13 +0000 [r28125]  Olle Johansson <oej@edvina.net>

	* apps/app_meetme.c: Video in meetme? Hmmm. Removed until we do
	  have some code for it.

2006-05-17 22:34 +0000 [r27973]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Fix codec priority stuff during
	  authentication (issue #6194 reported by jkoopmann)

2006-05-17 19:27 +0000 [r27927]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #7176 - Crash in expire_register (We
	  need to find out what's causing peer to be undefined, so this is
	  just a bandaid, not a real fix)

2006-05-17 17:07 +0000 [r27767-27847]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Priority jumping not working on VoiceMail
	  app with new syntax (issue #7164 reported and fixed by
	  alvaro_palma_aste)

	* apps/app_osplookup.c: OSPNext does not handle success/failure
	  correctly (issue #7147 reported and fixed by eborgstrom)

2006-05-17 09:21 +0000 [r27723]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: chan_sip did not use the TRANSFER_CONTEXT
	  for transfers, like res_features. Now fixed.

2006-05-17 02:19 +0000 [r27636]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 7125 - Fix race condition between
	  resequencing and leaving a message

2006-05-16 23:31 +0000 [r27594]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Inherit channel variables during call forwards
	  when going through chan_local (issue #7095 reported by raarts)

2006-05-16 20:05 +0000 [r27468]  Kevin P. Fleming <kpfleming@digium.com>

	* channel.c: don't leak frames when deferring DTMF or dropping
	  duplicate ANSWER frames (issue #7041, slightly different fix,
	  reported/patched by clausf)

2006-05-13 04:08 +0000 [r27093]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 7134 - File descriptor leak with ODBC
	  storage of voicemail

2006-05-11 23:02 +0000 [r27051]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_logic.c: Bug 7086 - pbx_checkcondition substitution,
	  so that arbitrary strings are true (for regex)

2006-05-11 09:05 +0000 [r26760-26773]  Kevin P. Fleming <kpfleming@digium.com>

	* rtp.c: backport fix from trunk for bug #6934, ensuring that RTP
	  mark bit is changed when SSRC changes

	* channels/chan_sip.c: ensure that we send a response to REGISTER
	  requests that are successfully authenticated but contain invalid
	  Contact URIs

2006-05-09 14:18 +0000 [r26050-26090]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_sip.c, doc/README.variables: Add the appropriate
	  jumping behavior that is the standard for 1.2.X to SIPGetHeader
	  that is now deprecated in /trunk. #7111 (blitzrage!!!)

	* apps/app_voicemail.c: Correct memory leak in find_user_realtime
	  #7118 (fnordian)

2006-05-08 15:09 +0000 [r25608]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 7103 - mikma - The header is named
	  "Require" - Don't reply to ACK (Not using patch against trunk)

2006-05-08 14:12 +0000 [r25518-25563]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_agent.c: Don't show agents as available when they
	  are in wrap-up time. #6726 (ZX81)

	* apps/app_queue.c: Make QueueStatusComplete event thread safe by
	  wrapping it inside the queue lock clause already there. #7013
	  (bziherl reporting)

	* apps/app_queue.c: Don't recheck valid_exit() after getting the
	  result from say_position (which already checks it). Should
	  prevent another loop if the caller hits digits during the
	  position announcement. #6776 (tgj reporting)

2006-05-08 11:16 +0000 [r25442]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Incorrect log statement when playing transfer
	  sounds (issue #7008 reported and fixed by nathan)

2006-05-07 13:38 +0000 [r25288-25322]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_meetme.c: Fix playback behavior to exit correctly when
	  we receive a hangup during playback of the invalid pin message.
	  #7091 (AntD reporting)

	* asterisk.c: Reset the value of ast_mainpid if we fork so future
	  remote unix connections display the correct PID. #7098 (tzafrir
	  reporting)

2006-05-06 02:32 +0000 [r25015-25165]  Russell Bryant <russell@digium.com>

	* frame.c: fix a problem where the frame's data pointer is
	  overwritten by the newly allocated data buffer before the data
	  can be copied from it. This is in the ast_frisolate() function
	  which is rarely used. (issue #6732, stefankroon)

	* channels/chan_zap.c: ensure that the appropriate manager events
	  are sent in all of the places where alarms are detected or
	  cleared (issue #6866, flefoll)

	* channels/chan_h323.c: update chan_h323 to reflect the new
	  prototype for rtp_set_peer (issue #6560, casper) This was fixed a
	  couple months ago in the trunk, but never in 1.2.

2006-05-05 20:44 +0000 [r25014]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_voicemail.c, include/asterisk/app.h, app.c: Voicemail
	  fixes along with an API change approved by russellb to fix the
	  bug(s). (jcollie and supczinskib) #7064

2006-05-05 17:39 +0000 [r24837-24911]  Russell Bryant <russell@digium.com>

	* apps/app_while.c, apps/app_macro.c: use pbx_checkcondition()
	  instead of ast_true() to evaluate the condition for MacroIf and
	  WhileIf (issue #7086)

2006-05-04 16:27 +0000 [r24706]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_queue.c: Bug 7023 - reload should not unpause members

2006-05-04 11:17 +0000 [r24567-24669]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_verbose.c: Make sure that only the "|" is a recognized
	  delimiter for Verbose(), as the app documentation already
	  specifies. #7080 (alessiof reporting)

	* apps/app_dial.c: Correct application documentation to make users
	  aware that certain options cannot be used in conjunction with
	  others. #6666 (chotaire)

2006-05-03 18:31 +0000 [r24496]  Russell Bryant <russell@digium.com>

	* redhat/asterisk.spec: fix up "make rpm" - don't reference the
	  gzipped man page, because we don't store them compressed anymore
	  - add some files that currently were not listed (issue #6837)

2006-05-03 12:39 +0000 [r24381]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #7074 - Problem with long contact
	  lines

2006-05-02 19:39 +0000 [r24295]  BJ Weschke <bweschke@btwtech.com>

	* file.c: Make certain ast_stopstream() sets the channel's stream
	  members to NULL after closing them. #7067 (jcomellas)

2006-05-02 02:12 +0000 [r24019-24097]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_privacy.c: Prompt does not request '#' to end input, so
	  the application should not require it

	* apps/app_nbscat.c, apps/app_festival.c, apps/app_mp3.c,
	  apps/app_zapras.c, asterisk.c, apps/app_externalivr.c,
	  apps/app_ices.c, res/res_musiconhold.c,
	  include/asterisk/options.h: Bug 6864 - drop realtime priority on
	  ALL external processes

2006-05-01 19:34 +0000 [r23985-23988]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_voicemail.c: Make sure that when someone 0's out while
	  recording a msg and then chooses to DELETE the recorded file, the
	  .txt file isn't left around by itself to cause problems later.
	  #7061 (dimitripietro reporting, blitzrage confirmed)

2006-05-01 15:12 +0000 [r23951]  Russell Bryant <russell@digium.com>

	* pbx.c: add missing locking of the dialplan functions list in the
	  "show functions" CLI command

2006-05-01 10:45 +0000 [r23305-23899]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_skel.c: fix this to actually compile so people can learn
	  from it

	* cdr/cdr_sqlite.c: eliminate compiler warning

	* channels/chan_iax2.c: remove a pointless comparison, since the
	  buffer is smaller than the length being checked for

	* Makefile, editline/configure, cdr/Makefile, channels/Makefile,
	  db1-ast/Makefile: allow top-level OPTIMIZE setting to affect
	  builds in these subdirectories too

	* Makefile: let the compiler determine whether hardware or software
	  floating point should be used (like we do in the editline
	  subdirectory)

	* Makefile, apps/Makefile: remove extraneous -m64 flag that is not
	  needed remove old 'look' target which is no longer needed (these
	  are coming from Debian patches <G>)

	* editline/makelist: ensure that the script output is correctly
	  generated when the system's character set does not use the
	  English lowercase/uppercase character groups

	* Makefile: do installation in subdirs as a separate target (so
	  external modules can use the Makefile more easily) generate final
	  messages -after- running any post-install script that may be
	  present

2006-04-28 16:40 +0000 [r23176]  Russell Bryant <russell@digium.com>

	* configs/zapata.conf.sample, configs/mgcp.conf.sample,
	  configs/sip.conf.sample: note that group assignments must be from
	  0 to 63 (issue #7048)

2006-04-27 19:11 +0000 [r22954]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Queue(somequeue,,,,) -> interpreted as
	  Queue(somequeue,,,,0) (issue #7044 reported nathan fixed by
	  jsmith - sort of)

2006-04-27 16:12 +0000 [r22866]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Fix buglet in channel reassignment on
	  SETUP_ACK

2006-04-26 19:18 +0000 [r22596]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c: do not allow for users to forward voicemail
	  to themselves, patch from 7001

2006-04-21 22:39 +0000 [r22112-22113]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channel.c: Bug 7004 - release all threads waiting on a condition
	  prior to freeing it

2006-04-19 21:10 +0000 [r21638]  Kevin P. Fleming <kpfleming@digium.com>

	* contrib/scripts/safe_asterisk.8, contrib/scripts/safe_asterisk:
	  support system-specific scripts in safe_asterisk, before starting
	  Asterisk proper

2006-04-19 18:43 +0000 [r21597]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* cdr/cdr_odbc.c: Bug 6553 - plug memory leaks when ODBC connection
	  is down

2006-04-18 23:31 +0000 [r21237]  Kevin P. Fleming <kpfleming@digium.com>

	* pbx.c: properly handle brace-wrapped strings in variable/function
	  references in the dialplan

2006-04-18 06:26 +0000 [r20966-21037]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_random.c: Bug 6984 - off by one error in Random()

	* res/res_musiconhold.c: Bug 6544 - when we remove a music class,
	  the thread servicing it should die

2006-04-14 17:21 +0000 [r20034-20037]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds.txt: uncomment files that actually do exist (oops)

	* sounds.txt: update text to match actual prompts being distributed
	  (thanks to Kinsey in the support department for reviewing all the
	  prompts!)

2006-04-13 20:37 +0000 [r19891]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 6947 - Allow vm broadcasts to more than
	  256 characters worth of mailboxes

2006-04-13 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.7.1 released

2006-04-13 17:40 +0000 [r19812]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_page.c: oops... let's not set a variable and then
	  immediately overwrite it while assuming its old value will
	  magically return

2006-04-13 15:56 +0000 [r19768]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* pbx.c: Bug 6957 - variable names beginning with CALLERID weren't
	  substituted correctly

2006-04-12 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.7 released

2006-04-11 22:39 +0000 [r19394-19397]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_dial.c: Bug 6490 - telco intercept should report
	  NOANSWER instead of CHANUNAVAIL

	* apps/app_voicemail.c: Bug 6061 - Fix ODBC storage of VM on PGSQL
	  and MSSQL

2006-04-11 21:58 +0000 [r19353]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: don't create a 'voicemail' symlink in the sounds
	  directory; app_voicemail has not needed it since January of 2005
	  (issue #6613)

2006-04-11 21:55 +0000 [r19351]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* asterisk.c: Bug 6097 - possible descriptor leak

2006-04-11 21:50 +0000 [r19345-19348]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_page.c: don't call the originating device as part of the
	  Page() operation (issue #6932)

	* channel.c: simplify spy queue flushing logic, and always force a
	  flush when one side gets full, even if the other side is not
	  empty (issue #6457)

	* pbx/pbx_config.c: don't destroy the entire dialplan during
	  'reload', just atomically replace it like 'extensions reload'
	  does (issue #6047)

2006-04-11 20:46 +0000 [r19303]  Joshua Colp <joshnet@nbnet.nb.ca>

	* include/asterisk/linkedlists.h: Minor linked lists bug fix. When
	  you're dealing with swapping entries around a lot it can cause a
	  seg fault.

2006-04-11 20:11 +0000 [r19301]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dial.c: handle call time limit properly when warning is
	  requested _after_ call would hae already ended (issue #6356)

2006-04-11 01:05 +0000 [r18866-19008]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_voicemail.c, app.c: When using the silence detector in
	  ast_play_and_record() and ast_play_and_prepend(), the truncation
	  code never gets called to remove the detected silence, because
	  the value of res is zero when control gets to that point. #6903
	  w/some mods (softins)

	* res/res_features.c: Don't say that we can pass an 'exten'
	  argument in the documentation of Park() when we really cannot.
	  #6902 (opsys)

2006-04-08 19:20 +0000 [r18436-18494]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 6914 - .txt file fails to rename on
	  operator out

	* formats/format_jpeg.c: Bug 6913 - fix for possible buffer
	  overflow

2006-04-07 14:16 +0000 [r18250-18260]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Fix cause codes - Add cause code for
	  incompatible formats

	* channels/chan_sip.c: - Fix possible minor memory leak in chan_sip
	  - Return proper cause code on memory allocation error

2006-04-06 22:15 +0000 [r18087-18089]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_meetme.c: fix typo

	* apps/app_meetme.c: small fix... don't try to check conference
	  details if it couldn't be created or found

	* apps/app_meetme.c: don't try to support 'i' or 'r' options if
	  chan_zap is not loaded, and warn the user when they attempt to
	  use them (issue #6675) update application help text to more
	  clearly define when Zaptel and chan_zap are required

2006-04-06 17:24 +0000 [r17945]  Russell Bryant <russell@digium.com>

	* apps/app_alarmreceiver.c: move continue out of block that checks
	  verbose level (issue #6880)

2006-04-06 17:00 +0000 [r17702-17905]  Joshua Colp <joshnet@nbnet.nb.ca>

	* pbx.c: Unlock channel on failure so that ast_mutex_destroy
	  doesn't throw a fit (issue #6647 reported by casper)

2006-04-05 06:50 +0000 [r17335-17489]  Olle Johansson <oej@edvina.net>

	* CREDITS, enum.c: Issue #6654: Enum crash on ADDRESS record,
	  possibly bad record, but still a crash

	* channels/chan_zap.c: Issue #6878 - Unhide DNDstate manager events
	  (thanks casper)

	* apps/app_queue.c: Issue #6882 - move "res=-1" out of verbosity
	  block, minor code cleanups (casper)

2006-04-04 15:24 +0000 [r17283]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_senddtmf.c: Adds documentation to show what the w flag.
	  Patch from Ian Kinner at Digium.

2006-04-03 20:38 +0000 [r17074-17150]  Olle Johansson <oej@edvina.net>

	* configs/features.conf.sample: Issue 6870 - document that parking
	  lots need to be numeric

	* channels/chan_sip.c: Issue #6848 take two - Use the tag provided
	  by the SUBSCRIBE request when sending NOTIFY

	* channels/chan_sip.c: Ugly patch to avoid hangup causes in
	  non-final responses

2006-03-31 19:11 +0000 [r16744-16771]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: move a NULL check to before the first time
	  the pointer is dereferenced (issue #6832)

	* channels/chan_iax2.c: fix the situation where bindport is
	  specified but bindaddr is not (issue #6616)

2006-03-31 18:24 +0000 [r16742]  Kevin P. Fleming <kpfleming@digium.com>

	* pbx.c: ensure that hint watchers (subscribers) cannot be added or
	  removed while the dialplan is being modified

2006-03-30 22:56 +0000 [r16579-16581]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Bug 6853 - Manager fixes: 1) extra ActionID,
	  2) missing colon

	* asterisk.c: Bug 6849 - trivial typo fix

2006-03-30 21:44 +0000 [r16534-16559]  Joshua Colp <joshnet@nbnet.nb.ca>

	* codecs/gsm/Makefile: Add another check for 64-bit goodness (issue
	  #6850 reported by evilbunny)

	* res/res_musiconhold.c: Do not exceed the array size for maximum
	  allowed moh files. (issue #6842)

2006-03-30 01:34 +0000 [r16303-16346]  Olle Johansson <oej@edvina.net>

	* res/res_features.c: Set initial value on adsipark

	* apps/app_groupcount.c: Typo fix.

	* configs/extensions.conf.sample: Typo (Issue 6839, casper)

2006-03-29 19:11 +0000 [r16082-16192]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk/pbx.h, apps/app_stack.c, pbx.c: Bug 6830 - Let
	  GosubIf work with the same conditions as a GotoIf (change in API
	  approved by Russell)

	* pbx.c: Bug 6835 - Updates to GotoIf help text

2006-03-29 04:15 +0000 [r16008]  Russell Bryant <russell@digium.com>

	* strcompat.c: tell unsetenv for solaris to return the result of
	  the setenv call

2006-03-29 00:58 +0000 [r15898]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #6823 - Portability issue with the
	  registration port number patch from yesterday. Be compatible with
	  more systems than OS/X :-) Thanks Rizzo for the advice.

2006-03-29 00:32 +0000 [r15896]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/linkedlists.h: ensure that list traversal loops
	  which skip entries properly update the 'previous entry' pointer
	  so when entries _are_ removed the list does not get damaged

2006-03-28 20:22 +0000 [r15703-15743]  Russell Bryant <russell@digium.com>

	* agi/Makefile, strcompat.c, astmm.c: backport astmm + sparc fixes
	  from the trunk

	* channels/chan_iax2.c: fix Bus Error on sparc (issue #6354)

2006-03-28 19:07 +0000 [r15699]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Fix breakage of NAT support for peers with
	  qualify=yes. Thanks Damin for access to your system, sorry folks.

2006-03-28 18:09 +0000 [r15658]  Russell Bryant <russell@digium.com>

	* pbx/pbx_ael.c: fix the order in which for loops are expanded
	  (issue #6810)

2006-03-28 17:48 +0000 [r15615]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/init.d/rc.redhat.asterisk: Bug 6815 - Adding quotes to
	  make bash happy

2006-03-27 23:45 +0000 [r15366-15381]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #6736 - Use flags for OPTION messages.
	  Thanks Casper!

	* channels/chan_sip.c: Issue #6597 - sip show registry shows
	  incorrect port

	* channels/chan_sip.c: Issue #6409 - Use "s" extension when there's
	  no username in the URI

2006-03-26 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.6 released

2006-03-25 05:07 +0000 [r14821-14868]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/init.d/rc.redhat.asterisk: Bug 6601 - More configuration
	  abilities for the RH init script

	* apps/app_voicemail.c: Fix incorrect size of zeroing (left over
	  from when maxmsg was hardcoded at 100)

	* apps/app_voicemail.c: Bug 6783 - When context is specified,
	  voicemail should look for mailboxes in that context

2006-03-24 14:48 +0000 [r14704]  Russell Bryant <russell@digium.com>

	* image.c: use the correct variable in an error message (issue
	  #6791)

2006-03-24 04:53 +0000 [r14610-14659]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_voicemail.c: Fix a typo in the app description

	* include/asterisk/sched.h: Doxygen comment typo corrections

2006-03-23 21:51 +0000 [r14523]  Joshua Colp <joshnet@nbnet.nb.ca>

	* res/res_features.c: Issue #6764 - Return BUSY signal when other
	  party is busy at Attended Transfer (Reported by mnachev)

2006-03-23 21:44 +0000 [r14522]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Fix SETUP_ACK handling so that we change
	  channels if so requested

2006-03-23 20:43 +0000 [r14467]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_meetme.c: Bug #5884 - fix a possible race state in
	  app_meetme when a channel has gone away and we are reading
	  continuously for more frames. (mneuhauser)

2006-03-23 20:13 +0000 [r14462]  Russell Bryant <russell@digium.com>

	* apps/app_readfile.c: don't crash when asked to read from a file
	  that doesn't exist (issue #6786)

2006-03-22 22:18 +0000 [r14191-14276]  Joshua Colp <joshnet@nbnet.nb.ca>

	* apps/app_voicemail.c: Fix a minor code issue

	* apps/app_voicemail.c: Issue #6781 - Verbose levels not enforced
	  in app_voicemail (Reported by flobi)

	* include/asterisk/cdr.h, cdr.c: Issue #5918 - Disposition showing
	  FAILED even though call is answered successfully (Reported by
	  tracinet)

	* pbx.c: Issue #6780 - ast_pbx_outgoing_cdr_failed description fix.
	  (Reported and fixed by casper)

2006-03-22 09:10 +0000 [r14140]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #6766 - fix ;user=phone functionality.
	  (Reported by alein, fix by russell - thanks!)

2006-03-21 18:59 +0000 [r13814-13964]  Russell Bryant <russell@digium.com>

	* configs/features.conf.sample: add a note explaining how to set
	  the DYNAMIC_FEATURES variable to allow the use of custom features
	  (issue #6747)

	* res/res_features.c: fix crash when using the ParkAndAnnounce
	  application. When using this application, there will be no peer
	  channel to play the parking announcement to. (issue #6756)

	* funcs/func_strings.c: fix REGEX on strings that contain quotes
	  (issue #6678)

	* sounds.txt: fix spelling of whiskey

	* apps/app_meetme.c: don't add conference participant if the user
	  hangs up while recording their name (issue #6661)

	* sample.call: re-add the Account parameter to the sample call file
	  since it's not really deprecated since the CDR function is no
	  longer built in

2006-03-21 06:24 +0000 [r13707-13748]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 6714 - Workaround to avoid retrieving
	  incomplete voicemail message

	* editline/term.c: Do away with some warnings and fix some
	  indentation

2006-03-20 17:36 +0000 [r13634]  Olle Johansson <oej@edvina.net>

	* channels/chan_iax2.c: Do not overwrite ANI if it's set by IE
	  (sendani=yes in the peer)

2006-03-19 09:59 +0000 [r13550]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c: revert the change made in revision 12927 in
	  favor of keeping the original behavior of the option. The
	  documentation has now been updated to reflect the actual
	  behavior. (issue #6523)

2006-03-19 09:25 +0000 [r13547]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Reset global_rtautoclear at sip reload

2006-03-16 20:05 +0000 [r13279]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* ast_expr2.y, ast_expr2.c: Bug 6737 - Fix compile warning on OS X

2006-03-16 17:58 +0000 [r13239]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Issue #6690 - clarify progressinband
	  default setting

2006-03-16 17:42 +0000 [r13237]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: always use the callerid signalling method
	  set in the zt_pvt strucutre as opposed to the last one read from
	  the config file (issue #6734, with mods)

2006-03-16 06:56 +0000 [r13197]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: To quote giant developers: "Oops". Thanks,
	  Tony!

2006-03-15 22:16 +0000 [r13161]  Russell Bryant <russell@digium.com>

	* cdr.c: - remove some calculations that will always result in 0 -
	  if a CDR was never started, don't try to calculate a duration and
	  consider it failed

2006-03-15 13:01 +0000 [r13026]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #6728: Remove parameters to Event:
	  header on SUBSCRIBE requests

2006-03-14 18:41 +0000 [r12925-12927]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c: when using the G() option to Dial, fix sending
	  the called channel to 1 priority beyond what was specified (issue
	  #6523)

	* apps/app_queue.c: fix a problem with not loading realtime queue
	  members by always reloading a realtime queue from the database
	  even if it is found in the list (issue #6680)

2006-03-12 19:26 +0000 [r12646]  Russell Bryant <russell@digium.com>

	* pbx.c: add locking to protect the list of global dialplan
	  variables

2006-03-12 17:57 +0000 [r12577]  Russell Bryant <russell@digium.com>

	* codecs/gsm/Makefile: fix build on parisc (issue #6704)

2006-03-10 12:13 +0000 [r12477-12495]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #5937 - Make sure SIP CANCEL's are
	  re-transmitted

	* channels/chan_sip.c: Issue #6576 - SIP_CODEC not used for early
	  media (reported by gpapadop73)

2006-03-08 10:51 +0000 [r12458]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #6657 - Ignore 183 session progress
	  without SDP

2006-03-07 00:05 +0000 [r12161-12195]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Bug 6020 - Race condition where packet could
	  be lost if first packet on list is acked

	* editline/np/vis.c, editline/readline.c: Bug 6664 - More fixes for
	  Solaris

2006-03-06 14:23 +0000 [r12036-12072]  Olle Johansson <oej@edvina.net>

	* channel.c: Revert earlier change

	* channel.c: Fix for astmm compilation

2006-03-06 02:32 +0000 [r11946]  Russell Bryant <russell@digium.com>

	* configs/zapata.conf.sample: fix a typo in the description of the
	  ringtimeout option

2006-03-05 12:40 +0000 [r11849]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Clear page2 flags at reload too

2006-03-04 11:45 +0000 [r11778]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_mixmonitor.c: Substitute variables in the post_process
	  string (if it exists) before those variables could possibly
	  disappear (channel hangup) #6462

2006-03-03 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.5 released

2006-03-03 00:38 +0000 [r11607-11635]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* Makefile: Bug 6638 - Use POSIX command for Solaris

	* build_tools/make_build_h: Bug 6638 - Change from a historic BSD
	  command to a POSIX command for determining username

	* asterisk.c: Bug 6637 - Fixes for Solaris

	* Makefile: If debugging, the frame pointer is helpful

2006-03-02 19:05 +0000 [r11528-11561]  Russell Bryant <russell@digium.com>

	* res/res_monitor.c: fix inaccurate ack message to ChangeMonitor
	  action (issue #6630)

	* asterisk.sgml: make the terminology used in the synopsis match
	  the option description

	* asterisk.sgml: add the -L option to the synopsis on the man page

2006-03-01 17:41 +0000 [r11479-11503]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* cdr/cdr_manager.c, cdr/cdr_tds.c, res/res_config_odbc.c,
	  include/asterisk/cdr.h, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, cdr.c:
	  Bug 6615 - Fix 64bit conversion errors by using a long int

	* build_tools/make_svn_branch_name: Bug 6618 - Solaris
	  compatibility fix

2006-02-28 19:46 +0000 [r11382-11410]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: fix the output that indicates whether
	  qualify smoothing is on or not (issue #6608)

	* asterisk.c: adjust the keys directory when astvarlibdir is
	  specified in asterisk.conf (issue #6602)

	* res/res_agi.c: add a missing newline in the agi app description
	  (thanks wunderkin!)

2006-02-27 15:20 +0000 [r11250-11281]  Russell Bryant <russell@digium.com>

	* cli.c: don't try to print the help text for a CLI command when
	  RESULT_SHOWUSAGE is returned if there is no help text available
	  (issue #6604)

	* channels/chan_sip.c: fix finding realtime peers that are not
	  dynamic by ip address (issue #6093)

	* channel.c: don't hang up the channel if its state is set to UP
	  before we return from ast_call (issue #6569)

2006-02-26 16:26 +0000 [r11165]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk/logger.h, logger.c: Bug 5950 - reenable queue
	  log rotation; also, eliminate redundant code

2006-02-25 19:54 +0000 [r11120]  Matt Frederickson <creslin@digium.com>

	* translate.c: Backport of fix to translation optimizations. Thanks
	  again file!

2006-02-25 05:08 +0000 [r11058-11089]  Kevin P. Fleming <kpfleming@digium.com>

	* translate.c: factor the number of translation steps required into
	  translation path decisions, so that equal cost paths that require
	  fewer translations are preferred

	* translate.c: reformat code to fit guidelines remember which
	  translation paths are multi-step paths

	* channel.c: ensure that spy frame queueing is able to deal with
	  translation failing for any reason (issue #6546)

2006-02-23 23:06 +0000 [r10952]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* Makefile: set PWD properly

2006-02-23 14:57 +0000 [r10736-10863]  Kevin P. Fleming <kpfleming@digium.com>

	* dnsmgr.c, include/asterisk/linkedlists.h: backport list handling
	  fix from trunk (solves memory leak problem in cdr variables and
	  device state watchers) remove unused variable to silence
	  compiler warning

	* configs/iax.conf.sample: add comment warning people about trying
	  to use hostnames/IPs in the sample config

2006-02-20 23:01 +0000 [r10577]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* app.c: Would be nice to tell people to look in the right file to
	  increase a constant

2006-02-20 06:17 +0000 [r10511-10535]  Mark Spencer <markster@digium.com>

	* channels/chan_sip.c: Handle ACKing properly (remove gratuitous
	  -1)

	* channels/chan_iax2.c: Fix numerous places in jitter buffer where
	  freed memory is referenced

2006-02-19 18:29 +0000 [r10462-10487]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* formats/format_sln.c: Okay, fseek doesn't return an offset

	* apps/app_voicemail.c: Fix possible lack of initialization of
	  useadsi

	* formats/format_sln.c: Bug 6539 - Division by two negates error
	  flag

2006-02-18 00:17 +0000 [r10409]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* app.c: Bug 6529 - memory leak in ast_play_and_prepend

2006-02-17 01:55 +0000 [r10301-10368]  Russell Bryant <russell@digium.com>

	* jitterbuf.c: fix incorrent index calculation for jitterbuffer
	  history (issue #6517)

	* apps/app_voicemail.c: when executing the Directory application
	  from voicemail and a context is not specified, use the "default"
	  context, not the channel's current context (issue #6507)

2006-02-15 01:21 +0000 [r10108-10137]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_agent.c: ensure that agents logged in via the
	  manager interface are stored in the persistence database (related
	  to issue #6301)

	* funcs/func_enum.c: handle longer ENUM lookup results (issue
	  #6476)

	* res/res_agi.c: ensure that FastAGI launcher can handle system
	  call interruption (issue #6449)

2006-02-14 20:56 +0000 [r10021]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_meetme.c: bug fix from 6485 with musiconhold not being
	  turned off by app_meetme

2006-02-14 20:20 +0000 [r10018]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: don't double-increment abandon counter for
	  calls that are hung up while dialing members (issue #6289)

2006-02-14 19:11 +0000 [r9990]  Mark Spencer <markster@digium.com>

	* apps/app_meetme.c: Fix stopstream in menus (bug #6137)

2006-02-14 18:50 +0000 [r9961-9964]  BJ Weschke <bweschke@btwtech.com>

	* asterisk.c: #ifdef the include too.

	* asterisk.c: #ifdef'd the prctl fix to only try and compile on
	  linux systems. Thanks rizzo for pointing this out.

2006-02-14 18:30 +0000 [r9953-9958]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: when answering INVITE, don't send codecs the
	  peer didn't offer (issue #6052)

	* rtp.c: revert yesterday's temporary fix for issue #6052

2006-02-14 04:45 +0000 [r9861-9870]  BJ Weschke <bweschke@btwtech.com>

	* asterisk.c: Fixed my silly backport error from r9861

	* asterisk.c: Merged changes from r9844 from /trunk. Make sure that
	  PR_SET_DUMPABLE is set to make certain that we still dump core if
	  Asterisk has setuid'd to run as non-root.

2006-02-14 00:46 +0000 [r9818]  Kevin P. Fleming <kpfleming@digium.com>

	* rtp.c: don't try to use peer's dynamic codec numbers, it leads to
	  duplication (issue #6052)

2006-02-13 17:37 +0000 [r9756]  Josh Roberson <josh@asteriasgi.com>

	* apps/app_meetme.c: Don't set the formats before we stop
	  indications. (issue #6380)

2006-02-11 19:23 +0000 [r9581-9609]  Russell Bryant <russell@digium.com>

	* channels/chan_mgcp.c, channels/chan_sip.c, pbx/pbx_dundi.c,
	  channels/chan_iax2.c: fix memory leak from not destroying the
	  scheduler context on module unload

	* apps/app_page.c: fix due to CDR changes

	* manager.c, pbx/pbx_spool.c, include/asterisk/channel.h,
	  include/asterisk/pbx.h, include/asterisk/manager.h, channel.c,
	  pbx.c: now that CDR is a loadable module, don't depend on it
	  elsewhere (issue #6460)

2006-02-11 15:22 +0000 [r9528]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c, cdr.c: clean up my mess from thread-starting
	  change

2006-02-11 06:29 +0000 [r9493]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_sip.c: kpfleming's fix from r9472 backported to 1.2

2006-02-10 20:38 +0000 [r9404]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_mgcp.c, dnsmgr.c, channels/chan_sip.c,
	  devicestate.c, channels/chan_modem.c, cdr.c: don't create monitor
	  threads in detached mode, when we need to be able to
	  pthread_join() them later if the module is unloaded (solve
	  crash-on-unload problem for these channel modules)

2006-02-09 21:10 +0000 [r9323-9326]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Revert behavior change from previous commit
	  (fixes only)

	* apps/app_voicemail.c: Backport 5929 to 1.2

2006-02-09 02:31 +0000 [r9246-9262]  Russell Bryant <russell@digium.com>

	* apps/Makefile: add another location for postgresql headers (issue
	  #6419)

	* channels/chan_iax2.c: reload peercontext on iax2 reload (issue
	  #6442)

2006-02-08 22:34 +0000 [r9233]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* cdr/Makefile: Leave it to RH/CentOS to put the freetds headers in
	  a completely nonstandard location.

2006-02-08 22:12 +0000 [r9232]  Matt O'Gorman <mogorman@digium.com>

	* logger.c, channels/chan_oss.c: Make logger report
	  error,warning,notice if logger.conf not found, also updated
	  chan_oss to give correct error message if its config file is not
	  found.

2006-02-05 17:10 +0000 [r9156]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_macro.c: Bug 6176 - Fix race condition

2006-02-02 18:37 +0000 [r9086]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: don't override ASTERISKVERSIONNUM to 000000 for non-svn
	  builds

2006-02-02 16:12 +0000 [r9073]  Matt Frederickson <creslin@digium.com>

	* res/res_odbc.c: Fix for (#6309), potential (highly unlikely)
	  memory leak in res_odbc

2006-01-30 Russell Bryant <russell@digium.com>

	* Asterisk 1.2.4 Released

2006-01-30 17:08 +0000 [r8905]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: disable buggy PRI user-user code until it
	  can be fixed

2006-01-28 13:52 +0000 [r8808]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 6182 - Don't remove scheduled event
	  until it's really done. (reported by malverian)

2006-01-27 08:02 +0000 [r8785]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 6362 - Register without Contact: and
	  Expires: fails (reporter: op)

2006-01-27 00:52 +0000 [r8758]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* ast_expr2.h, ast_expr2f.c, ast_expr2.c: Bug 6072 - Revisions to
	  the source bison and flex files don't auto-regenerate these files

2006-01-26 19:42 +0000 [r8729]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: fix problem with dtmf on e&m (issue #6364)

2006-01-26 14:39 +0000 [r8710]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 5898: Registrations does not get
	  deleted if there's an active SIP dialog

2006-01-25 19:14 +0000 [r8666-8677]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: don't call ast_update_realtime with
	  uninitialized variables if we get a registration with an expirey
	  of 0 seconds (issue #6173)

	* channels/chan_features.c: fix memory leak (inspired by issue
	  #6351)

2006-01-25 Russell Bryant <russell@digium.com>

	* Asterisk 1.2.3 Released

2006-01-25 09:46 +0000 [r8632]  Olle Johansson <oej@edvina.net>

	* channel.c: Issue #6439 - the "timebomb" bug. Patch by Markster
	  over GPRS

2006-01-25 05:38 +0000 [r8619]  Russell Bryant <russell@digium.com>

	* utils/astman.c: don't leak almost 200 bytes for each new channel
	  (issue #6330)

2006-01-25 01:50 +0000 [r8608]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dial.c: ensure hangup cause code is handled properly
	  when channel does not return a frame (issue #6346)

2006-01-24 22:55 +0000 [r8600]  Russell Bryant <russell@digium.com>

	* asterisk.c: completely arbitrary whitespace change for testing
	  something with svnmerge ...

2006-01-24 22:32 +0000 [r8588]  Kevin P. Fleming <kpfleming@digium.com>

	* channel.c: ensure that channel cannot become zombie after we
	  check but before we try to start indications

2006-01-24 20:37 +0000 [r8573]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Backport fix for #6229, hangup on polarity
	  reversal

2006-01-24 19:21 +0000 [r8537-8562]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 6114: Don't hangup on BYE/ALSO with no
	  channel.

	* channels/chan_sip.c: Issue #6308 - never send response to ACK.
	  (Reported by whiskerp)

2006-01-22 19:03 +0000 [r8437-8445]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: fix memory leak from not freeing the queue
	  member list when freeing an old queue

	* channel.c: fix MixMonitor crash (issue #6321, probably others)

2006-01-22 15:13 +0000 [r8433]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_sip.c: Bug fix: Correct some scenarios where
	  CALL_LIMIT could not be getting adjusted properly allowing
	  chan_sip to send calls when it really shouldn't. Bug #6111

2006-01-22 08:52 +0000 [r8429]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Bug 6281 - Cannot set more than a single
	  header with SIPAddHeader

2006-01-22 02:05 +0000 [r8412-8418]  Russell Bryant <russell@digium.com>

	* pbx.c: add a modified fix to prevent writing outside of the
	  provided workspace when calculating a substring (issue #6271)

	* pbx.c: temporarily revert substring fix pending the result of the
	  discussion in issue #6271

	* pbx.c: prevent the possibility of writing outside of the
	  available workspace (issue #6271)

2006-01-21 18:29 +0000 [r8394]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_queue.c: Bug 5936 - AddQueueMember fails on realtime
	  queue, if queue not yet loaded

2006-01-20 18:34 +0000 [r8347]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: fix invalid value of prev_q (issue #6302)

2006-01-20 01:00 +0000 [r8320]  Matt O'Gorman <mogorman@digium.com>

	* channels/chan_iax2.c: solved problem with delayreject and iax
	  trunking bug 4291

2006-01-19 19:40 +0000 [r8281]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Enable "musicclass" setting for sip peers as
	  per the config sample.

2006-01-19 19:14 +0000 [r8276]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* ast_expr2.y, ast_expr2.fl: Bug 6072 - Memory leaks in the
	  expression parser

2006-01-19 04:56 +0000 [r8232-8242]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: fix Message-Account header to use the ip
	  address if the fromdomain isn't set (issue #6278)

	* apps/app_milliwatt.c: fix a seg fault due to assuming that space
	  gets allocatted on the stack in the same order that we declare
	  the variables (issue #6290)

2006-01-18 21:02 +0000 [r8194]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_meetme.c: Solves issue with the login proccess in meetme
	  patch from 6136

2006-01-18 02:49 +0000 [r8173]  Russell Bryant <russell@digium.com>

	* ChangeLog (removed): remove ChangeLog from the 1.2 branch. It
	  will only be present in the tags.

2006-01-18 Russell Bryant <russell@digium.com>

	* Asterisk 1.2.2 Released

2006-01-18 00:47 +0000 [r8140-8162]  Matt O'Gorman <mogorman@digium.com>

	* loader.c: Changed order of autoload so that pbx_ comes before
	  channels, and in doing so cause bug 6002 to not be an issue

	* apps/app_festival.c: Stop any generators running on a channel
	  when festival is called as described in 5996

2006-01-17 18:29 +0000 [r8134]  Matt Frederickson <creslin@digium.com>

	* res/res_features.c: Backport of fix for #6094

2006-01-17 16:55 +0000 [r8124]  Matt O'Gorman <mogorman@digium.com>

	* logger.c: Fixed code ordering of logger_init and queue_log_init
	  bug 6263

2006-01-17 13:11 +0000 [r8112-8122]  Kevin P. Fleming <kpfleming@digium.com>

	* asterisk.c: update CLI copyright notice

	* asterisk.c: do rlimit check _after_ reading config file, in case
	  'dumpcore' is specified there

2006-01-14 19:06 +0000 [r8074]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_strings.c: Bug 6238 - Fix segfault when delimiter not
	  specified

2006-01-13 06:07 +0000 [r8047]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: fix spelling errors (issue #6227)

2006-01-12 06:14 +0000 [r7999]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, configs/voicemail.conf.sample: Bug 6211 -
	  Add option deletevoicemail as equivalent to option delete for
	  Realtime

2006-01-11 19:08 +0000 [r7965-7986]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: move variable to correct scope (issue
	  #6197)

	* apps/app_voicemail.c: fix temp greetings with ODBC storage (issue
	  #6078)

	* channels/chan_sip.c: fix mem leak on module unload (issue #6190)

	* app.c: don't override an error condition that occurred when
	  acting on the primary channel when stopping the autoservice on
	  the peer channel. (from issue #6087)

	* translate.c: lock list of translators *before* recalculating the
	  translation matrix

2006-01-11 04:38 +0000 [r7963]  Matt O'Gorman <mogorman@digium.com>

	* channel.c: Minor typo refrenced in 6191

2006-01-11 04:19 +0000 [r7957-7960]  Russell Bryant <russell@digium.com>

	* pbx.c: fix locking error - lock instead of unlock

	* apps/app_dial.c: fix a little typo

2006-01-11 01:30 +0000 [r7955]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 6192 - behave correctly when mailbox is
	  specified as argument

2006-01-10 08:48 +0000 [r7939]  Olle Johansson <oej@edvina.net>

	* doc/README.cdr: - Adding reference to README.tds - Reformatting
	  table

2006-01-09 22:48 +0000 [r7917]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: re-initialize _all_ sequence numbers when
	  transfer completes

2006-01-09 22:07 +0000 [r7915]  Russell Bryant <russell@digium.com>

	* file.c: add missing unlock (issue #6112)

2006-01-09 20:08 +0000 [r7904-7908]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* pbx/pbx_spool.c: Bug 6157 - Memory leak

	* doc/README.variables: Update variable documentation to match the
	  code

2006-01-09 18:11 +0000 [r7898-7900]  Kevin P. Fleming <kpfleming@digium.com>

	* asterisk.c: commit user/group-related changes from trunk

	* db.c: backport fix from revision 7856 of trunk

	* apps/app_voicemail.c: fix breakage introduced in revision 7871

2006-01-09 05:11 +0000 [r7870-7871]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: fix seg fault when using greek syntax in
	  VoicemMailMain (issue #6142)

	* manager.c: backport fix for unnecessary unlock (issue #6171)

2006-01-07 07:27 +0000 [r7848]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* pbx/pbx_spool.c: Bug 6156 - catch all threading errors, not just
	  simple failure

2006-01-06 00:34 +0000 [r7831]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* pbx/pbx_config.c: Dumb error messages - "Context 'context'
	  already included in 'in' context"

2006-01-06 00:21 +0000 [r7829]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_agent.c: update agent persistence when an agent
	  gets logged off by autologoff

2006-01-05 23:53 +0000 [r7827]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk/strings.h: Bug 6076 - Fix documentation of
	  ast_trim_blank return value

2006-01-05 23:49 +0000 [r7825]  Kevin P. Fleming <kpfleming@digium.com>

	* channel.c: eliminate rounding errors that caused call time limits
	  to be inaccurate (issue #5913) round 'time left' reported during
	  call limit warnings up to sound more accurate

2006-01-05 23:07 +0000 [r7823]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_features.c: Bug 6081 - fix for memory leak, formatting
	  fixes

2006-01-05 20:52 +0000 [r7819]  Kevin P. Fleming <kpfleming@digium.com>

	* formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that
	  variable is initialized

2006-01-05 09:13 +0000 [r7812]  Olle Johansson <oej@edvina.net>

	* res/res_features.c: Fix copyright of changed file

2006-01-05 00:58 +0000 [r7799-7809]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_agent.c: send device state updates for auto-logoff
	  of agents as well

	* formats/format_pcm.c, formats/format_pcm_alaw.c: doh... fseek()
	  has no useful return value

	* formats/format_pcm.c, formats/format_pcm_alaw.c: use proper
	  fwrite() parameters and return value

	* formats/format_pcm.c, formats/format_pcm_alaw.c: return properly
	  after extending file

	* formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that
	  ulaw/alaw sound files are filled with silence when extended (not
	  zeroes)

	* channel.c: make monitoring more tolerant of peers that deliver
	  frames in bursts

2006-01-04 21:46 +0000 [r7792-7795]  Olle Johansson <oej@edvina.net>

	* res/res_features.c: Issue #5980: Removing extra CR+LF in manager
	  events - needs port to trunk

	* channels/chan_sip.c: Fixing typo in XML for video updates.

2006-01-04 07:06 +0000 [r7773]  Russell Bryant <russell@digium.com>

	* funcs/func_moh.c: use a more correct way of determining the size
	  of the destination buffer

2006-01-04 05:27 +0000 [r7771]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_privacy.c: Fix the 'if' clause to be true under the
	  right conditions. Bug #6126

2006-01-03 20:22 +0000 [r7746]  Kevin P. Fleming <kpfleming@digium.com>

	* ast_expr.y (removed): remove unused 'old' expression parser

2006-01-03 18:15 +0000 [r7743]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_stack.c: Bug 6121 - typo in application description

2006-01-03 17:24 +0000 [r7736-7740]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/chanspy.h, apps/app_chanspy.c,
	  apps/app_mixmonitor.c, channel.c: revert incorrect fix for bug
	  #6048 from revision 7709 put in correct (simpler) fix add doxygen
	  docs for channel spy 'state' values

	* channels/chan_sip.c: backport rport scanning fix from trunk (bug
	  #6071)

	* ast_expr2f.c, ast_expr2.fl: don't leak memory for (most)
	  expression evaluations

2006-01-02 07:31 +0000 [r7709]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_mixmonitor.c: Bug 6084 - MixMonitor after a 'cli stop
	  monitor' deadlocks

2006-01-02 02:04 +0000 [r7706]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Fix compiler warnings.

2005-12-30 14:54 +0000 [r7677]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channel.c: Bug 6091 - Fix race condition around uniqueid

2005-12-28 17:35 +0000 [r7663-7665]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: fix memory leak in build_rpid (issue #6070)

	* apps/app_chanspy.c: backport fix for permissions of created
	  recordings (issue #6067)

2005-12-27 00:07 +0000 [r7641]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: backport fix to ensure that DSP is never
	  enabled on pseudo channels

2005-12-26 20:32 +0000 [r7637]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* cdr/cdr_tds.c: Remove copy of code in libc, preferring code in
	  utils.c (public domain code)

2005-12-26 18:19 +0000 [r7634]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c, channels/chan_agent.c, apps/app_sms.c,
	  asterisk.c, config.c, pbx/pbx_dundi.c, apps/app_externalivr.c,
	  apps/app_queue.c, channels/chan_iax2.c, cli.c,
	  apps/app_chanspy.c, res/res_monitor.c: cast time_t to an int in
	  printf/scanf (issue #5635)

2005-12-23 06:38 +0000 [r7608]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_hasnewvoicemail.c: Bug 6051 - VMCOUNT should work as
	  documented and count all, not quit after finding 1

2005-12-23 03:01 +0000 [r7606]  Kevin P. Fleming <kpfleming@digium.com>

	* asterisk.c: add license reference to copyright notice displayed
	  when CLI session begins add 'show warranty' and 'show license'
	  CLI commands (still need a complete list of non-GPL components
	  included in Asterisk)

2005-12-23 00:00 +0000 [r7605]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_waitforsilence.c: Another app documentation tweak.

2005-12-22 22:04 +0000 [r7601]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 6050 SQL requires the use of single
	  ticks to delimit values, not quotes

2005-12-22 20:36 +0000 [r7595-7599]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: revert changes to
	  videosupport to allow per-peer setting, since it isn't quite
	  complete and there is not an obvious fix at this point

	* channels/chan_sip.c: remove stray unlock (issue #5955)

2005-12-21 22:23 +0000 [r7586]  Josh Roberson <josh@asteriasgi.com>

	* channels/chan_sip.c: Actually put in the per-peer settings for
	  sip video, as they didn't make it in at astricon somehow, and
	  I've been too busy up until now to redo it.

2005-12-21 20:01 +0000 [r7582]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_alsa.c: Allow a chan_alsa that failed to open sound
	  devices to be unloaded.

2005-12-21 19:53 +0000 [r7580]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_agent.c: Bug #6040 - Documentation correction

2005-12-21 19:23 +0000 [r7577]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* pbx/pbx_ael.c: Bug 5777 - Remove parentheses on Goto in AEL, so
	  that it parses correctly

2005-12-20 20:21 +0000 [r7550-7557]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: check array bounds when parsing arguments to AGI
	  (issue #5868)

	* channels/chan_iax2.c: backport fix for reloading peer context
	  (issue #6007)

	* apps/app_directed_pickup.c: backport fix for segfault on directed
	  pickup when no CDR is available (issue #5998)

2005-12-20 12:58 +0000 [r7546]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_meetme.c: backport fix for larger-than-20ms-frames from
	  trunk (bug #5697)

2005-12-19 23:47 +0000 [r7529]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: I messed up and accidently committed this to
	  the trunk first ... - add note on required values of sip_methods
	  struct - remove duplicate function prototype - remove duplicate
	  ast_mutex_lock (issue #6025)

2005-12-19 19:06 +0000 [r7521-7523]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* file.c: Bug 5988 - record append option not working

	* cdr.c: Bug 6026 - segfault for the sequence NoCDR(),
	  SetAMAFlags()

2005-12-17 18:55 +0000 [r7517-7519]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* doc/README.ael: Document that curley braces must be on the same
	  line as the keyword.

	* apps/app_chanspy.c: Bug 6009 - off by one error

2005-12-17 03:59 +0000 [r7510-7515]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: Max-Forwards headers must only be present on
	  requests, not responses

	* channels/chan_sip.c: forcibly expire previous subscriptions from
	  a peer when they resubscribe (keeps them from building up and
	  waiting for expiration, and stops us sending unwanted NOTIFY
	  messages to devices)

	* build_tools/make_svn_branch_name: fix some buglet when building
	  team branch version strings

2005-12-17 01:02 +0000 [r7508]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk/linkedlists.h: We want to check the previous
	  value, not the current value (which was just changed).

2005-12-16 00:49 +0000 [r7497]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_cut.c: First field is truncated

2005-12-15 10:52 +0000 [r7490]  Christian Richter <christian.richter@beronet.com>

	* doc/README.misdn, channels/misdn/mISDNuser.patch (added),
	  channels/misdn/isdn_lib_intern.h, channels/misdn/mISDN.patch
	  (added), channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/Makefile, channels/misdn/chan_misdn_config.h,
	  channels/misdn/ie.c, channels/chan_misdn_config.c,
	  channels/misdn/isdn_msg_parser.c, channels/Makefile,
	  channels/misdn/isdn_lib.c: * Added mISDN/mISDNuser Echo cancel
	  Patch * Fixed Makefiles so that chan_misdn can be compiled again
	  * added some hints, that mISDN cannot be compiled against gcc-4,
	  SMP, Spinlock Debug * fixed some Minor issues in chan_misdn,
	  regarding Type Of Number and Presentation

2005-12-15 02:51 +0000 [r7482]  BJ Weschke <bweschke@btwtech.com>

	* channel.c: Bug #6003 - Don't free the channel structure until
	  after having sent the manager event.

2005-12-13 18:54 +0000 [r7435-7470]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/README.variables: clarify substring documentation

	* utils.c: correct broken math in tvfix() for timestamp values over
	  one million

	* apps/app_dial.c: restore ability of caller to hangup calls that
	  are still ringing (issue #5839)

	* channels/chan_sip.c, pbx.c: ensure that hangups while incoming
	  calls are in early state are handled properly (issue #5919)

	* channels/chan_agent.c: only report AGENT_IDLE for callback mode
	  agents when they are actually idle (issue #5902)

	* app.c: use the stream's current point when pausing/unpausing,
	  instead of elapsed time (which doesn't work when the stream has
	  been skipped forward or backward) (issue #5897)

	* apps/app_externalivr.c: set all the child file descriptors to
	  non-blocking so that we don't hang if the child fails to send a
	  newline-terminated command or error message

2005-12-12 17:19 +0000 [r7433]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk/linkedlists.h: Typo

2005-12-11 06:08 +0000 [r7430]  Russell Bryant <russell@digium.com>

	* utils/astman.c: silence a couple of compiler warnings about
	  pointer signedness

2005-12-11 01:26 +0000 [r7427-7429]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* include/asterisk/linkedlists.h: Bug 5965 - major bug in
	  AST_LIST_REMOVE

	* apps/app_voicemail.c: Bug 5967

2005-12-10 18:10 +0000 [r7425]  BJ Weschke <bweschke@btwtech.com>

	* channels/chan_zap.c: Bug #5877 Make sure the digit string from
	  E&M wink DNIS collection is properly null terminated as it grows.

2005-12-08 23:45 +0000 [r7404-7406]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 5960

	* configs/res_odbc.conf.sample: Documenting two keywords that were
	  previously missing

2005-12-08 01:05 +0000 [r7382-7386]  Kevin P. Fleming <kpfleming@digium.com>

	* pbx.c: initialize the buffer before using it...

	* pbx.c: ensure that hints are allowed to use global variable
	  references

2005-12-06 Kevin P. Fleming <kpfleming@digium.com>

	* Asterisk 1.2.1 Released

2005-12-05 06:47 +0000 [r7335-7340]  Russell Bryant <russell@digium.com>

	* Makefile: remove ASTERISKVERSIONNUM from the version string given
	  to doxygen

	* apps/app_queue.c: don't delete dynamic queue members when
	  reloading the static members from a realtime database (issue
	  #5922)

	* channels/chan_sip.c: fix the order of arguments to an error
	  message (issue #5927)

2005-12-04 18:03 +0000 [r7329]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/make_svn_branch_name: use a more efficient way to get
	  the revision number, that will also report if the working copy
	  contains uncommitted modifications

2005-12-03 19:55 +0000 [r7310]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 5925: check for "Unknown", as that's
	  what app_voicemail puts into the field for Unknown callerid Also,
	  remove useless res checks (initialized to 0; never set)

2005-12-03 01:24 +0000 [r7299]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Documenting the default registerattempts
	  setting as 0, continue hammering the server for ever and ever ;-)

2005-12-02 21:12 +0000 [r7285]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.mandrake.asterisk,
	  contrib/init.d/rc.redhat.asterisk,
	  contrib/init.d/rc.gentoo.asterisk,
	  contrib/init.d/rc.mandrake.zaptel,
	  contrib/init.d/rc.slackware.asterisk: Turn on executable bits for
	  startup scripts, and fix bash var interpolation for Mandrake

2005-12-02 00:52 +0000 [r7275]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Bug #5907. Improve SIP INFO DTMF debugging
	  output. (1.2 & Trunk)

2005-12-02 00:51 +0000 [r7266-7274]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_page.c, pbx.c: inherit channel variables into channels
	  created by Page() application (issue #5888)

	* apps/app_voicemail.c, configs/voicemail.conf.sample, UPGRADE.txt:
	  allow previous context-searching behavior to be used if desired
	  (issue #5899)

	* apps/app_voicemail.c: properly handle password changes when
	  mailbox is last line of config file and not followed by a newline
	  (issue #5870) reformat password changing code to conform to
	  coding guidelines (issue #5870)

	* channels/chan_agent.c: protect agent_bridgedchannel() from
	  segfaulting when there is no bridged channel (issue #5879)

	* channels/chan_local.c: allow variables to exist on both 'halves'
	  of the Local channel (issue #5810)

	* apps/app_festival.c: don't block waiting for the Festival server
	  forever when it goes away (issue #5882)

	* channel.c: ensure channel's scheduling context is freed (issue
	  #5788)

	* Makefile, patches (removed): Makefile 'update' target now
	  supports updating from Subversion repositories (issue #5875)
	  remove support for 'patches' subdirectory, it's no longer useful

2005-12-01 23:18 +0000 [r7261-7265]  Olle Johansson <oej@edvina.net>

	* doc/README.misdn: Changing bug report address to the Asterisk
	  issue tracker

	* doc/README.jitterbuffer, doc/README.realtime: Removing references
	  to 1.1dev, replacing with 1.2, in documentation files.

	* doc/README.misdn: Fixing some spelling errors, as well as
	  changing "cvs" to "subversion" in misdn documentation.

2005-12-01 19:25 +0000 [r7257]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/make_svn_branch_name: ensure that 'svn info' output
	  is in the expected language for the script to parse (issue #5880)

2005-12-01 02:33 +0000 [r7228-7251]  Russell Bryant <russell@digium.com>

	* apps/app_externalivr.c: use ast_app_separate_args to split
	  arguments (issue #5686)

	* apps/app_queue.c: fix queue weight feature - compare member
	  interfaces instead of pointers to the members, since each queue
	  has its own list of members. (issue #5863)

	* build_tools/make_svn_branch_name: use '=' instead of '==' for
	  string comparisons. /bin/bash is ok with this, but /bin/sh is
	  not. (issue #5885)

	* redhat/asterisk (removed), Makefile: remove outdated redhat init
	  script and provide the updated one in 'make rpm' (issue #5786)

	* contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.redhat.asterisk: Comment out LD_ASSUME_KERNEL
	  by default. Print error messages if the asterisk executable or
	  the asterisk configuration directory are not found. (issue #5785,
	  #5708)

	* apps/app_dial.c: fix DIALEDTIME when call has not been answered
	  (issue #5862)

	* rtp.c: do not allow an rtp message with zero type (issue #5749)

	* pbx.c: fix hint case sensitivity (issue #5856)

	* configs/sip.conf.sample: add description of the "fromdomain"
	  option (issue #5874)

2005-11-30 03:52 +0000 [r7227]  Josh Roberson <josh@asteriasgi.com>

	* apps/app_voicemail.c, UPGRADE.txt, ChangeLog: backport fix from
	  trunk

2005-11-30 03:37 +0000 [r7219-7226]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/cdr.txt, doc/CODING-GUIDELINES, include/asterisk.h,
	  doc/README.mp3: remove remaining CVS references

	* channel.c: port memory leak fix from rev 7223 in trunk

	* include/asterisk/lock.h: do the multiple-lock check for cond_wait
	  properly...

2005-11-29 06:12 +0000 [r7216-7218]  Russell Bryant <russell@digium.com>

	* apps/app_cut.c: print an error message if invalid arguments are
	  specified

	* apps/app_skel.c: fix a couple of typos and a buglet

2005-11-29 01:25 +0000 [r7199-7213]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/lock.h: if the lock protected a pthread_cond is
	  held recursively, warn before waiting onthe condition

	* Makefile, build_tools/make_svn_branch_name (added): port version
	  string computation from trunk

	* / (added): branch renames remove unneeded branches

2005-11-29  Josh Roberson <josh@asteriasgi.com>
	
	* apps/app_voicemail.c: Only look in 'default' context when no context defined to VoiceMailMain(). (issue #5887)

2005-11-25  Russell Bryant  <russell@digium.com>

	* apps/app_dial.c: Properly duplicate the string for ANI (issue #5850)

2005-11-23  Russell Bryant  <russell@digium.com>

	* configs/voicemail.conf.sample: Add note to indicate that #include should not be used for this file. (issue #5828)

	* indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
	* configs/indications.conf.sample: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
	* include/asterisk/indications.h: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
	* res/res_indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)

	* apps/app_voicemail.c: Remove left over "yay!" debugging message. (issue #5829)

2005-11-21  Kevin P. Fleming  <kpfleming@digium.com>

	* apps/app_cut.c: remove unnecessary include that causes spurious rebuilding

	* channels/chan_sip.c (build_peer): ensure that case changes made to peer names are not ignored during reload operations
 	(build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified

	* channels/chan_iax2.c (build_peer and build_user): ensure that case changes made to peer/user names are not ignored during reload operations
 	(build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified

2005-11-21  Russell Bryant  <russell@digium.com>

	* Makefile: Revert previous change for Darwin.

	* apps/app_osplookup.c: Properly populate the number of results. (issue #5789)

	* Makefile: Don't hard-code that poll functionality needs to be provided on Darwin.
	* apps/Makefile: Fix incorrect portion of the patch to fix 'make install' on Solaris.

	* channels/chan_iax2.c (iax2_getpeername): Return non-zero to indicate that a peer was found when using realtime (issue #5815)

2005-11-20  Russell Bryant  <russell@digium.com>

	* Makefile apps/Makefile: Fix 'make install' for Solaris. (issue #5775)

	* apps/app_record.c: Don't leak a frame if writing it to the file fails. (issue #5787)

	* Makefile: Create the monitor spool directory when the other spool directories are created.

	* channels/chan_sip.c channels/chan_iax2.c: Change warning messages about the number of scheduled events happening all at once to debug messages. (issue #5794)

	* pbx/pbx_spool.c: Fix crash when a value is not specified with a variable on a Set: line in a call file. (issue #5806)

	* apps/app_meetme.c: Fix the 'X' option to the MeetMe application. (issue #5773)

	* apps/app_voicemail.c: Correct the use of a mailbox entered by the calling party instead of indicated as an argument to the Voicemail application. (issue #5774)

	* apps/app_controlplayback.c: Fix logic in checking for success when jumping to priority n+101.
	* apps/app_md5.c: Fix logic in checking for success when jumping to priority n+101.

	* apps/app_hasnewvoicemail.c: Fix a typo in the application description.  Also, fix the logic in checking for success when jumping to priority n+101. (issue #5795) 

	* UPGRADE.txt: Add a note on a second way that the IAX2 channel naming convention has changed. (issue #5792)
	* channels/chan_iax2.c: Fix alignment of the output for the "iax2 show peer <peer>" CLI command (issue #5792)

	* channels/Makefile: Re-add chan_oss to the default build. (issue #5799)

	* res/res_musiconhold.c: Fix incorrect argument for the buffer size to an ast_copy_string call (issue #5803)

	* funcs/func_enum.c: Shorten the module description (issue #5791)

2005-11-17  Russell Bryant  <russell@digium.com>

	* Makefile: Fix the output of Makefile generated variables to doxygen

	* channels/chan_sip.c: Add missing carriage return and line feed to the SDP line indicating that we don't support VAD (issue #5780)

2005-11-16  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.2.0 released.
	
2005-11-16  Jeremy McNamara <jj@nufone.net>

	* apps/app_voicemail.c (load_config): do not terminate asterisk if no voicemail config file
	* channels/chan_skinny:  Don't register channel type until ready, code formatting updates

2005-11-16  Josh Roberson <josh@asteriasgi.com>

	* Makefile:  Update to fix non-responsive remote console on Darwin (OSX)(issue #5757)

2005-11-16  Kevin P. Fleming  <kpfleming@digium.com>

	* channels/Makefile: don't build chan_modem and sub-modules by default
	* configs/modules.conf.sample: explicitly 'noload' chan_modem.so and submodules, in case old versions exist
	
	* res/Makefile: issue mpg123 not-installed warning at 'make install' time, not 'make'

	* apps/app_forkcdr.c (forkcdr_exec): issue warning (and don't segfault) if ForkCDR is called on a channel that doesn't have a CDR (issue #5763)

	* channel.c (ast_queue_hangup): ensure that the channel lock is held before changing its fields... (issue #5770)

	* res/res_musiconhold.c: don't spit out incorrect log messages (and leak memory) during reload (issue #5766)

	* channels/chan_sip.c (process_sdp): don't pass video codec number into ast_getformatname(), it is not valid input for that function (issue #5764)

	* pbx/pbx_ael.c (match_assignment): properly parse equal signs surrounded by whitespace (issue #5761)

	* doc/README.realtime: document the limitations of using FreeTDS with Realtime (issue #5767)

2005-11-15  Kevin P. Fleming  <kpfleming@digium.com>

	* Makefile: use -g3 for compiler to include macro information for debugger

	* astmm.c (__ast_vasprintf): don't re-use the ap list without copying it; that's not safe on some platforms (issue #5035)

	* doc/README.backtrace: add note about properly building Asterisk to be able to produce backtraces; wrap text and remove DOS line endings

	* channels/chan_sip.c (add_codec_to_sdp): add 'annexb=no' to G.729A SDP (issue #5539)

	* channels/chan_alsa.c (alsa_hangup): handle autohangup properly (issue #5672)

	* channels/chan_misdn.c (and other files): various fixes (issue #5739)

	* channels/chan_sip.c (handle_request_info): properly forward 'flash' events received via SIP INFO (issue #5751, different patch)

	* apps/app_disa.c (disa_exec): don't duplicate constant strings when not needed

	* apps/app_playback.c (playback_exec): use correct logic tests for options (issue #5752)

	* apps/app_disa.c (disa_exec): use standard arg parsing routines (issue #5736)

2005-11-15  Russell Bryant <russell@digium.com>

	* manager.c: Don't crash on a SetVar action if the channel name is not set, or variable's value is not set (issue #5760)

	* doc/README.variables: Add application exit status variables

2005-11-14  Josh Roberson <josh@asteriasgi.com>
	
	* manager.c: Fix crash on variable passing from AMI originate (issue #5737)

2005-11-14  Russell Bryant  <russell@digium.com>

	* many files: Merge doxygen documentation updates. (issue #5605)

	* apps/app_dial.c: Fix typo in RetryDial description.

2005-11-12  Russell Bryant  <russell@digium.com>

	* channels/chan_oss.c: Fix a typo in an error message.

2005-11-11  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.2.0-rc2 released.
	
2005-11-11  Kevin P. Fleming  <kpfleming@digium.com>

	* channels/chan_sip.c (thread_safe_rand): ensure that threads don't get the same random number (issue #5712)

	* apps/app_voicemail.c (forward_message): correct bugs in message forwarding (issue #5718)
	(copy_message): use correct path for locking (issue #5704)

	* apps/app_dial.c (wait_for_answer): correct flag copying for automon feature (issue #5720)

	* channels/chan_iax2.c: correct comment

	* apps/app_voicemail.c (close_mailbox): correct previous commit (issue #5663)
	(vm_change_password): fix password change writing (issue #5721)

	* channels/chan_sip.c (transmit_invite): remove useless debug message; don't try to add OSP tokens to OPTIONS pings

	* apps/app_voicemail.c (close_mailbox): properly remove deleted messages at mailbox close time (issue #5663)

2005-11-11  Mark Spencer <markster@digium.com>

	* channels/chan_zap.c (zt_bridge): only enable/disable DTMF detection on SUB_REAL channels

2005-11-10  Kevin P. Fleming  <kpfleming@digium.com>

	* channels/chan_iax2.c: ensure that system headers that provide basic types are included first (issue #5713)

2005-11-11  Russell Bryant  <russell@digium.com>

	* many files in apps/: Clean up application descriptions. Clarify some wording and make sure they wrap at 80 characters.

2005-11-10  Mark Spencer <markster@digium.com>

	* rtp.c (ast_rtp_raw_write): use unsigned int for return value from calc_txstamp() (issue #5595)
	(calc_txstamp): never return a value that was less than zero before being turned into 'unsigned int' (issue #5595)

2005-11-10  Kevin P. Fleming  <kpfleming@digium.com>

	* include/asterisk/chanspy.h: move spy-related stuff into separate header so chan_h323 can build (issue #5590)

	* include/asterisk/linkedlists.h (AST_LIST_HEAD_SET_NOLOCK): properly initialize tail pointer when list head is directly set (issue #5669)

	* app.c (ast_app_parse_options): ok, so we aren't all perfect... let's make the while loop actually work properly here (issue #5684)

	* apps/app_disa.c (disa_exec): correct password file parsing (issue #5676)

	* apps/app_meetme.c (conf_run): don't restrict admin users from joining a locked conference (issue #5680)

	* channels/chan_misdn.c: include stdio.h (issue #5671)
	* channels/chan_misdn_config.c: fix prototype warning (issue #5671)

	* pbx.c: remove apps that were deprecated before 1.0 was released (issue #5673)

	* apps/app_striplsd.c, apps/app_substring.c: remove apps that were deprecated before 1.0 was released (issue #5673)

	* include/asterisk/lock.h (PTHREAD_MUTEX_RECURSIVE_NP): work around header problems on Cygwin (issue #5668)

	* pbx/pbx_ael.c: handle switch default cases inside macros properly (issue #5354)

	* configs/voicemail.conf.sample (format): add strong warning about changing format list when mailboxes contain messages (issue #5689)

	* many files: ensure that system headers are included before Asterisk headers (issue #5693)

	* channels/chan_iax2.c (complete_iax2_show_peer): don't return from function without releasing lock (issue #5685)

	* channels/iax2-provision.c (iax_provision_reload): don't leak memory (issue #5700)

	* pbx/pbx_ael.c (handle_macro): don't leak memory (issue #5701)
	(handle_context): ditto

	* res/res_features.c (load_config): properly initialize referenced variable (issue #5703)

	* apps/app_queue.c (rqm_exec): correct segfault problem (issue #5705)
	(aqm_exec): ditto

	* app.c (ast_app_parse_options): don't increment 's' until after checking for NULL (related to issue #5630)

	* apps/app_rpt.c: solve a memory leak (config structure was not freed) (issue #5706)

2005-11-10  Russell Bryant <russell@digium.com>

	* app.c (ast_app_separate_args): Don't consider the open parenthesis as part of the arguments to an option. (issue #5630)

	* many files: Change all references to ast_separate_app_args to ast_app_separate_args

	* many files in apps/: Clean up some application descriptions. Make sure all descriptions in changed files are wrapped at 80 characters.

2005-11-09  Russell Bryant <russell@digium.com>

	* pbx.c: Clean up descriptions of built-in dialplan applications. Changes include clearer wording and not referring to return values.

2005-11-09  Kevin P. Fleming  <kpfleming@digium.com>

	* channels/chan_iax2.c (update_registry): don't complain about unspecifed registration expiration intervals, just use the minimum

2005-11-08  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.2.0-rc1 released.

	* include/asterisk/file.h: add test to ensure that stdio.h is included before this file (issue #5658)

	* res/res_odbc.c (odbc_prepare_and_execute): add new API call for use to properly handle prepared statements across server disconnects (issue #5563)

	* pbx.c (pbx_substitute_variables_helper_full): use already-substituted buffer for parsing variable name (issue #5664)

	* channels/chan_zap.c (zt_request): return AST_CAUSE_CONGESTION when a group-channel is requested and the group exists but all channels are busy (issue #3360, related fix)
	* channels/chan_iax2.c (create_addr): treat UNREACHABLE as AST_CAUSE_UNREGISTERED so that it will generate CHANUNAVAIL from app_dial (issue #3360)

	* res/res_features.c (ast_bridge_call_thread_launch): set SCHED_RR separately from thread creation, so it won't fail when running as non-root (issue #5601, different fix)

	* pbx.c (pbx_builtin_pushvar_helper): add new API function for setting variables that can exist multiple times (issue #2720)
	* apps/Makefile (APPS): add app_stack (issue #2720)
	* apps/app_stack.c: new applications (issue #2720)

	* apps/app_meetme.c: fix two audio delay problems related to using non-Zap channels in conferences (issues #3599 and #4252)
	* configs/meetme.conf.sample: add documentation of new 'audiobuffers' setting to control buffering on incoming audio from non-Zap channels

	* channels/chan_local.c (local_call): move channel variables from incoming to outgoing instead of inheriting them (issue #5604)

	* many files: add explicit include of stdio.h (issue #5650)

2005-11-07  Kevin P. Fleming  <kpfleming@digium.com>

	* UPGRADE.txt (Parking): add note about new parking behavior (issue #5532)

	* many files: more Cygwin compatibility, and proper getloadavg() prototype/macro (issue #5569)

	* include/asterisk/lock.h (__ast_pthread_mutex_lock): correct build with DETECT_DEADLOCKS defined (issue #5570)

2005-11-07  Russell Bryant  <russell@digium.com>

	* apps/app_queue.c: upgrade to new arg/option API and implement priority jumping control (issue #5580)
	* many files: Add missing include of stdio.h, and remove some duplicate and unused header includes

	* include/asterisk/app.h: Increment the arg_index in the options structure to fix applicaiton options that have arguments to them

2005-11-07  Kevin P. Fleming  <kpfleming@digium.com>

	* cryptostub.c: include necessary headers
	* include/asterisk/crypto.h: don't include unnecessary headers

	* manager.c (action_setvar): add support for setting global variables (issue #5571)

	* Makefile: correct cross-compilation issue introduced in Cygwin patches (issue #5572)

	* apps/app_voicemail.c: upgrade to new arg/option API and implement priority jumping control (issue #5649)

	* asterisk.c (main): setpriority() failure is not a reason to stop the process (issue #5581)

	* say.c (ast_say_date_with_format_da): say hours properly (issue #5576)

	* manager.c (astman_get_variables): restore old multiple-variable behavior for "Variable" header (issue #5585)

	* many files: don't check for NULL before calling ast_strlen_zero, it can do it itself (issue #5648)

	* pbx.c (handle_show_hints): use proper state-to-string function for hint state (issue #5583)

	* rtp.c: use unsigned format for debug packet output (issue #5595)

	* asterisk.c (main): force a dnsmgr background refresh after all other modules are initialized (issue #5599)
	* dnsmgr.c: add ability to start a background refresh on demand (issue #5599)

	* apps/app_dial.c (HANDLE_CAUSE): set CDR disposition to match cause code (issue #5602)

	* asterisk.c: support 'runuser' and 'rungroup' options in asterisk.conf (issue #5621)

	* res/Makefile, apps/Makefile, channels/Makefile, Makefile: support WITHOUT_ZAPTEL define to forcibly avoid building Zaptel support (issue #5634)

	* Makefile: various fixes (issue #5633)

	* apps/app_osplookup.c: upgrade to new arg/option API and implement priority jumping control

	* channels/chan_misdn.c: various fixes (issue #5639)
	* channels/misdn/isdn_lib.c: various fixes (issue #5639)

	* apps/app_playback.c: upgrade to new arg/option API and implement priority jumping control

	* apps/app_privacy.c: upgrade to new arg/option API and implement priority jumping control

	* apps/app_sendtext.c: upgrade to new arg/option API and implement priority jumping control

	* apps/app_transfer.c: upgrade to new arg/option API and implement priority jumping control

	* apps/app_txtcidname.c: upgrade to new arg/option API and implement priority jumping control

	* Makefile: restore function of 'dont-optimize'

	* config.c (config_text_file_load): don't generate log message when stat() fails

	* many files: clean up application documentation to not refer to return values, since they cannot be used in the dialplan (work done by Neil Lewis)

2005-11-06  Russell Bryant <russell@digium.com>

	* many files: alphabetize options in applicaiton descriptions

	* channels/chan_iax2.c: Use an enum to define iax peer/user flags as well as the pvt structure state.  Use the ast_flags macros for checking or setting the state.

	* sounds.txt: Add missing words from the description of the vm-opts prompt

	* apps/app_externalivr.c: Add a space that fixes building on older versions of gcc

	* many files: Add doxygen updates to categorize modules into groups. Convert a lot of comments over to doxygen style. Add some text giving a basic overview of channels.

	* many files: Update applications to add an exit status variable, make priority jumping optional, and use new args parsing macros

	* pbx.c cdr.c res/res_features.c apps/app_dial.c include/asterisk/cdr.h: Convert some built-in applications to use new args parsing macros.  Change ast_cdr_reset to take a pointer to an ast_flags structure instead of an integer for flags.

	* channels/chan_agent.c: Don't loop forever on an invalid options string

	* apps/app_disa.c apps/app_forkcdr.c: Fix to use correct arguments to ast_cdr_reset

2005-11-05  Kevin P. Fleming  <kpfleming@digium.com>

	* Makefile: don't rebuild asterisk/build.h unless the asterisk binary is going to be relinked for some other reason (stops spurious recompile/link every time 'make' is issued); clean up variable substitutions to use consistent syntax
	* asterisk.c: don't include asterisk/build.h (it's unnecessary)
	* cli.c: don't include asterisk/build.h, use extern references to buildinfo.c
	* buildinfo.c: new file to hold version info strings

2005-11-04  Kevin P. Fleming  <kpfleming@digium.com>

	* apps/app_mixmonitor.c (mixmonitor_exec): correct app name in an error message

2005-11-04  Russell Bryant  <russell@digium.com>

	* channels/chan_iax2.c: Create a function that stores a peer's status in a given buffer.  Use this function in "iax2 show peers" and "iax2 show peer <peername>".  Also, add the peer's status as an option to the IAXPEER dialplan function.

2005-11-04  Kevin P. Fleming  <kpfleming@digium.com>

	* include/asterisk/compiler.h: don't try to use always_inline on old compilers

2005-11-03  Russell Bryant  <russell@digium.com>

	* res/res_agi.c: initialize buffer for result so that the contents are always valid in the response to GET FULL VARIABLE

2005-11-03  Kevin P. Fleming  <kpfleming@digium.com>

	* doc/README.variables: document DYNAMIC_FEATURES

	* res/res_features.c (ast_bridge_call): remove unused variables

	* apps/app_dial.c (dial_exec_full): simplify options and flag usage

	* include/asterisk/app.h: re-work application arg/option parsing APIs for consistent naming, add doxygen docs for option API
	* many files: update to new APIs

2005-11-02  Kevin P. Fleming  <kpfleming@digium.com>

	* apps/app_dial.c (dial_exec_full): convert to use API calls for argument/option parsing

	* include/asterisk/channel.h: add doxygen docs for silence generator APIs

	* channel.c (ast_channel_bridge): simplify native-bridge return logic, remove 'unsuccessful' message since it causes too many questions :-)

2005-11-01  Kevin P. Fleming  <kpfleming@digium.com>

	* stdtime/localtime.c: fix build failure on uClibc systems (issue #5558)
	* devicestate.c: same

	* many files: make chan_misdn actually build (issue #5566)

	* many files: more Cygwin build system support (issue #4678)

	* apps/app_parkandannounce.c (parkandannounce_exec): supply parent channel to ast_request_and_dial so channel variables can be inherited (issue #5564)
	* include/asterisk/channel.h: add parent_channel field
	* channel.c (__ast_request_and_dial): use parent_channel field to inherit variables into new channel

	* apps/app_cut.c (cut_internal): use ast_app_separate_args() instead of open code (issue #5560)

	* apps/app_mixmonitor.c (launch_monitor_thread): ast_strlen_zero can handle NULL input (issue #5561)
	(mixmonitor_exec): same

	* res/res_features.c (ast_feature_request_and_dial): ensure that channel variables are inherited from the channel placing the call (issue #5499)

	* utils.c (getloadavg): change to using _BSD_SOURCE as the indicator for whether this function is present or not (issue #5549)
	
	* include/asterisk/utils.h (ast_slinear_saturated_add): force to be inlined whenever possible
	(ast_slinear_saturated_multiply): same
	(ast_slinear_saturated_divide): same
	(inaddrcmp): same
	* include/asterisk/strings.h (ast_strlen_zero): force to be inlined whenever possible
	* include/asterisk/compiler.h (force_inline): add macro to force inlining of functions
	
	* app.c (ast_play_and_record): use ast_silence_generator during recording if requested
	* asterisk.c: add global option to enable silence-during-record (issue #5135)
	* channel.c (silence_generator_alloc): new
	(silence_generator_release): new
	(silence_generator_generate): new
	(ast_channel_start_silence_generator): new API call to start generating silence on a channel
	(ast_channel_stop_silence_generator): parallel call to stop silence generation
	* apps/app_record.c (record_exec): use ast_silence_generator during recording if requested

2005-11-01  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.2.0-beta2 released.