[asterisk-commits] lmadsen: tag 1.2.38 r245678 - in /tags/1.2.38: .lastclean .version ChangeLog

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Feb 9 09:58:13 CST 2010


Author: lmadsen
Date: Tue Feb  9 09:58:09 2010
New Revision: 245678

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=245678
Log:
Importing files for 1.2.38 release.

Added:
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    tags/1.2.38/.version   (with props)
    tags/1.2.38/ChangeLog   (with props)

Added: tags/1.2.38/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.2.38/.lastclean?view=auto&rev=245678
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--- tags/1.2.38/ChangeLog (added)
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+2010-02-09  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.2.38 released
+
+	* Previous regression commits were not properly rolled into
+	  releases. This release re-syncs the commits. 
+
+2009-11-30  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.2.37 released
+
+	* AST-2009-010
+
+2009-11-30 17:35 +0000 [r231518]  David Vossel <dvossel at digium.com>
+
+	* rtp.c: fixes crash caused by RTP comfort noise payload greater
+	  than 24 bytes AST-2009-010 (closes issue #16242) Reported by:
+	  amorsen Patches: issue16242.diff uploaded by oej (license 306)
+	  Tested by: amorsen, oej, dvossel
+
+2009-11-04  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.2.36 released
+
+	* AST-2009-008
+
+2009-09-03  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.2.35 released
+
+	* AST-2009-006
+
+2009-09-03 19:37 +0000 [r216005-216087]  Russell Bryant <russell at digium.com>
+
+	* UPGRADE.txt: Fix a typo.
+
+	* UPGRADE.txt: Add a note about IAX2 to UPGRADE.txt.
+
+	* doc/IAX2-security.pdf (added): Add IAX2 security document related
+	  to AST-2009-006.
+
+2009-09-03 16:57 +0000 [r215958]  David Vossel <dvossel at digium.com>
+
+	* Makefile, configs/iax.conf.sample, include/asterisk/acl.h, sha1.c
+	  (added), channels/iax2-parser.h, include/asterisk/utils.h, acl.c,
+	  utils.c, include/asterisk/astobj2.h, channels/iax2.h, astobj2.c,
+	  channels/chan_iax2.c, channels/iax2-parser.c,
+	  include/asterisk/sha1.h (added): Merge code associated with
+	  AST-2009-006 (closes issue #12912) Reported by: rathaus Tested
+	  by: tilghman, russell, dvossel, dbrooks
+
+2009-08-18 20:24 +0000 [r212903-212907]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* sounds/moh/fpm-calm-river.mp3 (removed),
+	  sounds/moh/macroform-cold_day.mp3 (added),
+	  sounds/moh/macroform-robot_dity.mp3 (added), CREDITS, README.fpm
+	  (removed), sounds/moh/fpm-world-mix.mp3 (removed),
+	  sounds/moh/manolo_camp-morning_coffee.mp3 (added),
+	  sounds/moh/LICENSE, README.opsound (added),
+	  sounds/moh/macroform-the_simplicity.mp3 (added),
+	  sounds/moh/reno_project-system.mp3 (added),
+	  sounds/moh/fpm-sunshine.mp3 (removed): Convert this branch to
+	  Opsound music-on-hold. For more details:
+	  http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
+
+	* /: remove extraneous property
+
+2009-06-05  Tilghman Lesher <tlesher at digium.com>
+
+	* Asterisk 1.2.34 released
+
+2009-08-10 19:13 +0000 [r211526-211527]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/misdn_config.c, frame.c, utils/frame.c,
+	  pbx/pbx_loopback.c, channels/chan_phone.c, apps/app_osplookup.c,
+	  pbx/pbx_spool.c, channels/chan_skinny.c, res/res_agi.c,
+	  indications.c, cli.c, channel.c, cdr.c, apps/app_groupcount.c,
+	  channels/chan_mgcp.c, manager.c, apps/app_adsiprog.c,
+	  apps/app_dial.c, channels/chan_zap.c, channels/chan_sip.c,
+	  apps/app_privacy.c, apps/app_waitforsilence.c,
+	  codecs/codec_speex.c, channels/chan_agent.c, funcs/func_math.c,
+	  apps/app_disa.c, channels/iax2-provision.c, pbx/dundi-parser.c,
+	  apps/app_talkdetect.c, apps/app_queue.c, pbx.c, dnsmgr.c,
+	  apps/app_math.c, Makefile, apps/app_waitforring.c,
+	  apps/app_zapbarge.c, apps/app_cut.c, channels/chan_misdn.c,
+	  acl.c, channels/chan_h323.c, res/res_osp.c, apps/app_macro.c,
+	  apps/app_sms.c, pbx/pbx_dundi.c, pbx/pbx_config.c,
+	  apps/app_verbose.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
+	  apps/app_voicemail.c, channels/chan_vpb.c, apps/app_readfile.c,
+	  muted.c, /, apps/app_meetme.c, res/res_features.c,
+	  apps/app_record.c, apps/app_sayunixtime.c, funcs/func_strings.c,
+	  apps/app_random.c, apps/app_alarmreceiver.c, asterisk.c,
+	  channels/chan_modem.c, channels/chan_iax2.c: AST-2009-005
+
+2009-07-14 14:45 +0000 [r206384]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Ensure apathetic replies are sent out on
+	  the proper socket. chan_iax2 supports multiple address bindings.
+	  The send_apathetic_reply() function did not attempt to send its
+	  response on the same socket that the incoming message came in on.
+
+2009-06-05  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.2.33 released
+
+2009-06-04 18:57 +0000 [r199137]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Additional updates to AST-2009-001
+
+2009-06-04  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c, apps/app_chanspy.c: Fixes REGAUTH loop
+	  related to AST-2009-001, also addresses a small compile time
+	  error in app_chanspy.c.
+
+2009-04-02  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.2.32 released
+
+2009-04-02 17:02 +0000 [r186056]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c, configs/sip.conf.sample: Fix for
+	  AST-2009-003
+
+2009-01-23  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.2.31.1 released
+
+2009-01-23 19:19 +0000 [r170580]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_iax2.c: Updates to AST-2009-001
+
+2009-01-15 01:15 +0000 [r168632]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_iax2.c: 1.2 regression on security fix AST-2009-001
+
+2009-01-09 22:10 +0000 [r168197]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* sounds/LICENSE (added): add license for Allison Smith prompts
+	  (AST-162)
+
+2009-01-06  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.2.31 released
+
+2009-01-06 20:44 +0000 [r167259]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_iax2.c: Security fix AST-2009-001.
+
+2008-12-10  Tilghman Lesher <tlesher at digium.com>
+
+	* Asterisk 1.2.30.4 released
+
+2008-12-10 21:06 +0000 [r162868]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_iax2.c: Fix for AST-2008-012
+
+2008-12-05 20:50 +0000 [r161421]  Sean Bright <sean.bright at 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 at digium.com>
+
+	* Asterisk 1.2.30.3 released
+
+2008-11-25 21:37 +0000 [r159245]  Tilghman Lesher <tlesher at 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 at digium.com>
+
+	* Asterisk 1.2.30.2 released
+
+2008-08-09 15:24 +0000 [r136945]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/compat.h, include/asterisk/astobj2.h: Regression
+	  fixes for Solaris
+
+2008-07-25 15:00 +0000 [r133577]  Russell Bryant <russell at digium.com>
+
+	* LICENSE: Fix the IAX2 URI for calling Digium
+
+2008-07-23  Tilghman Lesher <tlesher at digium.com>
+
+	* Asterisk 1.2.30.1 released
+
+2008-07-24 03:46 +0000 [r133360]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_iax2.c: This part was not correctly patched for
+	  AST-2008-010.
+
+2008-07-22  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.2.30 released
+
+2008-07-22 21:14 +0000 [r132711]  Tilghman Lesher <tlesher at 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 at digium.com>
+
+	* Asterisk 1.2.29 released
+
+2008-06-03 19:30 +0000 [r120109]  Joshua Colp <jcolp at 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 at 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 at digium.com>
+
+	* Asterisk 1.2.28.1 released
+
+2008-05-08 19:14 +0000 [r115564]  Russell Bryant <russell at 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 at 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 at 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 at 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 at digium.com>
+
+	* contrib/scripts/get_ilbc_source.sh: stop script from appending
+	  source code if run multiple times
+
+2008-04-22  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.2.28 released
+
+2008-04-22 22:20 +0000 [r114561]  Russell Bryant <russell at 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 at 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 at 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 at digium.com>
+
+	* Asterisk 1.2.27 released
+
+2008-03-18 16:27 +0000 [r109488]  Terry Wilson <twilson at digium.com>
+
+	* include/asterisk/astobj.h: Fix character string being treated as
+	  format string
+
+2008-03-18 15:08 +0000 [r109391]  Jason Parker <jparker at 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 at digium.com>
+
+	* Asterisk 1.2.26.2 released
+
+2008-01-07 20:46 +0000 [r96931]  Russell Bryant <russell at digium.com>
+
+	* configs/extensions.conf.sample: Change misery.digium.com to
+	  pbx.digium.com
+
+2007-12-23 01:30 +0000 [r94661]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c: Fix for fix for security fix (third time's
+	  the charm?)
+
+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
+

[... 6022 lines stripped ...]



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