[svn-commits] tilghman: tag 1.2.34 r211589 - in /tags/1.2.34: .lastclean .version ChangeLog
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Aug 10 14:59:20 CDT 2009
Author: tilghman
Date: Mon Aug 10 14:59:16 2009
New Revision: 211589
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=211589
Log:
Importing files for 1.2.34 release.
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tags/1.2.34/.lastclean (with props)
tags/1.2.34/.version (with props)
tags/1.2.34/ChangeLog (with props)
Added: tags/1.2.34/.lastclean
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+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
+
+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,
[... 5970 lines stripped ...]
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