[asterisk-commits] russell: tag 1.2.26.1 r94259 - in /tags/1.2.26.1: .lastclean .version ChangeLog

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Dec 20 14:31:13 CST 2007


Author: russell
Date: Thu Dec 20 14:31:12 2007
New Revision: 94259

URL: http://svn.digium.com/view/asterisk?view=rev&rev=94259
Log:
Importing files for 1.2.26.1 release

Added:
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    tags/1.2.26.1/.version   (with props)
    tags/1.2.26.1/ChangeLog   (with props)

Added: tags/1.2.26.1/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.2.26.1/.lastclean?view=auto&rev=94259
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URL: http://svn.digium.com/view/asterisk/tags/1.2.26.1/ChangeLog?view=auto&rev=94259
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--- tags/1.2.26.1/ChangeLog (added)
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+2007-12-20  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.2.26.1 released
+
+2007-12-20 20:21 +0000 [r94214-94255]  Russell Bryant <russell at 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 at digium.com>
+
+	* Asterisk 1.2.26 released
+
+2007-12-18 18:44 +0000 [r93667-93675]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c, channels/chan_iax2.c: Fixing AST-2007-027
+	  (Closes issue #11119)
+
+2007-11-29  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.2.25 released
+
+2007-11-29 21:10 +0000 [r90170]  Tilghman Lesher <tlesher at 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 at digium.com>
+
+	* LICENSE: clarify the OpenSSL and OpenH323 license exceptions
+
+2007-08-07  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.2.24 released
+
+2007-08-07 17:44 +0000 [r78370]  Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+	* Asterisk 1.2.23 released
+
+2007-07-24 16:32 +0000 [r76802]  Jason Parker <jparker at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+	* apps/app_queue.c: removed the word 'pissed' from ast_log(...)
+	  function call for BE-90
+
+2007-07-17  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.2.22 released
+
+2007-07-17 20:57 +0000 [r75440-75449]  Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+	* Asterisk 1.2.21.1 released
+
+2007-07-10 15:37 +0000 [r74316]  Jason Parker <jparker at 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 at 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 at 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 at digium.com>
+
+	* Asterisk 1.2.21 released
+
+2007-07-09 21:00 +0000 [r74165]  Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+	* Asterisk 1.2.20 released
+
+2007-06-29 16:30 +0000 [r72629]  Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+	* asterisk.c: Fixes 100% load when controlling terminal disappears.
+	  Issue #9654, #10010
+
+2007-06-27 20:23 +0000 [r72256]  Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+	* Asterisk 1.2.19 released
+
+2007-06-14 23:21 +0000 [r69469]  Jason Parker <jparker at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+	* channels/chan_sip.c: Issue #9802 - Change inuse counter on CANCEL
+
+2007-05-25 13:46 +0000 [r66127]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* say.c: Issue 9791 - Fix pronunciation of seconds in Dutch
+
+2007-05-24 14:40 +0000 [r65837]  Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at beronet.com>
+

[... 5634 lines stripped ...]



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