aboutsummaryrefslogtreecommitdiffstats
path: root/ChangeLog
diff options
context:
space:
mode:
authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-06 21:19:27 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-06 21:19:27 +0000
commit111358cd43b8cc909cb06edd9cfeeff53c3b7a42 (patch)
tree34af0ae22ae1b77b22cde550412dc8ba103a0b7c /ChangeLog
parent57f10b9925c2de146c0baed5811a2736adfbc439 (diff)
Importing files for 1.2.31 release
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.2.31@167272 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'ChangeLog')
-rw-r--r--ChangeLog6778
1 files changed, 6778 insertions, 0 deletions
diff --git a/ChangeLog b/ChangeLog
new file mode 100644
index 000000000..bf278509d
--- /dev/null
+++ b/ChangeLog
@@ -0,0 +1,6778 @@
+2009-01-06 Leif Madsen <lmadsen@digium.com>
+
+ * Asterisk 1.2.31 released
+
+2009-01-06 20:44 +0000 [r167259] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Security fix AST-2009-001.
+
+2008-12-10 Tilghman Lesher <tlesher@digium.com>
+
+ * Asterisk 1.2.30.4 released
+
+2008-12-10 21:06 +0000 [r162868] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Fix for AST-2008-012
+
+2008-12-05 20:50 +0000 [r161421] Sean Bright <sean.bright@gmail.com>
+
+ * include/asterisk/astobj2.h, astobj2.c: Fix build errors on
+ FreeBSD (uint -> unsigned int). (closes issue #14006) Reported
+ by: alphaque Patches: astobj2.h-patch uploaded by alphaque
+ (license 259) (Slightly modified by seanbright)
+
+2008-12-01 Tilghman Lesher <tlesher@digium.com>
+
+ * Asterisk 1.2.30.3 released
+
+2008-11-25 21:37 +0000 [r159245] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Regression fix for last security fix. Set
+ the iseqno correctly. (closes issue #13918) Reported by:
+ ffloimair Patches: 20081119__bug13918.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: ffloimair
+
+2008-08-09 Tilghman Lesher <tlesher@digium.com>
+
+ * Asterisk 1.2.30.2 released
+
+2008-08-09 15:24 +0000 [r136945] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/compat.h, include/asterisk/astobj2.h: Regression
+ fixes for Solaris
+
+2008-07-25 15:00 +0000 [r133577] Russell Bryant <russell@digium.com>
+
+ * LICENSE: Fix the IAX2 URI for calling Digium
+
+2008-07-23 Tilghman Lesher <tlesher@digium.com>
+
+ * Asterisk 1.2.30.1 released
+
+2008-07-24 03:46 +0000 [r133360] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: This part was not correctly patched for
+ AST-2008-010.
+
+2008-07-22 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.30 released
+
+2008-07-22 21:14 +0000 [r132711] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/iax.conf.sample, channels/chan_iax2.c: Fixes for
+ AST-2008-010 and AST-2008-011
+
+2008-06-03 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.29 released
+
+2008-06-03 19:30 +0000 [r120109] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Copy the From header into a variable so that
+ pedantic SIP handling does not try to mess with a NULL pointer.
+ (AST-2008-008) (closes issue #12607) Reported by: hooi
+
+2008-05-30 12:49 +0000 [r119008-119237] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: - Instead of only enforcing destination
+ call number checking on an ACK, check all full frames except for
+ PING and LAGRQ, which may be sent by older versions too quickly
+ to contain the destination call number. (As suggested by Tim
+ Panton on the asterisk-dev list) - Merge changes from
+ team/russell/iax2-frame-race, which prevents PING and LAGRQ from
+ being sent before the destination call number is known.
+
+ * channels/chan_iax2.c: Merge changes from
+ team/russell/iax2-another-fix-to-the-fix As described in the
+ following post to the asterisk-dev mailing list, only enforce
+ destination call numbers when processing an ACK.
+ http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
+
+2008-05-21 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.28.1 released
+
+2008-05-08 19:14 +0000 [r115564] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a race condition that bbryant just
+ found while doing some IAX2 testing. He was running Asterisk
+ trunk running IAX2 calls through a few Asterisk boxes, however,
+ the audio was extremely choppy. We looked at a packet trace and
+ saw a storm of INVAL and VNAK frames being sent from one box to
+ another. It turned out that what had happened was that one box
+ tried to send a CONTROL frame before the 3 way handshake had
+ completed. So, that frame did not include the destination call
+ number, because it didn't have it yet. Part of our recent work
+ for security issues included an additional check to ensure that
+ frames that are supposed to include the destination call number
+ have the correct one. This caused the frame to be rejected with
+ an INVAL. The frame would get retransmitted for forever, rejected
+ every time ... This race condition exists in all versions that
+ got the security changes, in theory. However, it is really only
+ likely that this would cause a problem in Asterisk trunk. There
+ was a control frame being sent (SRCUPDATE) at the _very_
+ beginning of the call, which does not exist in 1.2 or 1.4.
+ However, I am fixing all versions that could potentially be
+ affected by the introduced race condition. These changes are what
+ bbryant and I came up with to fix the issue. Instead of simply
+ dropping control frames that get sent before the handshake is
+ complete, the code attempts to wait a little while, since in most
+ cases, the handshake will complete very quickly. If it doesn't
+ complete after yielding for a little while, then the frame gets
+ dropped.
+
+2008-05-07 16:22 +0000 [r115511] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/dlinkedlists.h (removed), channels/chan_iax2.c:
+ Remove remnants of dlinkedlists. I didn't actually use them in
+ the final version of my IAX2 improvements.
+
+2008-05-06 19:54 +0000 [r115421] Jason Parker <jparker@digium.com>
+
+ * contrib/scripts/get_ilbc_source.sh: read requires an argument on
+ some non-bash shells (closes issue #12593) Reported by: bkruse
+ Patches: getilbc.sh_12593_v1.diff uploaded by bkruse (license
+ 132)
+
+2008-05-05 17:53 +0000 [r115296] Russell Bryant <russell@digium.com>
+
+ * Makefile, include/asterisk/astobj2.h (added), astobj2.c (added),
+ include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
+ Merge changes from team/russell/iax2_find_callno_1.2 These
+ changes address a critical performance issue introduced in the
+ latest release. The fix for the latest security issue included a
+ change that made Asterisk randomly choose call numbers to make
+ them more difficult to guess by attackers. However, due to some
+ inefficient (this is by far, an understatement) code, when
+ Asterisk chose high call numbers, chan_iax2 became unusable after
+ just a small number of calls. On a small embedded platform, it
+ would not be able to handle a single call. On my Intel Core 2 Duo
+ @ 2.33 GHz, I couldn't run more than about 16 IAX2 channels.
+ Ouch. These changes address some performance issues of the
+ find_callno() function that have bothered me for a very long
+ time. On every incoming media frame, it iterated through every
+ possible call number trying to find a matching active call. This
+ involved a mutex lock and unlock for each call number checked.
+ So, if the random call number chosen was 20000, then every media
+ frame would cause 20000 locks and unlocks. Previously, this
+ problem was not as obvious since Asterisk always chose the lowest
+ call number it could. A second container for IAX2 pvt structs has
+ been added. It is an astobj2 hash table. When we know the remote
+ side's call number, the pvt goes into the hash table with a hash
+ value of the remote side's call number. Then, lookups for
+ incoming media frames are a very fast hash lookup instead of an
+ absolutely insane array traversal. In a quick test, I was able to
+ get more than 3600% more IAX2 channels on my machine with these
+ changes.
+
+2008-04-29 12:52 +0000 [r114822] Kevin P. Fleming <kpfleming@digium.com>
+
+ * contrib/scripts/get_ilbc_source.sh: stop script from appending
+ source code if run multiple times
+
+2008-04-22 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.28 released
+
+2008-04-22 22:20 +0000 [r114561] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: When we receive a full frame that is
+ supposed to contain our call number, ensure that it has the
+ correct one. (closes issue #10078) (AST-2008-006)
+
+2008-03-26 19:49 +0000 [r110869-111125] Kevin P. Fleming <kpfleming@digium.com>
+
+ * UPGRADE.txt: update UPGRADE notes to document usage of the script
+
+ * contrib/scripts/get_ilbc_source.sh (added), codecs/ilbc: add a
+ script to make getting the iLBC source code simple for end users
+
+ * codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/packing.h
+ (removed), codecs/ilbc/getCBvec.c (removed),
+ codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
+ (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
+ (removed), codecs/ilbc/getCBvec.h (removed),
+ codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
+ (removed), codecs/ilbc/FrameClassify.c (removed),
+ codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
+ codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
+ (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
+ (removed), codecs/ilbc/anaFilter.c (removed),
+ codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
+ (removed), codecs/ilbc/doCPLC.h (removed),
+ codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
+ codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
+ (removed), codecs/ilbc/createCB.h (removed),
+ codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
+ (removed), codecs/ilbc/iCBSearch.c (removed),
+ codecs/ilbc/filter.c (removed), codecs/ilbc/gainquant.c
+ (removed), codecs/ilbc/hpInput.c (removed),
+ codecs/ilbc/hpOutput.c (removed), codecs/ilbc/iCBSearch.h
+ (removed), codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h
+ (removed), codecs/ilbc/gainquant.h (removed),
+ codecs/ilbc/LPCencode.c (removed), codecs/ilbc/hpOutput.h
+ (removed), codecs/ilbc/StateSearchW.c (removed),
+ codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
+ (removed), codecs/ilbc/iCBConstruct.c (removed),
+ codecs/ilbc/syntFilter.c (removed), codecs/ilbc/iCBConstruct.h
+ (removed), codecs/ilbc/syntFilter.h (removed),
+ codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
+ (removed): due to licensing restrictions, we cannot distribute
+ the source code for iLBC encoding and decoding... so remove it,
+ and add instructions on how the user can obtain it themselves
+
+2008-03-20 21:53 +0000 [r110335] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c, channels/chan_iax2.c: Fix some very broken
+ code that was introduced in 1.2.26 as a part of the security fix.
+ The dnsmgr is not appropriate here. The dnsmgr takes a pointer to
+ an address structure that a background thread continuously
+ updates. However, in these cases, a stack variable was passed.
+ That means that the dnsmgr thread would be continuously writing
+ to bogus memory.
+
+2008-03-18 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.27 released
+
+2008-03-18 16:27 +0000 [r109488] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/astobj.h: Fix character string being treated as
+ format string
+
+2008-03-18 15:08 +0000 [r109391] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Do not return with a successful
+ authentication if the From header ends up empty. (AST-2008-003)
+
+2008-01-22 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.26.2 released
+
+2008-01-07 20:46 +0000 [r96931] Russell Bryant <russell@digium.com>
+
+ * configs/extensions.conf.sample: Change misery.digium.com to
+ pbx.digium.com
+
+2007-12-23 01:30 +0000 [r94661] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Fix for fix for security fix (third time's
+ the charm?)
+
+2007-12-20 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.26.1 released
+
+2007-12-20 20:21 +0000 [r94214-94255] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix another potential seg fault ... (closes
+ issue #11606) Reported by: dimas
+
+ * channels/chan_iax2.c: Fix a couple of places where it's possible
+ to dereference a NULL pointer.
+
+2007-12-18 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.26 released
+
+2007-12-18 18:44 +0000 [r93667-93675] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, channels/chan_iax2.c: Fixing AST-2007-027
+ (Closes issue #11119)
+
+2007-11-29 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.25 released
+
+2007-11-29 21:10 +0000 [r90170] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_pgsql.c: Properly escape src and dst fields (Fixes
+ AST-2007-026)
+
+2007-09-13 18:10 +0000 [r82334] Kevin P. Fleming <kpfleming@digium.com>
+
+ * LICENSE: clarify the OpenSSL and OpenH323 license exceptions
+
+2007-08-07 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.24 released
+
+2007-08-07 17:44 +0000 [r78370] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: Revert patch committed for issue #9660. It
+ broke E&M trunks. (closes issue #10360) (closes issue #10364)
+
+2007-08-02 17:56 +0000 [r77942] Steve Murphy <murf@digium.com>
+
+ * fskmodem.c: This patch hopefully solves 10141; The user is
+ running with it, and it doesn't appear to harm asterisk's
+ operation, and may prevent a crash. I'll store it in 1.2, as we
+ have shut down support on 1.2, but since I developed the patch
+ before support finished, and it might affect 1.4 and trunk, I'm
+ going ahead with it.
+
+2007-07-31 19:19 +0000 [r77842] Steve Murphy <murf@digium.com>
+
+ * contrib/scripts/ast_grab_core: This probably isn't super-general,
+ but it's a first stab at using kill -11 to generate a core file
+ instead of gcore.
+
+2007-07-30 18:40 +0000 [r77782] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * res/res_agi.c: Revert change in revision 71656, even though it
+ fixed a bug, because many people were depending upon the (broken)
+ behavior.
+
+2007-07-30 14:50 +0000 [r77767] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_macro.c: (closes issue #10334) Reported by: ramonpeek
+ Pass through the return value from macro_exec through the MacroIf
+ application.
+
+2007-07-25 00:07 +0000 [r76978] Steve Murphy <murf@digium.com>
+
+ * channels/chan_zap.c: this fixes bug 10293, where the error
+ message because defaultzone or loadzone was not defined was
+ confusing
+
+2007-07-24 22:11 +0000 [r76934] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk/lock.h: Oops, res contains the error code, not
+ errno. I was wondering why a mutex was reporting "No such file or
+ directory"...
+
+2007-07-24 Jason Parker <jparker@digium.com>
+
+ * Asterisk 1.2.23 released
+
+2007-07-24 16:32 +0000 [r76802] Jason Parker <jparker@digium.com>
+
+ * channels/chan_iax2.c: Don't create the Asterisk channel until we
+ are starting the PBX on it. (ASA-2007-018)
+
+2007-07-23 18:28 +0000 [r76560-76653] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_agent.c: (closes issue #5866) Reported by: tyler Do
+ not force channel format changes when a generator is present. The
+ generator may have changed the formats itself and changing them
+ back would cause issues.
+
+ * channels/chan_sip.c: (closes issue #10236) Reported by: homesick
+ Patches: rpid_1.4_75840.patch uploaded by homesick (license 91)
+ Accept Remote Party ID on guest calls.
+
+2007-07-22 21:39 +0000 [r76409] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk/app.h: We should not use C++ reserved words in
+ API headers (closes issue #10266)
+
+2007-07-21 02:01 +0000 [r76226] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Backport a fix for a memory leak that was
+ fixed in trunk in reivision 76221 by rizzo. The memory used for
+ the localaddr list was not freed during a configuration reload.
+
+2007-07-20 17:16 +0000 [r76080] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: (closes issue #10247) Reported by:
+ fkasumovic Patches: chan_sip.patch uploaded by fkasumovic
+ (license #101) Drop any peer realm authentication entries when
+ reloading so multiple entries do not get added to the peer.
+
+2007-07-19 15:49 +0000 [r75757-75927] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: When processing full frames, take sequence
+ number wraparound into account when deciding whether or not we
+ need to request retransmissions by sending a VNAK. This code
+ could cause VNAKs to be sent erroneously in some cases, and to
+ not be sent in other cases when it should have been. (closes
+ issue #10237, reported and patched by mihai)
+
+ * channels/chan_iax2.c: When traversing the queue of frames for
+ possible retransmission after receiving a VNAK, handle sequence
+ number wraparound so that all frames that should be retransmitted
+ actually do get retransmitted. (issue #10227, reported and
+ patched by mihai)
+
+2007-07-18 20:31 +0000 [r75748] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Store prior to copy (closes issue #10193)
+
+2007-07-18 17:48 +0000 [r75657] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * apps/app_queue.c: removed the word 'pissed' from ast_log(...)
+ function call for BE-90
+
+2007-07-17 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.22 released
+
+2007-07-17 20:57 +0000 [r75440-75449] Russell Bryant <russell@digium.com>
+
+ * channels/chan_skinny.c: Properly check for the length in the
+ skinny packet to prevent an invalid memcpy. (ASA-2007-016)
+
+ * channels/iax2-parser.h, channels/chan_iax2.c,
+ channels/iax2-parser.c: Ensure that when encoding the contents of
+ an ast_frame into an iax_frame, that the size of the destination
+ buffer is known in the iax_frame so that code won't write past
+ the end of the allocated buffer when sending outgoing frames.
+ (ASA-2007-014)
+
+ * channels/chan_iax2.c: After parsing information elements in IAX
+ frames, set the data length to zero, so that code later on does
+ not think it has data to copy. (ASA-2007-015)
+
+2007-07-16 20:46 +0000 [r75251-75304] Kevin P. Fleming <kpfleming@digium.com>
+
+ * dns.c: provide proper copyright/license attribution for this
+ structure that was copied from a BSD-licensed header file long,
+ long ago...
+
+ * Makefile: install the LICENSE file along with the music files
+
+ * sounds/fpm-world-mix.mp3 (removed), sounds/moh/fpm-calm-river.mp3
+ (added), Makefile, sounds/moh (added),
+ sounds/moh/fpm-world-mix.mp3 (added), sounds/moh/LICENSE (added),
+ sounds/fpm-sunshine.mp3 (removed), sounds/moh/fpm-sunshine.mp3
+ (added), sounds/fpm-calm-river.mp3 (removed): move FreePlayMusic
+ files into a subdirectory, and include a license statement for
+ them
+
+2007-07-13 20:35 +0000 [r75107] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Fix a couple potential minor memory leaks.
+ load_moh_classes() could return without destroying the loaded
+ configuration.
+
+2007-07-13 20:10 +0000 [r75066] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Fixed an issue where chanspy flags were
+ uninitialized if no options were passed. What triggered this
+ investigation was an IRC chat where some people's quiet flags
+ were set while others' weren't even though none of them had
+ specified the q option.
+
+2007-07-13 20:07 +0000 [r75052-75059] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Ensure that adding a user to the list of
+ users of a specific music on hold class is not done at the same
+ time as any of the other operations on this list to prevent list
+ corruption. Using the global moh_data lock for this is not ideal,
+ but it is what is used to protect these lists everywhere else in
+ the module, and I am only changing what is necessary to fix the
+ bug.
+
+ * channels/chan_zap.c: (closes issue #9660) Reported by: mmacvicar
+ Patches submitted by: bbryant, russell Tested by: mmacvicar,
+ marco, arcivanov, jmhunter, explidous When using a TDM400P (and
+ probably other analog cards) there was a chance that you could
+ hang up and pick the phone back up where it has been long enough
+ to be not considered a flash hook, but too soon such that the
+ device reports that it is busy and the person on the phone will
+ only hear silence. This patch makes chan_zap more tolerant of
+ this and gives the device a couple of seconds to succeed so the
+ person on the phone happily gets their dialtone.
+
+2007-07-12 15:51 +0000 [r74814] Joshua Colp <jcolp@digium.com>
+
+ * res/res_musiconhold.c: Only print out a warning for situations
+ where it is actually helpful. (issue #10187 reported by denke)
+
+2007-07-11 22:53 +0000 [r74766] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: The function make_trunk() can fail and
+ return -1 instead of a valid new call number. Fix the uses of
+ this function to handle this instead of treating it as the new
+ call number. This would cause a deadlock and memory corruption.
+ (possible cause of issue #9614 and others, patch by me)
+
+2007-07-11 21:12 +0000 [r74719] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: The cli command "agent logoff Agent/x
+ soft" did not work...at all. Now it does. (closes issue #10178,
+ reported and patched by makoto, with slight modification for 1.4
+ and trunk by me)
+
+2007-07-11 18:33 +0000 [r74656] Russell Bryant <russell@digium.com>
+
+ * res/res_config_odbc.c: Make sure that the ESCAPE immediately
+ follows the condition that uses LIKE. This fixes realtime
+ extensions with ODBC. (closes issue #10175, reported by stuarth,
+ patch by me)
+
+2007-07-11 17:15 +0000 [r74587] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_phone.c, channels/Makefile: Use some Makefile magic
+ to determine if linux/compiler.h is present. (issue #10174
+ reported by francesco_r)
+
+2007-07-10 19:57 +0000 [r74373-74427] Jason Parker <jparker@digium.com>
+
+ * apps/app_queue.c: Fix an issue where it was possible to have a
+ service level of over 100% Between the time recalc_holdtime and
+ update_queue was called, it was possible that the call could have
+ been hungup. Move both additions to the same place, so this won't
+ happen. Issue 10158, initial patch by makoto, modified by me.
+
+ * channels/chan_agent.c: Fix an issue with wrapuptime not working
+ when using AgentLogin. Issue 10169, patch by makoto, with a minor
+ mod by me to not re-break issue 9618
+
+ * dns.c: Use res_ndestroy on systems that have it. Otherwise, use
+ res_nclose. This prevents a memleak on NetBSD - and possibly
+ others. Issue 10133, patch by me, reported and tested by scw
+
+2007-07-10 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.21.1 released
+
+2007-07-10 15:37 +0000 [r74316] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Fix a small typo in description in of
+ Voicemail() application. Issue 10170, patch by casper.
+
+2007-07-10 15:30 +0000 [r74313] Russell Bryant <russell@digium.com>
+
+ * res/res_config_odbc.c: Only use ESCAPE when LIKE is used. (issue
+ #10075, this part reported by jmls on IRC, patch by me)
+
+2007-07-10 14:48 +0000 [r74264] Joshua Colp <jcolp@digium.com>
+
+ * app.c: Ensure the group information category exists before trying
+ to do a string comparison with it. (issue #10171 reported by
+ mlegas)
+
+2007-07-09 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.21 released
+
+2007-07-09 21:00 +0000 [r74165] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: When the specified class isn't found,
+ properly fall back to the channel's music class or the default.
+ (issue #10123, reported by blitzrage, patches from juggie, qwell,
+ and me)
+
+2007-07-09 20:18 +0000 [r74158] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Several chan_zap options were not working on
+ reload because they were arbitrarily disallowed when reloading
+ some/most PRI options (such as signalling) was disallowed.
+ Options such as polarityonanswerdelay and answeronpolarityswitch
+ can safely be changed on a reload. This corrects that behavior.
+ Issue 9186, patch by tzafrir.
+
+2007-07-06 23:01 +0000 [r73678-73768] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: If a sip_pvt struct has already registered
+ an extension state callback, remove the old one before adding a
+ new one. If this isn't done, Asterisk will crash. (issue #10120)
+
+ * res/res_config_odbc.c: (closes issue #10075) Reported by: apsaras
+ Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
+ with MSSQL 2005 by explicitly stating that '\' is being used as
+ an escape character.
+
+ * channels/chan_sip.c: (closes issue #10125) Reported by: makoto
+ Patches submitted by: makoto This fixes a crash in chan_sip that
+ happens when the bindaddr setting is not valid on Asterisk
+ startup, gets fixed, and then a reload gets issued.
+
+2007-07-06 15:26 +0000 [r73674] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Fixed a bug wherein agents get stuck busy.
+ (issue 9618, reported by jiddings, patched by moi) closes issue
+ #9618
+
+2007-07-05 22:11 +0000 [r73547] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: we shouldn't allow G.723.1 endpoints to use
+ VAD, just like we don't support it for G.729
+
+2007-07-05 19:15 +0000 [r73315-73466] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Copy language information to the dialog
+ structure when calling a peer for situations where a PBX may be
+ started on the dialed channel. (issue #10121 reported by
+ clegall_proformatique)
+
+ * apps/app_chanspy.c, channel.c: Tweak spy locking. (issue #9951
+ reported by welles)
+
+ * channels/chan_local.c: Actually check to make sure a PBX was
+ started on one of the Local channels instead of blindly assuming
+ it was. (issue #10112 reported by makoto)
+
+ * apps/app_queue.c: Reset ServicelevelPerf variable back to 0 if we
+ are unable to calculate it each time... otherwise we will get
+ previous values. (issue #10117 reported by noriyuki)
+
+2007-07-04 14:50 +0000 [r73207-73252] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: bchannel configurations like
+ echocancel and volume control, need to be setuped on inbound
+ calls too.
+
+ * channels/chan_misdn.c: bad bug in overlapdial case, we called
+ start_pbx multiple times, because the state wasn't changed..
+
+2007-07-03 12:34 +0000 [r73052] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_dial.c: RetryDial should accept a 0 argument, but it
+ does not, because atoi does not distinguish between 0 and error
+ (closes issue #10106)
+
+2007-07-03 08:04 +0000 [r73004] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed issue, that misdn_l2l1_check could
+ only be called from mISDN Source channels.. #9449
+
+2007-07-02 17:58 +0000 [r72924] Jason Parker <jparker@digium.com>
+
+ * say.c: Fix an issue with playing "oclock" multiple times in
+ French with 24 hour time format. Issue 10101
+
+2007-07-01 23:51 +0000 [r72805] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_spool.c: When appending lines to call files to keep track
+ of retries, write a leading newline just in case the original
+ call file did not have a newline at the end. This fix is in
+ response to a problem I saw reported on the asterisk-users
+ mailing list.
+
+2007-06-29 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.20 released
+
+2007-06-29 16:30 +0000 [r72629] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Backport changes that make chan_iax2 not
+ start the PBX on an incoming channel until the three-way call
+ setup is completed. These changes are already in 1.4 and trunk.
+
+2007-06-29 13:08 +0000 [r72585] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: check if the
+ bchannel stack id is already used, if so don't use it a second
+ time. Also added a release_chan lock, so that the same chan_list
+ object cannot be freed twice. chan_misdn does not crash anymore
+ on heavy load with these changes.
+
+2007-06-27 23:24 +0000 [r72378] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_mixmonitor.c: Update documentation to clarify variable
+ usage with MixMonitor. (issue #9494 reported by netoguy)
+
+2007-06-27 23:22 +0000 [r72333-72373] Brett Bryant <bbryant@digium.com>
+
+ * asterisk.c: Reinstating patch. This actually fixes the problem,
+ however I was running a development branch without it and
+ mistakenly thought it wasn't fixed. Fixes issue #10010, and
+ #9654: 100% CPU usage caused by an asterisk console losing it's
+ controlling terminal.
+
+ * asterisk.c: Reverted changes for earlier revisions 72259 to
+ 72261. Issue #9654, #10010
+
+2007-06-27 22:43 +0000 [r72327] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_queue.c: Fix issue where queue log events might be
+ missing. (issue #7765 reported by mtryfoss)
+
+2007-06-27 21:06 +0000 [r72267] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_config.c: Fix a minor issue with parsing the priority
+ number. You could have as much whitespace as you want around a
+ numeric priority, but you couldn't have any whitespace around a
+ special priority like "n" or "hint". (issue #10039, reported by
+ mitheloc, fixed by me)
+
+2007-06-27 20:43 +0000 [r72259] Brett Bryant <bbryant@digium.com>
+
+ * asterisk.c: Fixes 100% load when controlling terminal disappears.
+ Issue #9654, #10010
+
+2007-06-27 20:23 +0000 [r72256] Joshua Colp <jcolp@digium.com>
+
+ * channel.c: I may possibly get shot for doing this... but... defer
+ CDR processing until after the channel has been dealt with. This
+ should eliminate all of the issues with channels going funky
+ (SIP/PRI) when you are posting CDRs to a database that is either
+ slow or unavailable and do not want to enable batching.
+
+2007-06-27 18:40 +0000 [r72184] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Fix another problem in voicemail with
+ missing symbols. Issue 10074, patch by kryptolus, extended to
+ include #if 0'd blocks (just in case)
+
+2007-06-27 13:22 +0000 [r72040-72099] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
+ simplified generation for dummy bchannels, also we mark them as
+ dummies, so they are not used later as real-bchannels, optimized
+ the RESTART mechanisms, we block a channel now on cause:44, and
+ send out a RESTART automatically, then on reception of
+ RESTART_ACKNOWLEDGE we unblock the channel again.
+
+ * channels/misdn/isdn_lib.h, channels/misdn/isdn_lib.c: simplified
+ channel finding and locking a lot. removed unnecessary #ifdefed
+ areas.
+
+ * channels/misdn/isdn_lib.c: isdn_lib.c didn't compile
+
+ * channels/misdn/isdn_lib.c: for inbound TE calls, we setup the
+ bchannel when we get the CONNECT_ACKNOWLEDGE, to make sure mISDN
+ has everything ready. removed some #if 0 areas which weren't used
+ anymore.
+
+2007-06-26 17:49 +0000 [r71847] Jason Parker <jparker@digium.com>
+
+ * Makefile: Don't try to install an init script that doesn't exist.
+ Reported to me on #asterisk on Freenode IRC.
+
+2007-06-26 12:25 +0000 [r71656-71750] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Issue 10062 - Trying to move a message
+ without selecting one first results in memory corruption
+
+ * res/res_agi.c: Issue 10035 - handle_exec returns a result
+ inconsistent with all of the other AGI commands
+
+2007-06-25 01:02 +0000 [r71414] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Ignore other URIs after the first in a 300
+ Multiple Choice response. (issue #10041 reported by homesick)
+
+2007-06-24 20:04 +0000 [r71358] Russell Bryant <russell@digium.com>
+
+ * asterisk.c: Revert the patch from issue 9654 due to an unexpected
+ side effect
+
+2007-06-24 17:32 +0000 [r71288] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * manager.c, db.c: Issue 10043 - There is a legitimate need to be
+ able to set variables to the empty string.
+
+2007-06-22 16:02 +0000 [r71124] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Send an unhold indication when going off
+ hold. (issue #10036 reported by speedy)
+
+2007-06-22 14:52 +0000 [r71065] Jason Parker <jparker@digium.com>
+
+ * file.c, res/res_agi.c: Fix a few silly usages of ast_playstream()
+ - it only ever returns 0... Issue 10035
+
+2007-06-22 14:39 +0000 [r71064] Brett Bryant <bbryant@digium.com>
+
+ * asterisk.c: Fixed infinite loop when controlling terminal was
+ lost and return value of input function wasn't checked for
+ errors. This would cause 100% cpu to be taken up. (closes issue
+ #9654, issue #10010) Reported by: mnicholson, and eserra Idea for
+ the patch from mnicholson, patched by me
+
+2007-06-21 22:29 +0000 [r70948] Steve Murphy <murf@digium.com>
+
+ * cdr.c: This little fix is in response to bug 10016, but may not
+ cure it. The code is wrong, clearly. In a situation where you set
+ the CDR's amaflags, and then ForkCDR, and then set the new CDR's
+ amaflags to some other value, you will see that all CDRs have had
+ their amaflags changed. This is not good. So I fixed it.
+
+2007-06-21 21:37 +0000 [r70898] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Don't explode if the gain option is
+ specified without a value. (issue #9274 reported by mfarver)
+
+2007-06-21 19:13 +0000 [r70804] Steve Murphy <murf@digium.com>
+
+ * cdr/cdr_custom.c: it was pointed out that the cdr_custom config
+ load could get a lock, and under certain circumstances, would
+ never release it. I also noted that the situation where more than
+ one mapping spec was warned about, but did not ignore further
+ mappings as it had promised. I think I have fixed both
+ situations.
+
+2007-06-21 13:11 +0000 [r70672] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: we activate the
+ bchannels in TE mode on incoming calls only when we want to
+ connect the call.
+
+2007-06-20 22:20 +0000 [r70551] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Don't overwrite the configured username
+ setting upon a REGISTER. (issue #8565 reported by jsmith)
+
+2007-06-20 19:25 +0000 [r70444] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_dial.c: Issue 9997 - Timelimit times out the wrong
+ channel
+
+2007-06-20 18:45 +0000 [r70396] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: Fix a problem where an established call
+ would not be properly disconnected when a PRI disconnect is
+ received depending on which cause code was received. (issue
+ #9588, original patch by softins, updated patch from jtexter3,
+ and some additional feedback from mhardeman)
+
+2007-06-20 15:42 +0000 [r70311-70342] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: forgot one place ..
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/isdn_lib.c: fixed a bug that was introduced by
+ copy and paste in the last commit ..bchannels weren't cleaned
+ properly.
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/isdn_lib.c: on receiption of cause:44 we mark the
+ channel as in use and inform the user about the situation, we
+ need to test the RESTART stuff then. Also shuffled the
+ empty_chan_in_stack function after the bchannel cleaning
+ functions, to avoid race conditions.
+
+2007-06-19 18:07 +0000 [r70053] Steve Murphy <murf@digium.com>
+
+ * channel.c: This fixes 9246, where channel variables are not
+ available in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
+ consolidate the channel variables during a masquerade, and then
+ copy the merged variables back onto the clone, so the zombie has
+ the same vars that the 'original' has.
+
+2007-06-19 17:00 +0000 [r69992] Joshua Colp <jcolp@digium.com>
+
+ * rtp.c: Handle the CC field in the RTP header. (issue #9384
+ reported by DoodleHu)
+
+2007-06-19 16:45 +0000 [r69990] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Backport fix for crashes related to
+ subscriptions from 1.4 ... Fix a crash that could occur when
+ handing device state changes. When the state of a device changes,
+ the device state thread tells the extension state handling code
+ that it changed. Then, the extension state code calls the
+ callback in chan_sip so that it can update subscriptions to that
+ extension. A pointer to a sip_pvt structure is passed to this
+ function as the call which needs a NOTIFY sent. However, there
+ was no locking done to ensure that the pvt struct didn't
+ disappear during this process. (issue #9946, reported by
+ tdonahue, patch by me, patch updated to trunk to use the sip_pvt
+ lock wrappers by eliel)
+
+2007-06-19 16:21 +0000 [r69894-69986] Joshua Colp <jcolp@digium.com>
+
+ * channel.c: Update BRIDGEPEER variable if set to the new channel
+ name when a masquerade happens. (issue #9699 reported by dimas)
+
+ * apps/app_meetme.c: Perform an extra hangup check just in case.
+ (issue #9589 reported by bcnit)
+
+2007-06-19 13:23 +0000 [r69887] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: when we send out a SETUP, but get no
+ response, we should cleanup everything after reception of a
+ hangup.
+
+2007-06-19 12:57 +0000 [r69765-69846] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Add parked call extension AFTER the parking
+ slot has been announced, otherwise two threads will try to handle
+ the same channel and it will go kaboom. (issue #9191 reported by
+ japple)
+
+ * channels/chan_sip.c: Set the peer name on the dialog to the one
+ configured in sip.conf and NOT the username to be used for
+ authentication attempts. (issue #9967 reported by achauvin)
+
+2007-06-18 17:45 +0000 [r69743] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * contrib/scripts/safe_asterisk: Issue 9998 - Remove SIG prefix,
+ since it's not supported by ksh
+
+2007-06-15 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.19 released
+
+2007-06-14 23:21 +0000 [r69469] Jason Parker <jparker@digium.com>
+
+ * config.c: Fix an issue where the line number in an unterminated
+ comment block error message would show the wrong line number.
+ "Reported" to me on #asterisk (somebody posted an error message,
+ and I happened to catch it)
+
+2007-06-14 20:56 +0000 [r69347] Russell Bryant <russell@digium.com>
+
+ * channel.c: Backport rev 69010 from the 1.4 branch ... In
+ ast_channel_make_compatible(), just return if the channels' read
+ and write formats already match up. There are code paths that
+ call this function on a pair of channels multiple times. This
+ made calls fail that were using g729 in some cases. The reason is
+ that codec_g729a will unregister itself from the list of
+ available translators will all licenses are in use. So, the first
+ time the function got called, the right translation path was
+ allocated. However, the second time it got called, the code would
+ not find a translation path to/from g729 and make the call fail,
+ even if the channel actually already had a g729 translation path
+ allocated. (SPD-32)
+
+2007-06-14 15:15 +0000 [r69258] Jason Parker <jparker@digium.com>
+
+ * funcs/func_groupcount.c: Change a quite broken while loop to a
+ for loop, so "continue;" works as expected instead of eating 99%
+ CPU... Issue 9966, patch by me.
+
+2007-06-13 18:12 +0000 [r69127] Joshua Colp <jcolp@digium.com>
+
+ * app.c: Return group counting to previous behavior where you could
+ only have one group per category. (issue #9711 reported by
+ irroot)
+
+2007-06-13 09:55 +0000 [r69053] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_msg_parser.c: restart indicator 0x80 is
+ correct, at least that's what libpri does.
+
+2007-06-12 14:18 +0000 [r68921] Joshua Colp <jcolp@digium.com>
+
+ * rtp.c: Bring RTP back to Asterisk at the end of a native bridge
+ no matter what.
+
+2007-06-12 08:35 +0000 [r68732-68887] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: if the bridged partner is mISDN too we
+ should not send dtmf tones, they are transmitted inband always
+
+ * channels/chan_misdn.c: if we have already some digits, we just
+ stop the tones.
+
+ * channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
+ check for NULL Pointer when calling misdn_new. Asterisk does not
+ allow us to create channels anymore when stop gracefully is used
+ :). also modified the restart_indicator to 0
+
+2007-06-11 14:29 +0000 [r68682] Joshua Colp <jcolp@digium.com>
+
+ * channel.c: Improve deadlock handling of the channel list. (issue
+ #8376 reported by one47)
+
+2007-06-11 09:18 +0000 [r68631] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
+ channels/misdn/isdn_lib.c: fixed problem that the dummybc chanels
+ had no lock, checking for the lock now. Also fixed the channel
+ restart stuff, we can now specify and restart particular channels
+ too.
+
+2007-06-08 22:22 +0000 [r68526] Russell Bryant <russell@digium.com>
+
+ * apps/app_dictate.c: Don't automatically hang up after running
+ Dictate so that callers can exit cleanly using '#' (closes issue
+ #9577, patch from Thomas Andrews)
+
+2007-06-08 00:15 +0000 [r68368-68397] Joshua Colp <jcolp@digium.com>
+
+ * say.c: Don't call ast_waitstream_full when the control file
+ descriptor and audio file descriptor are not set, simply call
+ ast_waitstream! (issue #8530 reported by rickead2000)
+
+ * dnsmgr.c: Do a DNS lookup immediately upon calling the dnsmgr
+ function, don't wait until a refresh happens. (issue #9097
+ reported by plack)
+
+2007-06-07 23:13 +0000 [r68351] Russell Bryant <russell@digium.com>
+
+ * say.c: Fix a problem where saying a character wouldn't properly
+ break out when the caller pressed '#' (issue #8113, reported by
+ patbaker82, patch from jamesgolovich (hey, long time no see!) and
+ patbaker82)
+
+2007-06-07 20:02 +0000 [r68204] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Don't try to save voicemail greetings
+ unless the user presses '1' to accept/save. Issue 9904, patch by
+ me.
+
+2007-06-07 14:19 +0000 [r67938-68070] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Allow the 'g' option to work if used with the
+ 'S' option. (issue #9888 reported by gasparz)
+
+ * channels/chan_sip.c: Only notify the devicestate system of a peer
+ state change when the peer is built from the config file. (issue
+ #9900 reported by arkadia)
+
+2007-06-06 16:40 +0000 [r67715] Russell Bryant <russell@digium.com>
+
+ * channel.c: We have some bug reports showing crashes due to a
+ double free of a channel. Add a sanity check to
+ ast_channel_free() to make sure we don't go on trying to free a
+ channel that wasn't found in the channel list. (issue #8850, and
+ others...)
+
+2007-06-06 13:28 +0000 [r67593-67649] Joshua Colp <jcolp@digium.com>
+
+ * rtp.c: Reinvite the RTP back to the Asterisk machine when the
+ timeout happens. (issue #9888 reported by gasparz)
+
+ * devicestate.c: Revert channel name splitting fix for Zap. The
+ moral of the story is don't use - in your user/peer names. (issue
+ #9668 reported by stevedavies)
+
+2007-06-05 15:42 +0000 [r67306-67307] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/chan_misdn_config.h: briding is a bool, fixed copy
+ and paste issue.
+
+ * channels/chan_misdn.c: simplified the EVENT_SETUP handling in the
+ cb_events function a lot. Commented the different possibilities a
+ bit and made functions of shared code. When the dialed extension
+ does not exist in the extensions.conf we'll jump into the 'i'
+ extension if this does exist, else we disconnect the call with
+ the cause:1 = No Route to Destination.
+
+2007-06-05 11:18 +0000 [r67239] Nadi Sarrar <ns@beronet.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h: Backport of the overlap_dial
+ functionality from asterisk-1.4's chan_misdn.
+
+2007-06-05 10:05 +0000 [r67209] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h: added possibility to
+ deactivate bridging per port
+
+2007-06-04 23:41 +0000 [r67060-67161] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * funcs/func_math.c: According to MATH, 0+1181000386 = 1181000448.
+ Oops.
+
+ * contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.mandrake.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.mandrake.zaptel,
+ contrib/init.d/rc.slackware.asterisk: Add revision Id tags (by
+ request of tzafrir)
+
+2007-05-31 16:12 +0000 [r66764] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: It is now possible for this path of
+ execution to have the frame pointer be NULL, therefore we need to
+ check for it before trying to access it. (issue #9836 reported by
+ barthpbx)
+
+2007-05-31 15:58 +0000 [r66744] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_macro.c: Issue 9818 - Fix for issue 8329 breaks
+ pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime,
+ but only because we lack core API to do it.
+
+2007-05-29 21:49 +0000 [r66537] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * funcs/func_strings.c: If the value of a variable passed to
+ FIELDQTY is blank, then FIELDQTY should return 0, not 1.
+
+2007-05-29 07:53 +0000 [r66349] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #9802 - Change inuse counter on CANCEL
+
+2007-05-25 13:46 +0000 [r66127] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * say.c: Issue 9791 - Fix pronunciation of seconds in Dutch
+
+2007-05-24 14:40 +0000 [r65837] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Allow RFC2833 to be negotiated when an
+ INVITE comes in without SDP and is not matched to a user or peer.
+ (issue #9546 reported by mcrawford)
+
+2007-05-24 09:19 +0000 [r65767] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: we should only activate the generator in
+ chan_misdn, when asterisk hask not yet taken the call
+ (WAITING4DIGS state). Alerting audio will be generated fomr
+ asterisk for example.
+
+2007-05-23 20:46 +0000 [r65676-65682] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: ensure that variables are set on a newly
+ created channel before we start a PBX on it
+
+ * channels/chan_iax2.c: if we are going to set variables on a newly
+ created channel, it should be done *before* we start the PBX on
+ it
+
+2007-05-23 13:06 +0000 [r65389-65588] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: Revert revision 62417 as someone reported
+ problems with it to Mark. This was related to issue #9588.
+
+ * apps/app_queue.c: Fix a memory leak that I just noticed in the
+ device state handling in app_queue. On most device state changes,
+ it would leak roughly 8 to 64 bytes (the length of the name of
+ the device).
+
+2007-05-22 07:46 +0000 [r65328] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: we stop the tones only when we're in the
+ pre-call phase, otherwise e.g. when in CONNECTED state we should
+ not stop tones when we receive an Information Message
+
+2007-05-18 20:56 +0000 [r65172] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, include/asterisk/cdr.h, cdr.c: This update will
+ fix the situation that occurs as described by 9717, where when
+ several targets are specified for a dial, if any one them reports
+ FAIL, the whole call gets FAIL, even though others were ringing
+ OK. I rearranged the priorities, so that a new disposition, NULL,
+ is at the lowest level, and the disposition get init'd to NULL.
+ Then, next up is FAIL, and next up is BUSY, then NOANSWER, then
+ ANSWERED. All the related set routines will only do so if the
+ disposition value to be set to is greater than what's already
+ there. This gives the intended effect. So, if all the targets are
+ busy, you'd get BUSY for the call disposition. If all get BUSY,
+ but one, and that one rings is not answered, you get NOANSWER. If
+ by some freak of nature, the NULL value doesn't get overridden,
+ then the disp2str routine will report NOANSWER as before.
+
+2007-05-18 18:10 +0000 [r65075-65122] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Not getting an ACK to a 200 OK in the
+ initial invite is critical to the call.
+
+ * channels/chan_sip.c: Issue 9235 - part of the problem, maybe not
+ all. Please retry with this patch (and no other patch) if you
+ have problems with hanging SIP channels. Thank you. A special
+ Thank You to WeBRainstorm that gave me access to his system.
+
+2007-05-18 11:23 +0000 [r64902-65007] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: fixed a
+ warning regarding Keypad encoding. encode the IE sending_complete
+ at the right position.
+
+ * channels/chan_misdn.c: we *need* to send a PROCEEDING when
+ sending_complete is set, even if need_more_infos is requested.
+
+2007-05-17 21:14 +0000 [r64819] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk/linkedlists.h: How is it that we never caught
+ that this is returning the opposite of our documentation, until
+ now?
+
+2007-05-17 16:52 +0000 [r64758] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: If we have a negative current message, we
+ shouldn't go back even further... Issue 9727.
+
+2007-05-16 10:55 +0000 [r64514-64603] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Fixing possible bug in auth of BYE
+
+ * channels/chan_sip.c: Support SIP uri's starting with SIP: and
+ sip: (reported by Tony Mountfield on the mailing list. Thanks!)
+
+ * channels/chan_sip.c: Issue #9726 - rlister - Better logging for
+ ACL denials While at it, also added better logging and handling
+ of peers that are not supposed to register. My patch, stole the
+ issue report from Russell. My apologies, Russell :-)
+
+2007-05-16 08:23 +0000 [r64513] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: in the case immediate=yes, we directly
+ jump into the dialplan, where people can use PlayTones to
+ indicate a Dialtone, so we don't need to to that by ourself. also
+ we should not do a dialtone_indicate for incoming calls on a TE
+ port in overlapdialmode.
+
+2007-05-14 18:34 +0000 [r64275] Joshua Colp <jcolp@digium.com>
+
+ * devicestate.c: Only perform stripping of - strings from the
+ channel name for Zap channels. Anywhere else we might remove a
+ legitimate part of a device name. (issue #9668 reported by
+ stevedavies)
+
+2007-05-11 16:31 +0000 [r63903] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * Makefile, contrib/scripts/safe_asterisk: Issue 9121 - fixups for
+ safe_asterisk script
+
+2007-05-10 23:14 +0000 [r63828] Jason Parker <jparker@digium.com>
+
+ * channels/chan_iax2.c: Fix an issue with trying to kill a thread
+ before it gets created. Issue 9709, patch by nic_bellamy.
+
+2007-05-10 20:38 +0000 [r63748] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Do not allocate SIP pvt's for PEERs we can
+ not reach. This was seen as a lot of dialogs being created then
+ immediately destroyed at reload/restart of the SIP channel.
+
+2007-05-09 17:20 +0000 [r63653] Matthew Fredrickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Make sure we only create a DSP if it's
+ requested on SUB_REAL
+
+2007-05-09 16:51 +0000 [r63610] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Properly handle hints that point to multiple
+ devices in chan_sip. Why chan_sip is even doing this I have no
+ idea but I would rather not go into a rant. (issue #9536 reported
+ by rlister)
+
+2007-05-09 14:48 +0000 [r63565] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_directory.c: Replicate fix from 51158 (app_voicemail) to
+ app_directory (Issue 9224)
+
+2007-05-09 11:26 +0000 [r63519] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: release_chan frees ch, so we should never
+ touch ch after release_chan, this may cause segfaults.
+
+2007-05-08 22:19 +0000 [r63477] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_macro.c: Issue 9602 - segfault in app_macro
+
+2007-05-08 15:07 +0000 [r63402] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: added
+ application misdn_check_l2l1 which tries to pull up the L1/L2 on
+ all ports that have the layers down in a group. It waits then for
+ a timeout. This helps for scenarios where multiple PMP BRIs are
+ grouped together, or where a provider has a faulty PTP
+ Implementation, that looses the L2 after a while.
+
+2007-05-08 06:20 +0000 [r63359] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Issue 9527 - upon entering a folder, no
+ message is selected (curmsg == -1), so deleting causes memory
+ corruption (beyond bounds)
+
+2007-05-07 21:39 +0000 [r63285] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/app.h, app.c, channel.c: Properly handle what
+ happens during a masquerade in relation to group counting. (issue
+ #9657 reported by ramonpeek)
+
+2007-05-07 14:06 +0000 [r63195] Olle Johansson <oej@edvina.net>
+
+ * config.c: Releasing the whole in-memory configuration while
+ you're adding to it is not a good thing. Please review this
+ change. Caused by additions (+) to non-existing contexts.
+
+2007-05-03 16:42 +0000 [r62987] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: When a peer is seeded or built tell the
+ devicestate core to update it's status. This is easier then
+ having chan_sip load before pbx_config. (issue #9658 reported by
+ dlynes)
+
+2007-05-03 15:39 +0000 [r62885-62945] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: when
+ we're in state WAITING4DIGS, we use the asterisk tone-generator
+ which prods us, so we can't just return -1 in misdn_write in this
+ case. Added a MISDN_KEYPAD channel variable, and fixed the
+ sending of keypad. this enables us to modify the call forward
+ parameters in the switch.
+
+ * channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: fixed
+ the problem that misdn_write did not return -1 when called with 0
+ samples in a frame this resultet in a deadlock in some
+ circumstances, when the call ended because of a busy extension.
+ added encoding of keypad.
+
+2007-05-03 00:23 +0000 [r62796-62841] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_config_odbc.c: doh... initializing the pointer variable
+ will work just a bit better
+
+ * res/res_config_odbc.c: increase reliability and efficiency of
+ static Realtime config loading via ODBC: don't request fields we
+ aren't going to use don't request sorting on fields that are
+ pointless to sort on explicitly request the fields we want,
+ because we can't expect the database to always return them in the
+ order they were created (reported by blitzrage in person (!),
+ patch by me)
+
+2007-05-02 20:10 +0000 [r62737] Steve Murphy <murf@digium.com>
+
+ * cdr.c, pbx.c: Some tweaks to satisfy CDR bug 8796, where being in
+ 'h' extension louses up the dst field
+
+2007-05-02 17:38 +0000 [r62691] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_iax2.c: Issue 9638 - if a text frame is sent with
+ no terminating NULL through a bridged IAX connection, the remote
+ end will receive garbage characters tacked onto the end.
+
+2007-05-01 21:55 +0000 [r62496-62547] Russell Bryant <russell@digium.com>
+
+ * res/res_features.c: Remove an unnecessary check that makes it so
+ if you hang up after doing an attended transfer before the target
+ extension answers the channel, the transfer is not successful.
+ (issue #9338, patch by svanlund)
+
+ * configs/indications.conf.sample: Add indications.conf information
+ for the Philippines. (issue #9525, reported and patched by
+ loloski)
+
+2007-04-30 15:57 +0000 [r62417] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: This patch fixes an issue where depending on
+ the cause code, when the network sends a PRI disconnect, the call
+ may not be properly hung up. (issue #9588, reported and patched
+ by softins)
+
+2007-04-30 14:34 +0000 [r62368] Joshua Colp <jcolp@digium.com>
+
+ * asterisk.c: Update copyright notice. It's now the year 2007!
+
+2007-04-27 16:16 +0000 [r62173] Jason Parker <jparker@digium.com>
+
+ * codecs/codec_zap.c: This transcoder message needn't be a NOTICE.
+ I've seen it cause confusion more than a few times.
+
+2007-04-27 13:57 +0000 [r62126] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #7351 - SIP Cancel fails due to the
+ wrong contact uri. Reported by PPYY, failed to fix by OEJ final
+ fix by wojtekka - THANKS!!!! THis was a hard one to catch.
+
+2007-04-26 16:30 +0000 [r61958-62037] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Revert previous fix for when the IAX2
+ channel goes funky (that's the technical term). This is causing
+ legit calls to be prematurely hung up. (issue #9600 reported by
+ justdave)
+
+ * config.c: Don't count failed include attempts against the
+ configuration include level. (issue #9593 reported by mostyn)
+
+2007-04-25 22:24 +0000 [r61913] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: handle a very bizarre race condition with
+ channels being redirected before a simple switch can be started
+ on them (issue #9286)
+
+2007-04-25 21:55 +0000 [r61862-61866] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: If the callerid= option is specified, but
+ empty, clear any previous data.
+
+ * channels/chan_iax2.c: Ensure that callerid settings are reset on
+ a reload.
+
+2007-04-25 18:52 +0000 [r61804] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/app.h, funcs/func_groupcount.c, app.c,
+ apps/app_groupcount.c, channel.c: Merge rewritten group counting
+ support. No more storing data on the variable list of the
+ channels. That was bad, mmmk? (issue #7497 reported by sabbathbh)
+
+2007-04-25 16:20 +0000 [r61798] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: Fix a typo where cid_num got copied instead
+ of cid_ani. (issue #9587, reported and patched by xrg)
+
+2007-04-24 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.18 released
+
+2007-04-24 21:33 +0000 [r61786] Russell Bryant <russell@digium.com>
+
+ * manager.c: Don't crash if a manager connection provides a
+ username that exists in manager.conf but does not have a
+ password, and also requests MD5 authentication. (ASA-2007-012)
+
+2007-04-24 18:20 +0000 [r61776-61777] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * channels/chan_phone.c, channels/chan_zap.c,
+ channels/chan_modem.c: removed #if 0 block from chan_phone,
+ chan_zap, and chan_modem restart_monitor()
+
+ * channels/chan_modem.c: removed pthread_join in restart_monitor()
+ to make it like chan_zap and other channel drivers
+
+2007-04-24 16:05 +0000 [r61771] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Allow RFC2833 to be sent in the response SDP
+ when an INVITE comes in without SDP. (issue #9546 reported by
+ mcrawford)
+
+2007-04-24 13:50 +0000 [r61770] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
+ channels/misdn/isdn_lib.c: added lock for sending messages to
+ avoid double sending. shuffled some empty_chans after the
+ cb_event calls, this avoids that a release_complete from a quite
+ different call releases a fresh created setup by accident.
+
+2007-04-20 21:14 +0000 [r61692-61704] Jason Parker <jparker@digium.com>
+
+ * loader.c: Fix an issue that I noticed while looking over issue
+ 9571. The reload timestamp was getting set after reloading the
+ built-in stuff, and before the modules.
+
+ * apps/app_queue.c: If the '* to hangup' option is not enabled, we
+ don't need to disable * as a valid exit key. If it was enabled,
+ this statement would've never been checked in the first place.
+ Issue #9552
+
+2007-04-19 07:56 +0000 [r61685] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Send NOTIFY to Contact: in SUBSCRIBE - as
+ reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp.
+
+2007-04-19 02:30 +0000 [r61680] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * funcs/func_language.c, funcs/func_moh.c, funcs/func_groupcount.c,
+ funcs/func_timeout.c, funcs/func_cdr.c, funcs/func_callerid.c:
+ Bug 9557 - Specifying the GetVar AMI action without a Channel
+ parameter can cause Asterisk to crash. The reason this needs to
+ be fixed in the functions instead of in AMI is because Channel
+ can legitimately be NULL, such as when retrieving global
+ variables.
+
+2007-04-16 14:08 +0000 [r61663] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't stop RTP on errors on INFO messages.
+ Disclaimer: This patch was needed for Edvina AstHoloApp and was
+ meant to be included in 1.2, but never made it in time so I felt
+ I could add it now. No, just joking, patching error found while
+ testing T.140 with Omnitor earlier this spring.
+
+2007-04-13 19:15 +0000 [r61655] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Add OUTBOUND_GROUP_ONCE variable to app_dial.
+ This behaves the same as OUTBOUND_GROUP except it will get unset
+ after use so it won't get accidentally inherited. (issue #BE-140)
+
+2007-04-11 16:01 +0000 [r61376-61476] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: If someone sets the "useragent" option in
+ sip.conf to be empty, then don't add the User-Agent header at
+ all. It is an optional header, anyway. Also, the bug report says
+ that some of Japan's SIP providers don't allow it for some weird
+ reason. (issue #9488, reported by makoto, fixed by me)
+
+ * channels/chan_sip.c: Fix a bug with switching between
+ host=dynamic and using specific hosts for peers. The code would
+ only reset the peer's address when it is dynamic if it was a new
+ peer structure. Now, it will also reset the address if it was
+ already in the peer list, but before the reload, it was not
+ dynamic. (issue #9515, reported by caio1982, fixed by me)
+
+ * channels/chan_sip.c: Remove the attempt at reporting
+ configuration errors in sip.conf. This can cause a bunch of
+ improper messages when using realtime. I give up. As oej tried to
+ convince me when I put this in, there is just no easy way to do
+ it. (inspired by a message on the -dev list)
+
+2007-04-11 12:05 +0000 [r61357] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: some fixes for
+ PMP Hold/Retrieve, it should work now, when briding=no
+
+2007-04-10 12:31 +0000 [r61170] Nadi Sarrar <ns@beronet.com>
+
+ * channels/misdn_config.c: msns config parameter defaults to '*'
+
+2007-04-09 19:38 +0000 [r61038] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: - Don't send ActionID before Response:
+ header. - Don't use a blank in an AMI header
+
+2007-04-09 17:22 +0000 [r60935] Jason Parker <jparker@digium.com>
+
+ * apps/app_directory.c: Allow matching on names shorter than 3
+ chars. This also fixes the case where somebody wants to match on
+ less then 3 chars. Issue 9071
+
+2007-04-09 02:49 +0000 [r60846-60849] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk.h, asterisk.c: Don't check for error when
+ lowering priority (according to the manpage, it should never
+ happen anyway). It might could happen, though, if another thread
+ messed with the priority, so safeguard against that (reported via
+ -dev list).
+
+ * channels/chan_local.c: Bug 9505 - If the return value for
+ local_queue_frame is set, then p->lock is no longer valid.
+
+2007-04-09 00:59 +0000 [r60797] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: When calling a device that then forwards us
+ elsewhere... we have to make our channels compatible if it is the
+ only channel being dialed. (issue #9445 reported by
+ marcelbarbulescu)
+
+2007-04-08 14:00 +0000 [r60660-60711] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_macro.c: Gosub called within a Macro resets the
+ arguments improperly and causes general weirdness. (Issue 8329)
+
+ * file.c: Bug 9486 - memory leak when opening a filestream
+
+2007-04-06 17:03 +0000 [r60456] Matt Frederickson <creslin@digium.com>
+
+ * channels/chan_zap.c: There should only be one code path for doing
+ DTMF conditionals on channels. This fixes it.
+
+2007-04-06 14:41 +0000 [r60398] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/codec_zap.c: remove undocumented 'cardsmode' parameter and
+ stop searching for transcoders during reload()
+
+2007-04-05 16:09 +0000 [r60267] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Just because we can't find the voicemail
+ configuration file, doesn't mean that the module failed to load.
+ The user could be using realtime. Issue #9473
+
+2007-04-05 12:52 +0000 [r60213] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only unlock our pvt and net locks if we are
+ actually going to try to lock the owner again. (issue #9472
+ reported by zoa)
+
+2007-04-04 17:38 +0000 [r59886-60134] Russell Bryant <russell@digium.com>
+
+ * manager.c: It is valid to redirect channels via the manager
+ interface that are not in the UP state. Instead of checking for
+ that to prevent to ensure a dead channel doesn't get redirected,
+ just use the ast_check_hangup() API call. (issue #9457, reported
+ by Callmewind, patch by me) (related to issue #8977)
+
+ * channels/chan_sip.c: Fix the return value of
+ handle_common_options() so that it always properly indicates
+ whether it handled the option or not. (issue #9455, reported by
+ Netview, fixed by me)
+
+ * channels/chan_sip.c: Add a missing "\r\n" in the body of the
+ NOTIFY that is sent to indicate the status of a transfer. (issue
+ #9388, reported by rarritt)
+
+ * mkpkgconfig: Use the more generic check for "sed -r" support that
+ was already present in 1.4. (related to issue #9399)
+
+ * mkpkgconfig: On Darwin, the -r argument to sed is not valid. It
+ has to be -E. (issue #9399, reported by jcovert)
+
+ * channels/chan_sip.c: Don't attempt to report configuration errors
+ in build_user(). oej pointed out that for a "friend" entry, this
+ won't work, because all user options are valid for peers, but not
+ the other way around.
+
+ * channels/chan_sip.c: Make chan_sip report when it encounters an
+ unknown option. (issue #9440, reported by nightcrawler)
+
+ * app.c: When doing a built-in blind or attended transfer, restore
+ the ability to use '#' to terminate the number and immediately do
+ the transfer instead of having to dial the number and just wait
+ for the feature digit timeout. (issue #8366, xueliangliang)
+
+2007-04-03 10:40 +0000 [r59788-59803] Nadi Sarrar <ns@beronet.com>
+
+ * channels/misdn_config.c: ptp is the 5th bit, not the 4th.
+
+ * channels/misdn_config.c, channels/misdn/chan_misdn_config.h: Use
+ the new sysfs way of mISDN 1.2 to check if a port is NT or not.
+
+2007-04-02 18:55 +0000 [r59723] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Increase the maximum size for a string of
+ mailboxes to 1024. (issue #9270 reported by rtucker)
+
+2007-04-02 12:08 +0000 [r59623-59639] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h: added option which allows us
+ to accept incoming SETUP Messages without automatically sending
+ Proceeding or Setup Acknowledge, this is useful with some broken
+ switches and if you want to Release incoming calls without
+ previously having acknowledged them. The new option is
+ noautorespond_on_setup=yes|no default is no, so we don't break
+ the existing behaviour
+
+ * channels/chan_misdn.c: don't be verbose if no need
+
+ * channels/misdn/isdn_lib.c: we can now make 30 channels on a PRI
+ (before we forgot chan 31..)
+
+2007-04-01 22:35 +0000 [r59608] Russell Bryant <russell@digium.com>
+
+ * netsock.c: Add the SO_REUSEADDR flag to sockets handled by
+ netsock. This is needed by the patch that went in for issue 7874.
+ chan_iax2 needs to be able to create socket that is lisetning on
+ INADDR_ANY, but also be able to bind sockets to specific
+ addresses. (Thanks to Stevenson on the asterisk-dev mailing list
+ for explaining why this flag was needed.)
+
+2007-03-29 17:33 +0000 [r59360] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Keep a global array of variables indicating
+ whether certain conference rooms are in use. This ensures that
+ two people going into a new dynamic conference when the 'e'
+ option is set don't go into the same conference room. (issue
+ #8835 reported by eliel)
+
+2007-03-29 17:14 +0000 [r59355-59357] Russell Bryant <russell@digium.com>
+
+ * rtp.c: If an error occurs when reading from an RTP socket, and
+ the error code does not indicate that we should try again, then
+ return NULL instead of a "null frame". This will prevent Asterisk
+ from trying over and over again, and eventually causing the
+ system to crash. (issue #8285, john)
+
+ * channels/chan_iax2.c: Backport the change to chan_iax2 to return
+ NULL instead of a "null frame" from its read callback. See
+ revision 59341 to the 1.4 branch for more info.
+
+2007-03-29 16:04 +0000 [r59299-59301] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * cdr/cdr_odbc.c: Issue 9415 - No point to getting a diagnostic
+ field if we aren't doing anything with the information. (Plus, it
+ tends to crash the Postgres ODBC driver.)
+
+ * res/res_odbc.c: Change ENV section to use setenv, instead of
+ putenv (Alexandru Pirvulescu <sigxcpu@gmail.com>, reported via
+ -dev list)
+
+2007-03-27 23:36 +0000 [r59280-59283] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Oops
+
+ * apps/app_voicemail.c: Fix a few remaining bad mmap(2) return
+ values
+
+2007-03-27 23:19 +0000 [r59258-59277] Russell Bryant <russell@digium.com>
+
+ * apps/app_directory.c: Fix the check of the return value from
+ mmap(). Thanks to Corydon for catching this one.
+
+ * channels/chan_iax2.c: Fix the use of the "sourceaddress" option
+ when "bindaddr" is set to 0.0.0.0 instead of having each
+ interface explicitly listed. (issue #7874, patch by stevens)
+
+2007-03-27 13:56 +0000 [r59252] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed #9355
+
+2007-03-26 10:21 +0000 [r59199] Nadi Sarrar <ns@beronet.com>
+
+ * channels/misdn_config.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, channels/misdn/Makefile,
+ channels/misdn/chan_misdn_config.h, channels/Makefile,
+ channels/misdn/isdn_lib.c: mISDN >= 1.2 provides a dsp pipeline
+ for i.e. echo cancellation modules, make chan_misdn use it.
+
+2007-03-24 01:35 +0000 [r59194] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only try to handle a response if it has a
+ response code. (ASA-2007-011)
+
+2007-03-23 16:07 +0000 [r59186-59187] Steve Murphy <murf@digium.com>
+
+ * apps/app_macro.c: Ugh. that was dumb. Fixed an error.
+
+ * apps/app_macro.c: Added a few words in the Macro doc strings
+ about the behavior of macros with hangups (et al.), as per 9337
+
+2007-03-21 18:03 +0000 [r59086] Joshua Colp <jcolp@digium.com>
+
+ * res/res_monitor.c: Indicate the filename changed when it is
+ changed. (issue #9311 reported by jsmith)
+
+2007-03-20 09:23 +0000 [r59062-59063] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: modified a loglevel
+
+ * channels/chan_misdn.c: avoid sending a disconnect when we already
+ received one.
+
+2007-03-19 Jason Parker <jparker@digium.com>
+
+ * Asterisk 1.2.17 released
+
+2007-03-14 16:38 +0000 [r58896] Russell Bryant <russell@digium.com>
+
+ * SECURITY: Add a note to the security file that the Asterisk CLI
+ and log files may contain sensitive information, and that people
+ should keep this in mind.
+
+2007-03-13 12:58 +0000 [r58849-58850] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed the crypt_keys stuff
+
+ * channels/misdn_config.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
+ channels/misdn/isdn_lib.c: added method standard_dec for dialing
+ out on groups, to avoid conflicts, which caused issues with some
+ ISDN providers
+
+2007-03-13 11:45 +0000 [r58847] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #9229 - No port in request URI on
+ register to non default SIP ports (neelakantan)
+
+2007-03-12 16:49 +0000 [r58832] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/lock.h: We can't use the assembler version of
+ fetchadd_int under Intel Macs. (issue #9254 reported by darrell
+ budic)
+
+2007-03-09 20:46 +0000 [r58579] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: If we are unable to lookup the host in a c
+ line we have to abort, otherwise the previous data is gone and we
+ will (potentially) have no data when all is said and done.
+
+2007-03-09 14:43 +0000 [r58558] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: we can free channel 31 as well, since
+ we can occupy it
+
+2007-03-08 16:04 +0000 [r58388] Joshua Colp <jcolp@digium.com>
+
+ * dsp.c: Only print out debug message if the definition that makes
+ the variables shows up was actually defined. (issue #9233
+ reported by serginuez)
+
+2007-03-07 18:17 +0000 [r58164-58242] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a problem where the Asterisk channel
+ name could be that of the wrong IAX2 user for a call. This is
+ because the first step of choosing this name is to look for an
+ IAX2 peer that happens to have the same IP/port number that this
+ call is coming from and assuming that is it. However, this is not
+ always correct. So, I have made it change this name after
+ authentication happens since at that point, we have an exact
+ match.
+
+ * manager.c: Fix a misplaced block of code in the 1.2 version of
+ the patch to fix issue #8977
+
+ * manager.c: If the channels acquired using the manager Redirect
+ action are not up, then don't attempt to do anything with them.
+ It could lead to weird behavior, including crashes. (issue #8977)
+
+2007-03-06 22:52 +0000 [r58115] Steve Murphy <murf@digium.com>
+
+ * channels/chan_sip.c: Fix for 9220: Eyebeam cannot renew
+ subscriptions for presence info. Reason: re-SUBSCRIBE requests
+ don't include Accept headers, which the rfc says are optional (to
+ put it tersely), (it uses MAY), and luckily, the sip_pvt struct
+ has the format info stored, so we simply leave it if the format
+ is set, and the accept header null.
+
+2007-03-06 20:33 +0000 [r58052] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Change error message to proper message
+
+2007-03-06 15:17 +0000 [r58008] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/h323/Makefile: Cleanup the Makefile so that we only
+ attempt to include a file when we're building and not 'clean'ing
+ so 'make clean' completes successfully. chan_h323 maintainer:
+ please check to make sure I haven't broken your build target.
+ From: jsmith in #asterisk-dev
+
+2007-03-05 23:18 +0000 [r57962] Christian Richter <christian.richter@beronet.com>
+
+ * channels/Makefile: subdirs like misdn and h323 should be cleaned
+ by the clean target from channels/Makefile as well
+
+2007-03-05 17:49 +0000 [r57869] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Make create_dirpath use our standard for
+ return values. -1 is failure, 0 is success. (issue #9205 reported
+ by ballares)
+
+2007-03-05 14:53 +0000 [r57825] Steve Murphy <murf@digium.com>
+
+ * pbx.c: Fixed a typo introduced via 9156
+
+2007-03-04 10:39 +0000 [r57753] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed another
+ place where the out_cause was hardcoded to 16
+
+2007-03-03 06:36 +0000 [r57648] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Memory leak of a list, if call recording
+ was abandoned
+
+2007-03-02 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.16 released
+
+2007-03-02 18:32 +0000 [r57523] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: fixed typo
+
+2007-03-02 17:02 +0000 [r57475] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: If a SIP message comes in and goes to a
+ method handler that requires additional values that may not be
+ present then send back an error.
+
+2007-03-02 16:39 +0000 [r57458] Steve Murphy <murf@digium.com>
+
+ * pbx.c: further refinement in wording of goto documentation, as
+ per 9156, goto not proceeding to next instruction
+
+2007-03-01 22:19 +0000 [r57317] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c: Don't even attempt to optimize things when
+ a proxy channel is involved. It will just explode in weird and
+ unexplaineable ways. (issue #9175 reported by
+ clegall_proformatique)
+
+2007-02-28 19:12 +0000 [r57118] Steve Murphy <murf@digium.com>
+
+ * pbx.c: a small documentation update, to reflect reality in the
+ goto doc strings, as per 9156, Goto does not proceed to next prio
+ if jump fails
+
+2007-02-28 18:55 +0000 [r57092] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_agent.c: Fix a few more issues with the agent
+ logoff CLI command. (issue #9123 reported by arbrandes)
+
+2007-02-28 16:09 +0000 [r57034] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed
+ bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302,
+ #303, #304
+
+2007-02-26 20:05 +0000 [r56850] Joshua Colp <jcolp@digium.com>
+
+ * pbx/pbx_config.c: Obey the clearglobalvars option in extensions
+ reload (or dialplan reload depending on your version). (issue
+ #9146 reported by ramonpeek)
+
+2007-02-26 00:34 +0000 [r56729] Russell Bryant <russell@digium.com>
+
+ * utils.c: Ensure that lock.h is included in utils.c with
+ AST_API_MODULE defined so that the implementations will be
+ properly included when the AST_INLINE_API functions are not going
+ to be inlined. (issue #9124, festr)
+
+2007-02-25 14:38 +0000 [r56684] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channel.c: Issue 9130 - If prev is the last item on the channel
+ list, then evaluating additional conditions (e.g. name prefix)
+ will cause a NULL dereference.
+
+2007-02-23 23:20 +0000 [r56406-56504] Russell Bryant <russell@digium.com>
+
+ * asterisk.c: Fix up a couple more signal handlers to not do bad
+ things that could cause various undesirable results. The other
+ day, I made Asterisk deadlock by hitting Control-C because of a
+ bad signal handler. Now, signal handlers just set a flag and
+ write to an alert pipe for the flag to be handled. Then, there is
+ another thread that is monitoring for these flags. If being run
+ in console mode, it is just the main thread. If Asterisk is in
+ the background, a thread is created to do it.
+
+ * channels/chan_iax2.c: Don't destroy mutexes before unregistering
+ all of the entry points from the core. Also, fix a potential
+ memory leak from not destroying the locks for all of the possible
+ call numbers (about 32k of them).
+
+2007-02-22 23:19 +0000 [r56230-56279] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_agent.c: Always defer Agent logoff if any channels
+ are up until they hang up. (issue #9123 reported by arbrandes)
+
+ * channels/chan_sip.c, channel.c: Only change the original or clone
+ channel if it's the channel behind the proxy channel, not if it's
+ just a regular bridged channel.
+
+2007-02-22 00:53 +0000 [r56010] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: If we receive a frame that is not in any of
+ the negotiated formats, then drop it. (potentially issue #8781
+ and SPD-12)
+
+2007-02-21 20:32 +0000 [r55956] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Change naughty warning message to provide
+ useful information. If a write now fails on a channel in meetme
+ it will tell you the channel name instead of spitting out the
+ wrong error message.
+
+2007-02-21 14:03 +0000 [r55868] Kevin P. Fleming <kpfleming@digium.com>
+
+ * build_tools/make_svn_branch_name: use new tag version script
+
+2007-02-21 00:19 +0000 [r55750] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c, utils.c, include/asterisk/lock.h: Fix random
+ crashes when using the MeetMe application. This patch converts
+ list handling to use the linked list macros and most importantly,
+ implements reference counting on the ast_conference objects. The
+ reference counting was first backported from 1.4. However, that
+ code has some problems that caused the reference count to never
+ hit zero. Those problems are fixed in this patch and will be
+ resolved in 1.4 and trunk next, with a different patch. (issues
+ #7647, #9073, #9106, BE-115).
+
+2007-02-20 22:39 +0000 [r55669] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_agent.c: Defer clearing callback information if
+ channels are up until they are hung up. This ensures the hangup
+ process goes smoothly and no channels get hung in limbo. (issue
+ #8088 reported by kebl0155)
+
+2007-02-20 19:49 +0000 [r55588] Russell Bryant <russell@digium.com>
+
+ * apps/app_page.c: Convert a tab to spaces so that the
+ documentation is printed out properly aligned.
+
+2007-02-19 18:09 +0000 [r55434] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: forcename and forcegreetings options should
+ check to see if the recording already exists
+
+2007-02-18 12:32 +0000 [r55249-55277] Olle Johansson <oej@edvina.net>
+
+ * apps/app_record.c: Documentation update (#9053, jsmith)
+
+ * channels/chan_sip.c: Issue #9020 - SIP message retransmission
+ time too short. Backporting fix implemented in 1.4, where we have
+ a minimum level for the T1 timer.
+
+2007-02-17 03:53 +0000 [r55073-55153] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Answer the channel before recording privacy
+ information. (issue #8926 reported by lmamane)
+
+ * channels/chan_sip.c: Allow chan_sip to handle attended transfers
+ from a SIP phone that is sitting behind chan_agent. Yes folks,
+ all it took was one line of code. (issue #8784 reported by
+ pzieba)
+
+2007-02-16 22:48 +0000 [r55005] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c, configs/meetme.conf.sample: Revert the change
+ I did in revisions 54955, 54969, and 54970, in 1.2, 1.4, and
+ trunk. I decided that once a conference is created from
+ meetme.conf, it is acceptable behavior that the pin can not be
+ changed until the conference goes away. I also added a note in
+ meetme.conf to describe this behavior. We still have another
+ issue in 1.4 and trunk where some conferences with no users don't
+ go away. That is the real bug that needs to be addressed here.
+
+2007-02-16 22:13 +0000 [r54999] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_agent.c: Do not send indications through
+ ast_indicate in chan_agent but instead go directly to the
+ technology. This way when indications are emulated they happen on
+ the Agent channel and do not screw up formats on the channels.
+ (issue #8439 reported by punkgode)
+
+2007-02-16 20:56 +0000 [r54955] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: For conferences that are configured in
+ meetme.conf, check the configuration file every time someone
+ joins the conference instead of only when the conference is first
+ created. This is to ensure that changes to the pin numbers in the
+ config file are always honored. (issue #9073)
+
+2007-02-16 11:38 +0000 [r54771] Olle Johansson <oej@edvina.net>
+
+ * res/res_agi.c: Issue #9069 - If we open with TH we should not
+ close with /TD. (seanbright)
+
+2007-02-15 16:14 +0000 [r54622] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Use a separate variable to indicate execution
+ should continue instead of the return value. (issue #8842
+ reported by pluto70)
+
+2007-02-14 18:40 +0000 [r54438] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c:
+
+2007-02-14 16:25 +0000 [r54373] Matt Frederickson <creslin@digium.com>
+
+ * channels/chan_zap.c: When handling glare on a PRI, move the
+ requested channel rather than hang up the old one. Fix for 8957
+ and 9011.
+
+2007-02-13 18:35 +0000 [r54179] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Make sure that outbound calls are applied to
+ the peer. This fixes some issues with "hints not working", but
+ only in 1.2.
+
+2007-02-08 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.15 released
+
+2007-02-08 22:17 +0000 [r53658] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/codec_zap.c: ensure channelcount is cleared before we
+ enumerate transcoders, so 'reload' doesn't double the channel
+ count
+
+2007-02-08 13:36 +0000 [r53529] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Issue 9003 - If fullname is empty, quote()
+ passes back "\""
+
+2007-02-07 15:38 +0000 [r53357] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Fix a few potential memory leaks with
+ realtime users and peers. (issue #8999 reported by bsmithurst)
+
+2007-02-07 15:30 +0000 [r53354] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_macro.c: Issue 7440 - Macro called from Macro from the h
+ extension exits prematurely
+
+2007-02-06 06:58 +0000 [r53245] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * manager.c: Issue 8987 - Status could return two responses
+ (mnicholson)
+
+2007-02-06 00:08 +0000 [r53224] Jason Parker <jparker@digium.com>
+
+ * configs/dnsmgr.conf.sample: Add a proper newline at the end of
+ this sample config file.
+
+2007-02-03 20:39 +0000 [r53133-53134] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c, utils.c, include/asterisk/lock.h: Revert some
+ changes that accidentally got committed as a part of another fix.
+
+ * apps/app_dial.c, apps/app_meetme.c, utils.c,
+ include/asterisk/lock.h: set the DIALSTATUS variable to contain
+ "INVALIDARGS" when the dial application exits early because of
+ invalid arguments instead of just leaving it empty. (issue #8975)
+
+2007-02-02 16:58 +0000 [r53117] Joshua Colp <jcolp@digium.com>
+
+ * config.c: Pass the glob expanded filename to process_text_line so
+ that error messages contain the actual filename, not the original
+ include one. (issue #8959 reported by tzafrir)
+
+2007-02-01 23:14 +0000 [r53107] Jason Parker <jparker@digium.com>
+
+ * apps/app_chanspy.c: Fix a small typo. Synopsis lines shouldn't
+ have a newline
+
+2007-02-01 22:21 +0000 [r53095-53103] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Copy noncodeccapability over to the joint
+ variable so that telephone-event will get transmitted in the sent
+ INVITE.
+
+ * channels/chan_sip.c: Don't negotiate RFC2833 when not configured
+ to do so. (issue #8799 reported by mdu113)
+
+2007-02-01 21:12 +0000 [r53090] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: - Make sure we release call from call
+ counter before we destroy call (maybe #7744 and more) -
+ Backported by accident from 1.4
+
+2007-02-01 21:03 +0000 [r53084] Joshua Colp <jcolp@digium.com>
+
+ * res/res_musiconhold.c: Return previous behavior of having MOH
+ pick up where it was left off. (issue #8672 reported by
+ sinistermidget)
+
+2007-02-01 20:07 +0000 [r53069-53074] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * funcs/func_strings.c: Bug 8965
+
+ * funcs/func_strings.c, pbx.c: No wonder FIELDQTY doesn't work with
+ functions... the documentation in pbx.c was wrong
+
+2007-01-31 21:25 +0000 [r53045] Russell Bryant <russell@digium.com>
+
+ * channels/chan_mgcp.c, manager.c, channels/chan_zap.c,
+ pbx/pbx_spool.c, apps/app_meetme.c, apps/app_page.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_queue.c,
+ channels/chan_iax2.c, apps/app_rpt.c, cdr.c, pbx.c: Fix a bunch
+ of places where pthread_attr_init() was called, but
+ pthread_attr_destroy() was not.
+
+2007-01-31 18:58 +0000 [r53044] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/codec_zap.c: update to match modified transcoder API
+
+2007-01-31 17:41 +0000 [r53039] Russell Bryant <russell@digium.com>
+
+ * rtp.c: Use the proper format string to print unsigned values in
+ the rtp debug output. (issue #8954, wmis)
+
+2007-01-31 17:28 +0000 [r53034] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/Makefile: allow codec_zap to build again, now that
+ transcoder support is in zaptel 1.2
+
+2007-01-30 19:41 +0000 [r52857-52954] Russell Bryant <russell@digium.com>
+
+ * channel.c: Don't print a message indicating that we don't know
+ what to do with a proceeding control frame in
+ ast_request_and_dial(). We just need to ignore it. (reported by
+ JerJer on #asterisk-dev)
+
+ * asterisk.c: The SIGHUP handler was implemented to allow admins to
+ send SIGHUP to a running Asterisk process to reload the
+ configuration. However, doing the actual reload in the signal
+ handler itself is a very bad thing to do, because the reload
+ process includes calling non-reentrant functions such as
+ malloc/calloc/etc. If Asterisk is running in the background, then
+ the reload will happen immediately. However, if running in
+ console mode, the reload doesn't work until something is typed at
+ the console. That sort of defeats the purpose, but I don't see an
+ easy way to get around it at this point.
+
+ * codecs/Makefile: Comment out the parts in the Makefile that make
+ codec_zap get built. It will not yet build against zaptel 1.2, so
+ I am disabling it to prevent further bug reports until it gets
+ merged. (issue #8940)
+
+2007-01-30 14:38 +0000 [r52843] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed some
+ possible segfaults. also fixed an very important bug which occurs
+ on high load (when calls are very fast generated)
+
+2007-01-30 00:15 +0000 [r52762] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix the extraction of the timestamp from
+ video frames. It was using the mapping for a mini-frame instead
+ of a video-frame, which caused it to get invalid data. (issue
+ #8795, mihai)
+
+2007-01-29 23:39 +0000 [r52716] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_mixmonitor.c: Now that filename is part of the structure
+ and since it comes before postprocess... we have to add it to our
+ postprocess line. (reported on asterisk-dev by Boris Bakchiev)
+
+2007-01-29 16:48 +0000 [r52503] Jason Parker <jparker@digium.com>
+
+ * codecs/codec_zap.c: Use the correct zaptel header file location.
+ Currently, this will not build - transcoder support will be added
+ to zaptel later today.
+
+2007-01-27 02:09 +0000 [r52360-52415] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_queue.c: Make COMPLETECALLER and COMPLETEAGENT output to
+ queue_log follow documentation. (issue #7677 reported by amilcar)
+
+ * channels/chan_iax2.c: Make the last context entry read in the
+ dominant one. (issue #8918 reported by pj)
+
+2007-01-25 19:15 +0000 [r52162-52264] Joshua Colp <jcolp@digium.com>
+
+ * jitterbuf.c: Allow dequeueing of frames with negative timestamp
+ by moving jitterbuffer frames check to jb_next. (issue #8546
+ reported by harmen)
+
+ * apps/app_mixmonitor.c: Add another note about audio files being
+ played back to each bridged party. (issue #8718 reported by ppyy)
+
+2007-01-25 00:39 +0000 [r52137] Russell Bryant <russell@digium.com>
+
+ * apps/app_groupcount.c: Fix a seg fault when running this
+ application with no arguments from AGI. (issue #8905, junky)
+
+2007-01-24 17:43 +0000 [r52002] Steve Murphy <murf@digium.com>
+
+ * utils/check_expr.c, utils/Makefile: updated check_expr via 8322
+ (refactoring of expression checking impl); elfring contributed a
+ nice code reorg, I contributed some time to get it working again,
+ better messages
+
+2007-01-24 10:48 +0000 [r51966] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed the busy problem (dialstatus was not
+ busy when we called a busy extension)
+
+2007-01-24 00:57 +0000 [r51828-51843] Russell Bryant <russell@digium.com>
+
+ * channel.c: Fix an issue related to synchronization of recordings
+ when using Monitor(). The bug is a miscalculation of the amount
+ to seek the stream for writing to disk when the number of samples
+ coming in and out of a channel do not match up. (issue #8298,
+ #8887, report and patch by guillecabeza, patch files created and
+ testing done by whoiswes)
+
+ * apps/app_while.c: Don't set a new value for the END_ variable on
+ the channel before using the old value. If you do, it will lead
+ to accessing a memory address that has been free()'d. (issue
+ #8895, arkadia)
+
+2007-01-23 01:41 +0000 [r51512] Joshua Colp <jcolp@digium.com>
+
+ * res/res_musiconhold.c: Yield before reading from zaptel timing
+ source under Solaris so that other threads get a chance to do
+ things. (issue #7875 reported by bob)
+
+2007-01-22 19:39 +0000 [r51410] Russell Bryant <russell@digium.com>
+
+ * codecs/Makefile, codecs/codec_zap.c (added): Merge codec_zap
+ support for the transcoder card. This is a standalone codec
+ module so it will not affect anything else.
+
+2007-01-22 19:08 +0000 [r51359-51406] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_mixmonitor.c: Move filestream creation to Mixmonitor
+ loop. This will prevent a blank file from being created if no
+ frames ever pass through to be recorded. (issue #7589 reported by
+ steve_mcneil)
+
+ * channels/chan_h323.c: Explicitly declare what codecs are
+ supported by default globally since using a bitmask for all may
+ include ones we don't need. (issue #8357 reported by
+ gknispel_proformatique)
+
+2007-01-19 16:44 +0000 [r51300] Russell Bryant <russell@digium.com>
+
+ * asterisk.c: Fix a memory leak on command line tab completion. The
+ container for the matches was freed, but the individual matches
+ themselves were not. (issue #8851, arkadia)
+
+2007-01-18 23:47 +0000 [r51271] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * channels/chan_zap.c: issue 7877: chan_zap module reload does not
+ use default/initialized values on subsequent loads. Reset
+ configuration variables to default values prior to parsing
+ configuration file.
+
+2007-01-18 23:35 +0000 [r51269] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: support echo cancellers that can handle 64ms
+ or 128ms of echo cancellation
+
+2007-01-18 21:11 +0000 [r51235-51255] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * stdtime/localtime.c: If a timezone is not specified, assume
+ localtime (instead of gmtime) (Issue #7748)
+
+ * contrib/scripts/vmdb.sql: Document all the fields, including the
+ indication that "uniqueid" should not be renamed.
+
+2007-01-17 21:17 +0000 [r51197] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Move the check for a failure of
+ ast_channel_alloc() to before locking the pvt structure again.
+ Otherwise, on a failure, this will cause a deadlock.
+
+2007-01-17 20:52 +0000 [r51158-51194] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * utils.c: When ast_strip_quoted was called with a zero-length
+ string, it would treat a NULL as if it were the quoting character
+ (and would thus return the string in memory immediately following
+ the passed-in string).
+
+ * doc/voicemail_odbc_postgresql.txt (added): Add documentation
+ walkthrough on getting Postgres to work with voicemail (from
+ Issue 8513)
+
+ * apps/app_voicemail.c: Postgres driver doesn't like a NULL pointer
+ when retrieving the length (Bug 8513)
+
+2007-01-16 17:36 +0000 [r51085-51145] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Return previous behavior. ParkedCalls will be
+ able to do DTMF based transfers again. trunk however will get an
+ option to allow this to be set on/off. (issue #8804 reported by
+ nortex)
+
+ * channels/chan_zap.c: Add none as a valid callgroup/pickupgroup
+ option. I consider it a bug that it would inherit it all the way
+ down and not have any way to reset it to nothing - so that's why
+ it is in 1.2. (issue #8296 reported by gkloepfer)
+
+2007-01-15 23:09 +0000 [r50987] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_groupcount.c: Check return value before dereferencing
+ (Bug 8822)
+
+2007-01-15 20:44 +0000 [r50946] Matt O'Gorman <mogorman@digium.com>
+
+ * apps/app_voicemail.c: Solves issue with forwarding voicemails
+ from folders other than inbox. patch by anthonyl.
+
+2007-01-14 05:01 +0000 [r50781] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * db1-ast/recno/rec_utils.c, db1-ast/hash/hash_bigkey.c,
+ db1-ast/hash/hsearch.c, db1-ast/recno/rec_open.c,
+ db1-ast/recno/rec_delete.c, db1-ast/btree/bt_page.c,
+ db1-ast/hash/hash_buf.c, db1-ast/hash/hash_page.c,
+ db1-ast/recno/rec_close.c, db1-ast/recno/rec_search.c,
+ db1-ast/btree/bt_get.c, db1-ast/hash/hash.c,
+ db1-ast/recno/rec_put.c, db1-ast/include/ndbm.h, db1-ast/db/db.c,
+ db1-ast/btree/bt_debug.c, db1-ast/mpool/mpool.c,
+ db1-ast/btree/bt_seq.c, db1-ast/recno/rec_get.c,
+ db1-ast/btree/bt_split.c, db1-ast/hash/hash_func.c,
+ db1-ast/btree/bt_utils.c, db1-ast/btree/bt_open.c,
+ db1-ast/recno/rec_seq.c, db1-ast/btree/bt_delete.c,
+ db1-ast/btree/bt_overflow.c, db1-ast/hash/hash_log2.c,
+ db1-ast/btree/bt_search.c, db1-ast/btree/bt_conv.c,
+ db1-ast/btree/bt_close.c, db1-ast/btree/bt_put.c: Bug 8814 - db
+ should look for its header using a relative path, instead of the
+ system path (Fixes FreeWRT)
+
+2007-01-12 14:34 +0000 [r50561] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx.c: minor documentation clarification
+
+2007-01-11 18:11 +0000 [r50517] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #8793 bad response for Unsupported
+ Extension (different fix).
+
+2007-01-11 14:45 +0000 [r50495-50506] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: when we get L2 UP, the L1 is UP
+ definitely too, so we set the L1 state up as well.
+
+ * channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c,
+ channels/misdn/isdn_lib.c: * more additions to make the RESTART
+ message work * added fix for misdn_call to allow SETUPs with
+ empty extensions, replaced the strtok_r functions with strsep for
+ that (inspired by Sandro Cappellazzo, thanks)
+
+2007-01-10 09:51 +0000 [r50335] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/fac.c, channels/misdn/ie.c,
+ channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: more
+ fixes regarding warnings for gcc-4 and first additions for the
+ restart Information element, in the first step we initiate a
+ restart with a CLI command
+
+2007-01-10 04:51 +0000 [r50295] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Add another return value to dial_exec_full that
+ indicates execution is going to continuing at a new
+ extension/context/priority and to just let it slide. (issue #8598
+ reported by jon)
+
+2007-01-10 02:16 +0000 [r50227] Russell Bryant <russell@digium.com>
+
+ * Makefile: Make the number that represents the major version
+ number a single digit instead of 2. Using two digits makes it an
+ octal number when put into version.h, which breaks the
+ compilation of any out of tree module that checks the version for
+ any version after 1.2.7 (reported by Matteo Brancaleoni on the
+ asterisk-dev mailing list, who gave credit to vihai for pointing
+ it out)
+
+2007-01-09 13:30 +0000 [r50150] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: The advent of realtime has enabled people
+ to use commas in the fullname field. This could cause an issue
+ with sending voicemails, when the field is unquoted. (Issue 8595)
+
+2007-01-08 08:37 +0000 [r49922] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.h, channels/misdn/ie.c,
+ channels/misdn/isdn_lib.c: make gcc 4 happy, remove some warnings
+
+2007-01-08 05:10 +0000 [r49889] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Ensure we use the default refresh value of
+ 60 if the remote server does not send one. (issue #8746 reported
+ by maethor)
+
+2007-01-07 21:43 +0000 [r49833] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_dictate.c: If openstream fails, then we crash (Issue
+ 8564)
+
+2007-01-05 16:56 +0000 [r49635] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_iax2.c: ensure that threads which are supposed to
+ be detached (because we aren't going to wait on them) are created
+ properly
+
+2007-01-04 17:45 +0000 [r49354-49447] Matt O'Gorman <mogorman@digium.com>
+
+ * apps/app_voicemail.c: converted a lot of 256 to PATH_MAX and some
+ white space fixes.
+
+ * apps/app_voicemail.c: good catch russell sorry i missed that. fix
+ magic number with proper sizeof
+
+ * apps/app_voicemail.c: When using ODBC_STORAGE VoicemailMain
+ doesn't create the subdirectories for a mailbox such as the INBOX
+ directory. this patch solves that problem, was written by anthony
+ be-125
+
+2007-01-03 08:24 +0000 [r49135-49303] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: * Added check
+ for bridging in misdn_call to avoid setting echocancellation when
+ 2 mISDN channels are involved and when bridging is set. That lead
+ to a kernel panic before under different situations, because we
+ switched about 2 times between hardware bridging and
+ echocancelation * readded MISDN_URATE variable which got lost
+ before, this should make app_v110 work again * fixed typo
+
+ * channels/misdn_config.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, configs/misdn.conf.sample,
+ channels/misdn/isdn_lib.c: added check for channel ranges in the
+ set/empty channel functions. set pmp_l1_check default to no.
+ added misdn restart pid cli command. added cleaning of channel
+ when we send a RELEASE_COMPLETE.
+
+2006-12-29 00:32 +0000 [r49045] Kevin P. Fleming <kpfleming@digium.com>
+
+ * BUGS: location of the bug posting guidelines has changed
+
+2006-12-27 15:43 +0000 [r48974] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 8596: Set CAN_BYE flag for 100 trying
+ too
+
+2006-12-25 05:19 +0000 [r48939-48955] Russell Bryant <russell@digium.com>
+
+ * funcs/func_math.c: Fix an error introduced by copying and pasting
+ the handling of the >= operator for the MATH function. If a
+ single equal sign was used as an operator, the function would
+ treat it is as if it were the >= operator. Now, it properly
+ handles it as an invalid operator. (issue #8665, patch by
+ tempest1)
+
+ * channels/chan_iax2.c: Check for the proper return value on an
+ error in a call to mmap(). This was reported by Andy Wang on the
+ asterisk-dev list. Thanks!
+
+ * channels/chan_sip.c: Remove a couple of misplaced dots in log
+ messages. This was reported by Andrea Spadaccini on the
+ asterisk-dev mailing list.
+
+2006-12-21 20:25 +0000 [r48782] Joshua Colp <jcolp@digium.com>
+
+ * redhat/asterisk.spec: Add new silence sound files to the spec for
+ Redhat. (issue #8652 reported by alvaro_palma_aste)
+
+2006-12-19 21:10 +0000 [r48584] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Free localuser structure when we fail to dial
+ (issue #8612 reported by rizzo)
+
+2006-12-19 13:08 +0000 [r48576] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: when we reject a channel, because it's
+ in use already, we shouldn't process the setup anymore. made the
+ channel allocation a bit easier and more understandable, removed
+ a few unused lines
+
+2006-12-18 10:19 +0000 [r48552] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: when our PTP
+ Partner sends us a SETUP with a preselected channel we just
+ accept it, even when we're NT. added some checks for segfaults.
+
+2006-12-15 10:51 +0000 [r48484] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #8592 - handle 504 as 503 - congestion
+
+2006-12-14 13:03 +0000 [r48467] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: removed FIXUP
+ state. added check for channel allocation conflict when we create
+ a setup while the other site creates a setup on the same channel,
+ besides the check we resolve this conflict.
+
+2006-12-14 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.14 released
+
+2006-12-13 04:23 +0000 [r48434] Steve Murphy <murf@digium.com>
+
+ * channel.c: This small patch fixes bug 8541, where the L option to
+ the Dial app wasn't working right. A similar bug (8386) was filed
+ and fixed earlier, but an intervening bug fix to a DTMF problem
+ broke the L() code in a different way. Hopefully, everything is
+ happy now.
+
+2006-12-12 05:11 +0000 [r48403] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds/silence (added), sounds/silence/1.gsm (added),
+ sounds/silence/10.gsm (added), sounds/silence/2.gsm (added),
+ sounds/silence/3.gsm (added), sounds/silence/4.gsm (added),
+ sounds/silence/5.gsm (added), sounds/silence/6.gsm (added),
+ sounds/silence/7.gsm (added), sounds/silence/8.gsm (added),
+ sounds/silence/9.gsm (added): add silence files
+
+2006-12-11 23:00 +0000 [r48394-48398] Matt O'Gorman <mogorman@digium.com>
+
+ * Makefile, apps/app_externalivr.c, sounds.txt: app_externalivr
+ needs a real silence file, and additional changes to add silence
+ files into core instead of extra patch provided by bug 8177 with
+ minor additions.
+
+2006-12-11 00:33 +0000 [r48374] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_nbscat.c, apps/app_festival.c, apps/app_mp3.c,
+ res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
+ apps/app_ices.c, res/res_musiconhold.c: When doing a fork() and
+ exec(), two problems existed (Issue 8086): 1) Ignored signals
+ stayed ignored after the exec(). 2) Signals could possibly fire
+ between the fork() and exec(), causing Asterisk signal handlers
+ within the child to execute, which caused nasty race conditions.
+
+2006-12-10 02:14 +0000 [r48371] Steve Murphy <murf@digium.com>
+
+ * channels/chan_zap.c: This version applies the patch suggested by
+ stevens in bug 7836 (make inbound channel RINGING state
+ consistent with other channels).
+
+2006-12-09 15:45 +0000 [r48361] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Use locking when accessing the
+ registrations list. This list is not actually used very often, so
+ the likelihood of there being a problem is pretty small, but
+ still possible. For example, if the CLI command to list the
+ registrations was called at the same time that a reload was
+ occurring and the registrations list was getting destroyed and
+ rebuilt, a crash could occur.
+
+2006-12-07 18:14 +0000 [r48356] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Ensure that the file position is not
+ incremented beyond the total number of files available for
+ playback. (issue #8539, ulogic)
+
+2006-12-06 16:05 +0000 [r48322] Russell Bryant <russell@digium.com>
+
+ * configs/iax.conf.sample: Fix the name of the rtignoreregexpire
+ option in the sample configuration file. (issue #8526, arkadia)
+
+2006-12-06 15:48 +0000 [r48321] Christian Richter <christian.richter@beronet.com>
+
+ * doc/README.misdn, channels/chan_misdn.c,
+ channels/misdn/isdn_msg_parser.c: added the export and import of
+ the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the
+ extension is already completely dialed or if there might come
+ additional digits by information elements. also added some docs
+ for that.
+
+2006-12-06 15:42 +0000 [r48320] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #8528 - make sure we don't delete the
+ dialog too quickly after receiving a 487. Move 487 handling into
+ handle_response_invite where it really belongs and don't add an
+ ALREADYGONE flag to the dialog.
+
+2006-12-06 14:35 +0000 [r48319] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: changed a few debugs to higher debug
+ levels
+
+2006-12-06 12:14 +0000 [r48272-48315] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't add Contact header on BYE, CANCEL,
+ MESSAGE requests (Bye, Cancel backported from 1.4, MESSAGE bug
+ reported to me by Gunnar at Omnitor)
+
+ * channels/chan_sip.c: Only set the ALREADYGONE flag once in
+ handle_response()
+
+2006-12-05 01:26 +0000 [r48251] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: If the recording in the database is too
+ large, it will fail to retrieve with an mmap error. Not too sure
+ why this doesn't happen when we put it in the database, also, but
+ since that doesn't seem to be broken, I'm not going to fix it (at
+ least until someone reports it). Solution is to ask for the file
+ in smaller chunks. (Bug 8385)
+
+2006-12-04 21:20 +0000 [r48236-48246] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Revert change from 8016 - this breaks other
+ stuff... Needs further review. Tip: When you've reported a bug
+ about something and somebody has put up a patch for it.. It's not
+ a good idea to open a completely new bug and say that something
+ is broken because of the patch in the other bug - PLEASE mention
+ something in the bug where the patch was actually created.
+
+ * apps/app_voicemail.c: Fix an issue where a message isn't saved
+ correctly when using ODBC storage and reviewing a message. Issue
+ 8016 - patch by sokhapkin.
+
+2006-12-04 18:14 +0000 [r48233] Joshua Colp <jcolp@digium.com>
+
+ * channel.c: If the generic bridge tells us not to retry, and we
+ have a frame to spit out then break the bridge. Props to markit
+ in #asterisk-bugs for bringing this up.
+
+2006-12-01 23:30 +0000 [r48192] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_dial.c: if Dial() is going to send music-on-hold to the
+ calling party, it has to send PROGRESS first to ensure that the
+ reverse audio path has been setup first (BE-106)
+
+2006-12-01 20:19 +0000 [r48183] Jason Parker <jparker@digium.com>
+
+ * configs/extensions.conf.sample: Fix a small typo - issue 8848,
+ reported by pabelanger
+
+2006-11-30 20:47 +0000 [r48165] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 8319 - noriyuki - nonce-count updated
+ *after* use
+
+2006-11-30 20:27 +0000 [r48142-48161] Joshua Colp <jcolp@digium.com>
+
+ * channel.c: Don't write AST_FRAME_NULL or AST_FRAME_IAX frames out
+ to the channel driver. (issue #8390 reported by hselasky)
+
+ * channels/chan_iax2.c: Only print out debug message if bridged
+ channel is not NULL. (issue #8412 reported by jubilex)
+
+ * res/res_features.c: Do not listen for DTMF on the bridge that
+ comes into existence when ParkedCall is executed. This means
+ native bridging can now occur for this. (issue #8406 reported by
+ kebl0155)
+
+ * cdr.c: Print certain CDR messages out at the NOTICE level versus
+ WARNING since they can occur when used with the CDR applications
+ and are perfectly fine. (issue #8367 reported by dartvader)
+
+ * res/res_features.c: Remember the pointer to the allocated block
+ of memory so that we can free it and not cause a memory leak.
+ (issue #8449 reported by arkadia)
+
+ * configs/sip.conf.sample: Document 'port' for SIP peers, came up
+ because of the current mailing list thread. (issue #8450 reported
+ by blitzrage)
+
+2006-11-30 09:05 +0000 [r48127] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Do proper test and don't leave dialogs
+ hanging...
+
+2006-11-29 16:47 +0000 [r48053-48106] Joshua Colp <jcolp@digium.com>
+
+ * rtp.c: If the frame was duplicated before writing out then we
+ need to free it. (issue #8429 reported by edguy3)
+
+ * channels/chan_phone.c: According to the research I have done we
+ never needed to include compiler.h in the first place so let's
+ not! (issue #8430 reported by edguy3)
+
+ * apps/app_voicemail.c: Use the proper function to get the new
+ message count instead of always using the filesystem. (issue
+ #8421 reported by slimey)
+
+2006-11-27 17:15 +0000 [r48045] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * res/res_musiconhold.c: Random MOH wasn't really random (bug 8381)
+
+2006-11-27 15:30 +0000 [r48037] Joshua Colp <jcolp@digium.com>
+
+ * pbx/pbx_spool.c: Do not reference the freed outgoing structure in
+ the debug message. (issue #8425 reported by arkadia)
+
+2006-11-24 14:33 +0000 [r47987] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Change some logging levels. Not having hints
+ is not an ERROR, but still should be reported.
+
+2006-11-23 16:10 +0000 [r47968] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c,
+ channels/misdn/isdn_lib.c: fixed a litle bug regarding
+ HOLD/RETRIEVE. beatufied some logs, changed some loglevels.
+ changed the default value of block_on_alarm
+
+2006-11-23 10:54 +0000 [r47958] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Remove unused variable (rizzo)
+
+2006-11-22 02:19 +0000 [r47910] Steve Murphy <murf@digium.com>
+
+ * channel.c: This is the fix for bug 8386, wherein the time-limit
+ args to dial didn't work correctly
+
+2006-11-20 19:59 +0000 [r47862] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Failing to trap -1 error from mmap causes
+ segfault (Issue 8385)
+
+2006-11-20 19:50 +0000 [r47855-47859] Joshua Colp <jcolp@digium.com>
+
+ * frame.c: Don't forget to byte swap if we are exiting the smoother
+ feed early. (issue #8287 reported by arturs)
+
+ * channels/chan_sip.c: Free history items at the end of use of the
+ temporary SIP pvt structure. (issue #8383 reported by benh)
+
+2006-11-20 10:17 +0000 [r47842] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Just to be safe, disable all the scheduled
+ items after deleting a scheduler entry (rizzo)
+
+2006-11-17 19:02 +0000 [r47802] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channel.c: backport proper channel_find_locked behavior from 1.4
+ branch (noted by Steve Davies on asterisk-dev list)
+
+2006-11-16 23:16 +0000 [r47780] Jason Parker <jparker@digium.com>
+
+ * apps/app_dial.c, apps/app_cut.c, apps/app_directory.c,
+ apps/app_db.c: Fix a couple of typos in applications.. Initially
+ spotted by mrobinson.
+
+2006-11-16 22:57 +0000 [r47776] Kevin P. Fleming <kpfleming@digium.com>
+
+ * doc/README.cdr: update clearly wrong documentation regarding
+ cdr_custom
+
+2006-11-16 20:29 +0000 [r47750-47761] Joshua Colp <jcolp@digium.com>
+
+ * cdr/Makefile: Look for the header file specifically in all cases,
+ not just the existence of the directory. (issue #8358 reported by
+ mrness)
+
+ * channels/chan_local.c: Because of the way chan_local is written
+ we should be extra careful and make sure our callback functions
+ have a tech_pvt. (issue #8275 reported by mflorell)
+
+2006-11-16 16:44 +0000 [r47743] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't fixup if we haven't got PVT.
+ Suggestion from Martin Vit on -dev mailing list inspired by
+ file's commit to chan_local. "This shouldn't happen" ;-)
+
+2006-11-15 22:29 +0000 [r47711] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c: Make sure that the pvt structure exists
+ before trying to do fixup on Local channels. (issue #7937
+ reported by mada123, fix by alamantia with mods by me)
+
+2006-11-15 21:18 +0000 [r47705] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: CANCEL requests are never authenticated
+ (according to RFC 3261)
+
+2006-11-15 20:30 +0000 [r47666-47696] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_voicemail.c: correct argument name typo that caused
+ global variable to be used instead of the one for the specified
+ voicemail user
+
+ * config.c: when re-writing the config file, don't repeat the path
+ if it hasn't changed
+
+ * config.c: when appending a list of variable to a category, ensure
+ the tail pointer points to the last variable in the list
+
+ * config.c: clear the category's variable tail pointer as well when
+ variables are detached from it
+
+ * config.c: ouch... don't use printf, use ast_log/ast_verbose
+
+ * apps/app_voicemail.c, include/asterisk/config.h: ensure that
+ message duration is included in email notifications for forwarded
+ messages (BE-96, fix by me after corydon used his clue-bat on me)
+ ensure that duration in the message metadata is updated if
+ prepending is done during forwarding (related to BE-96) remove
+ prototype for API call that does not exist
+
+2006-11-15 15:17 +0000 [r47648-47655] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Send error message if we fail to allocate
+ sip socket, possibly caused by too few file handles (wasn't
+ possible before, but with the new way of sending temp messages,
+ it is). Found this bug under heavy load testing with SIPP.
+
+ * channels/chan_sip.c: Sending 200 OK and not getting ACK is
+ considered critical for the call.
+
+2006-11-14 22:15 +0000 [r47631] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Update copyright information in the ADSI
+ logo blob.
+
+2006-11-14 11:06 +0000 [r47596] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Avoid collissions between the peerpoke
+ system and the retransmits. Issue #8272. In some cases, changed
+ timers caused the retransmit system to destroy the dialog before
+ peerpoke_expire got a chance.
+
+2006-11-13 21:26 +0000 [r47583] Joshua Colp <jcolp@digium.com>
+
+ * cdr/cdr_pgsql.c: Initialize global pointers for connection and
+ result to NULL. (issue #8356 reported by james)
+
+2006-11-13 20:18 +0000 [r47580] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_sip.c: Having more than 255 old messages caused
+ corruption in the new/old count
+
+2006-11-13 19:04 +0000 [r47571] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't send 487 if we've already sent 200 OK
+ on invite at time of receiving a BYE in the same transaction.
+ (SIPP testing)
+
+2006-11-13 17:05 +0000 [r47549] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_sms.c: When sending an SMS with a user data header
+ properly set the UDH flag in the first byte. (issue #8347
+ reported by hoffmeis)
+
+2006-11-13 05:45 +0000 [r47522-47525] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * res/res_odbc.c: If the execute fails a second time, make sure
+ that we don't pass back a stale handle
+
+ * channels/chan_zap.c: Don't play dialtone if the seizing the
+ channel fails (Bug 7754)
+
+2006-11-12 06:09 +0000 [r47496] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Only do the check to determine whether the
+ channel calling this function is an IAX2 channel when getting the
+ IP address using the special argument, CURRENTCHANNEL. (issue
+ #8341, jcovert)
+
+2006-11-10 20:46 +0000 [r47452-47470] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Clear dialog on loop (backport from 1.4 by
+ mistake)
+
+ * channels/chan_sip.c: - Don't check for ignore in blocks that
+ isn't reached if ignore is on... - return properly after sending
+ reply in handle_request_invite
+
+ * channels/chan_sip.c: Fix multipart/mixed SDP support (issue 8010,
+ alphaque)
+
+2006-11-09 16:48 +0000 [r47379] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_phone.c: Don't include compiler.h on kernels 2.6.18
+ and higher as, well, it's apparently going to be removed. This
+ should make all you FC6 fans happy as your Asterisk will now
+ build without any mods.
+
+2006-11-09 13:09 +0000 [r47359] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h: Fixed segfault when no
+ misdn.conf exists, reported by Igor Neves, thanks.
+
+2006-11-08 07:40 +0000 [r47307-47308] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Remove dialog properly at unload of module
+ (rizzo)
+
+2006-11-07 18:22 +0000 [r47274] Steve Murphy <murf@digium.com>
+
+ * include/asterisk/channel.h, channel.c: This mod for bug_7506, to
+ make the manager code output the proper event
+
+2006-11-07 13:02 +0000 [r47248] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't ever reply to an ACK. (Issue 8265)
+
+2006-11-07 01:22 +0000 [r47238] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: If random order is enabled for files mode
+ music on hold, set a random initial position, instead of always
+ starting at the first file, and doing the random operation only
+ when switching to the next file. (bug reported by John Lange on
+ the asterisk-dev mailing list)
+
+2006-11-02 17:47 +0000 [r46964] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: ignore files in a music on hold directory
+ that begin with '.' (issue #8249, cboie)
+
+2006-11-02 15:15 +0000 [r46899] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't overwrite flags in the packet
+
+2006-11-02 13:55 +0000 [r46876] Russell Bryant <russell@digium.com>
+
+ * callerid.c: Add a missing call to free before returning in an
+ error condition (issue #8268, mrness)
+
+2006-11-01 21:20 +0000 [r46838] Matt O'Gorman <mogorman@digium.com>
+
+ * logger.c: fix for bug #8083 crash caused by double free on m->msg
+
+2006-11-01 19:52 +0000 [r46803] Steve Murphy <murf@digium.com>
+
+ * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
+ accept longer strings or mass confusion and a lot of lost time is
+ the result
+
+2006-11-01 18:24 +0000 [r46776] Russell Bryant <russell@digium.com>
+
+ * res/res_monitor.c: soxmix and Asterisk expect different file
+ extensions for certain formats. This was already handled for the
+ wav49 format. However, it was not handled for ulaw and alaw. I
+ fixed this in such a way that using the alternate extensions for
+ ulaw and alaw will only happen if we know we're calling soxmix,
+ and not a custom script defined using the MONITOR_EXEC variable.
+ The wav49 processing was left alone so that external scripts will
+ see no behavior change. (issue #7550, reported by mnicholson,
+ proposed patch by junky, committed fix is a bit different)
+
+2006-10-31 15:46 +0000 [r46662] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_curl.c: Move thread-unsafe initializer to the module
+ loading code; add the corresponding function to the module unload
+ to fix a memory leak.
+
+2006-10-31 09:49 +0000 [r46585-46610] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Another try to fix
+ ;rport NAT traversal support (issue #7473)
+
+ * channels/chan_sip.c: If peer fails ACL check, fail the REGISTER
+ attempt
+
+ * channels/chan_sip.c: On the other hand, we already copy the NAT
+ flags... Reverting.
+
+ * channels/chan_sip.c: Issue 7473 - support ;rport on REGISTER
+ requests too.
+
+2006-10-31 06:18 +0000 [r46557-46560] Russell Bryant <russell@digium.com>
+
+ * utils.c: When handling the case where the hostname is just an
+ IPV4 numeric address, be sure to set the address type. (issue
+ #8247, alexr)
+
+ * res/res_agi.c: fix some copy/paste bugs in the checking of
+ arguments for the "control stream file" AGI command (issue #8255,
+ mnicholson)
+
+2006-10-30 16:00 +0000 [r46402-46430] Olle Johansson <oej@edvina.net>
+
+ * rtp.c: Bind rtcp to proper IP address
+
+ * channels/chan_sip.c: Issue #7869 - Stop sending 302 redirect when
+ not getting an answer...
+
+ * channels/chan_sip.c: issue #7608: Notifications with wrong
+ content-type. Reported by jsiddall.
+
+2006-10-27 17:36 +0000 [r46361] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c, asterisk.c, apps/app_externalivr.c,
+ res/res_musiconhold.c: We should always be using _exit() after a
+ fork() or vfork() instead of exit(). This is because exit() does
+ some extra cleanup which in some implementations of vfork(), for
+ example, can actually modify the state of the parent process,
+ causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
+
+2006-10-27 09:24 +0000 [r46350] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
+ fixed a bug which caused chan_misdn to try to allocate 2 times
+ the same channel on high load, which then caused instability of
+ mISDN. removed a useless function from isdn_lib.c
+
+2006-10-26 20:06 +0000 [r46344] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #7240, by mistake only committed to
+ trunk (now 1.4), reported by edgreenberg in Issue #7966. Thanks
+ Ed!
+
+2006-10-26 17:47 +0000 [r46332-46337] Jason Parker <jparker@digium.com>
+
+ * contrib/scripts/astgenkey.8: oops - somebody forgot to change
+ this - long ago, probably.
+
+ * channels/chan_skinny.c: Remove a useless ast_mutex_unlock. Issue
+ #8186, patch by anthonyl (fix suggested by benh).
+
+2006-10-25 19:28 +0000 [r46213-46258] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Working to resolve #7608 - adding debug
+ output
+
+ * channels/chan_sip.c: Fix the attack shield for 1.2 too. REFER and
+ NOTIFY can create dialogs in the world of Asterisk.
+
+2006-10-25 08:41 +0000 [r46176] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c,
+ channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
+ added nttimeout option to configure wether we disconnect calls on
+ NT timeouts or not during an overlapdial session
+
+2006-10-23 00:25 +0000 [r45927] Joshua Colp <jcolp@digium.com>
+
+ * cdr/cdr_odbc.c: Don't leak memory mmmk?
+
+2006-10-21 12:35 +0000 [r45808] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed issue, that if chan_misdn is loaded
+ and couldn't be initialized it would cause a segfault after
+ 'reload'. Reported by Drew/Matt thx.
+
+2006-10-19 17:16 +0000 [r45691] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_externalivr.c: Respect language selection when seeing if
+ the file exists (issue #8178 reported by mnicholson)
+
+2006-10-17 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.13 released
+
+2006-10-17 20:37 +0000 [r45380] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Don't create a "real" pvt structure for
+ requests that shouldn't be able to create one. Instead use a
+ temporary pvt and fill it with enough information so we can send
+ a reply.
+
+2006-10-17 17:50 +0000 [r45332] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Fix an integer signedness problem.
+
+2006-10-17 17:22 +0000 [r45326] Kevin P. Fleming <kpfleming@digium.com>
+
+ * LICENSE: provide licensing language for IAXy firmware file
+
+2006-10-17 15:50 +0000 [r45306] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: After some
+ research, we realized that the default behaviour since a long
+ time was doing the right thing, even though the change optimized
+ a bit and removed a lot of potential risks. Conclusion: No need
+ for a configuration option at all.
+
+2006-10-16 19:59 +0000 [r45260-45265] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Use responses
+ rather then replies even though they mean the same thing.
+
+ * channels/chan_sip.c, configs/sip.conf.sample: Add
+ 'ignoreoodreplies' option which will not create a pvt structure
+ on a SIP response but instead basically drop it.
+
+2006-10-14 00:16 +0000 [r45134] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: Made a small update to solve bug 8128; The
+ switch-case fallthru goto to a pattern extension needed to
+ resolved the wildcards to an appropriate digit for extension
+ matching to work
+
+2006-10-13 22:57 +0000 [r45119] Kevin P. Fleming <kpfleming@digium.com>
+
+ * acl.c: don't drop the entire permit/deny list when an attempt is
+ made to add an invalid entry (BE-92)
+
+2006-10-13 19:27 +0000 [r45090] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: avoiding warning, fixing potential bug
+ (backported from 1.2)
+
+2006-10-13 17:01 +0000 [r45060] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_chanspy.c: Turn on volume adjustment if it needs to be
+ on (issue #8136 reported by mnicholson)
+
+2006-10-13 16:18 +0000 [r45048] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: when sending a call to a peer, use the
+ proper socket if we have multiple bindings (reported on
+ asterisk-dev)
+
+2006-10-13 15:49 +0000 [r45030] Joshua Colp <jcolp@digium.com>
+
+ * dnsmgr.c: Pass the right value to usleep for sleeping, and always
+ add the background refresh item back into the scheduler if
+ enabled since it is deleted during reload. (issue #8142 reported
+ by p_lindheimer)
+
+2006-10-13 13:11 +0000 [r44993-45020] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed some
+ echocandisable issues when bridged. this caused a kernel panic
+ sometimes..also some minor formatting fixes
+
+ * channels/misdn/isdn_msg_parser.c: fixed issue, that the
+ hangupcause got a wrong isdn cause at RELEASE_COMPLETE
+
+2006-10-12 18:31 +0000 [r44955] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/utils.h, channels/chan_sip.c, utils.c,
+ netsock.c: ensure that IAX2 and SIP sockets allow UDP
+ fragmentation when running on Linux (thanks to Brian Candler on
+ the asterisk-dev list for the tip)
+
+2006-10-10 13:34 +0000 [r44785] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c, channels/misdn/isdn_lib.c: (re)added
+ support of dynamical enabling hdlc on bchannels
+
+2006-10-09 14:36 +0000 [r44757] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #8101 - wrong parameter for screening
+ in remote-party-id
+
+2006-10-06 16:52 +0000 [r44501-44580] Joshua Colp <jcolp@digium.com>
+
+ * file.c: Even more frames to treat as though the remote side
+ disappeared (issue #8097 reported by eldadran)
+
+ * file.c: Treat busy control frames as hangup in the file streaming
+ core (issue #8097 reported by eldadran)
+
+2006-10-05 10:02 +0000 [r44460] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed segfault which happens during
+ hold/transfer action
+
+2006-10-05 01:27 +0000 [r44392-44432] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix Polycom presence notification again
+
+ * channels/chan_sip.c: remove workaround for old Polycom firmware
+ SUBSCRIBE requests add workaround for new Polycom firmware
+ SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware)
+
+2006-10-04 16:02 +0000 [r44343] Steve Murphy <murf@digium.com>
+
+ * apps/app_macro.c: For bug 7776, I have inserted a warning about
+ Macro nesting vs. stack limitations
+
+2006-10-04 15:26 +0000 [r44334-44335] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: if INFORMATION Message come with keypad
+ instead of called party number, we just use the keypad as called
+ party number.
+
+ * channels/misdn_config.c, channels/misdn/isdn_lib.h,
+ channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
+ configs/misdn.conf.sample, channels/misdn/isdn_lib.c: added the
+ option 'reject_cause' to make it possible to set the
+ RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is
+ automatically rejected because chan_misdn does not support that
+ kind of callwaiting. Therefore chan_misdn supports now 3 incoming
+ channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the
+ info if the requested channel is incoming or outgoing to make the
+ 3. channel possible
+
+2006-10-03 20:14 +0000 [r44296] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: fix a logic error in my previous fix to the
+ queue reload code
+
+2006-10-02 20:07 +0000 [r44168-44213] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Change the fd on the I/O context in case it
+ changed during the reload, which is indeed possible. (issue #7943
+ reported by eclubb)
+
+ * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
+ instead of hardcoding the path for the error message (issue #7942
+ reported by eclubb)
+
+ * io.c: Shrink when current_ioc is unused. It is set to -1 when
+ unused, not 0. (issue #7941 reported by eclubb)
+
+2006-10-02 13:28 +0000 [r44149] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/isdn_lib.c: fixed the hold/retrieve/transfer
+ issues, removed a useless bc field, added setting of
+ frame.delivery fields, some minor code cleanups
+
+2006-10-01 15:19 +0000 [r44110] Russell Bryant <russell@digium.com>
+
+ * configs/queues.conf.sample: Fix the name of the
+ "eventmemberstatus" option in the sample queues.conf (issue
+ #8065, adamg)
+
+2006-09-29 13:44 +0000 [r43977] Kevin P. Fleming <kpfleming@digium.com>
+
+ * cli.c: proper fix for ast_group_t change
+
+2006-09-28 18:00 +0000 [r43924] Joshua Colp <jcolp@digium.com>
+
+ * frame.c, include/asterisk/logger.h, channels/chan_misdn.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ funcs/func_timeout.c, apps/app_festival.c, res/res_features.c,
+ apps/app_hasnewvoicemail.c, apps/app_alarmreceiver.c,
+ channels/iax2-provision.c, res/res_musiconhold.c,
+ res/res_monitor.c: Put in missing \ns on the end of ast_logs
+ (issue #7936 reported by wojtekka)
+
+2006-09-28 17:31 +0000 [r43916] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: fix buggy (and overly complex) loop used during
+ reload of app_queue for static member list updating
+
+2006-09-28 16:37 +0000 [r43897] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_queue.c: app_queue is comparing the device names
+ incorrectly while checking their statuses. It's internal list of
+ interfaces includes the dial string, while the argument passed to
+ this function does not have the dial string (/n for a local
+ channel). This causes it to ignore the device state changes
+ because it thinks it belongs to none of its members. (#8040
+ reported and patch by tim_ringenbach)
+
+2006-09-28 16:32 +0000 [r43895] Kevin P. Fleming <kpfleming@digium.com>
+
+ * cli.c: eliminate compiler warning introduced by recent changes
+
+2006-09-28 16:13 +0000 [r43891] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Stop the stream after waitstream returns so
+ that our formats get restored. (issue #7370 reported by
+ kryptolus)
+
+2006-09-28 15:18 +0000 [r43871] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_queue.c: Fix race condion crash with get_member_status
+ (#7864 - tim_ringenbach reported and patched)
+
+2006-09-27 20:20 +0000 [r43815] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Avoid inability to lock directory log
+ message by creating the directory ahead of time. (Issue 7631)
+
+2006-09-27 19:35 +0000 [r43800] Jason Parker <jparker@digium.com>
+
+ * apps/app_playback.c, pbx.c: Playback() wasn't setting
+ PLAYBACKSTATUS under several circumstances. Playback() returns -1
+ on missing args - so should Background()
+
+2006-09-27 16:54 +0000 [r43778] Russell Bryant <russell@digium.com>
+
+ * res/res_features.c, channel.c: Fix a problem that occurred if a
+ user entered a digit that matched a bridge feature that was
+ configured using multiple digits, and the digit that was pressed
+ timed out in the feature digit timeout period. For example, if
+ blind transfer is configured as '##', and a user presses just
+ '#'. In this situation, the call would lock up and no longer pass
+ any frames. (issue #7977 reported by festr, and issue #7982
+ reported by michaels and valuable input provided by mneuhauser
+ and kuj. Fixed by me, with testing help and peer review from
+ Joshua Colp). There are a couple of issues involved in this fix:
+ 1) When ast_generic_bridge determines that there has been a
+ timeout, it returned AST_BRIDGE_RETRY. Then, when
+ ast_channel_bridge gets this result, it calls ast_generic_bridge
+ over again with the same timestamp for the next event. This
+ results in an endless loop of nothing until the call is
+ terminated. This is resolved by simply changing
+ ast_generic_bridge to return AST_BRIDGE_COMPLETE when it sees a
+ timeout. 2) I also changed ast_channel_bridge such that if in the
+ process of calculating the time until the next event, it knows a
+ timeout has already occured, to immediately return
+ AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
+ anyway. 3) In the process of testing the previous two changes, I
+ ran into a problem in res_features where ast_channel_bridge would
+ return because it determined that there was a timeout. However,
+ ast_bridge_call in res_features would then determine by its own
+ calculation that there was still 1 ms before the timeout really
+ occurs. It would then proceed, and since the bridge broke out and
+ did *not* return a frame, it interpreted this as the call was
+ over and hung up the channels. The reason for this was because
+ ast_bridge_call in res_features and ast_channel_bridge in
+ channel.c were using different times for their calculations.
+ channel.c uses the start_time on the bridge config, which is the
+ time that the feature digit was recieved. However, res_features
+ had another time, 'start', which was set right before calling
+ ast_channel_bridge. 'start' will always be slightly after
+ start_time in the bridge config, and sometimes enough to round up
+ to one ms. This is fixed by making ast_bridge_call use the same
+ time as ast_channel_bridge for the timeout calculation.
+
+2006-09-27 12:51 +0000 [r43764] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/isdn_lib.c: fixed a bug which led to chan_list
+ zombies, when the call could not be properly established in
+ misdn_call. also removed the ACK_HDLC stuff which is not really
+ needed.
+
+2006-09-26 20:49 +0000 [r43708] Russell Bryant <russell@digium.com>
+
+ * asterisk.c: Back in revision 4798, this message was changed from
+ using ast_cli() to directly calling write(). During this change,
+ checking if this was a remote console was removed. This caused
+ this message about using "exit" or "quit" to exit an Asterisk
+ console to come up in times where it did not make sense. This
+ change restores the check to see if this is a remote console
+ before printing the message. (fixes BE-4)
+
+2006-09-26 20:38 +0000 [r43705-43706] Joshua Colp <jcolp@digium.com>
+
+ * .cleancount: I changed the channel structure... let's be sure
+ this gets updated!
+
+ * channels/chan_sip.c, include/asterisk/channel.h: Use proper type
+ to represent the group variable (issue #8025 reported by makoto)
+
+2006-09-26 20:23 +0000 [r43699] Russell Bryant <russell@digium.com>
+
+ * apps/app_voicemail.c: When parsing the sections of voicemail.conf
+ that contain mailbox definitions, don't introduce a length limit
+ on the definition by using a 256 byte temporary storage buffer.
+ Instead, make the temporary buffer just as big as it needs to be
+ to hold the entire mailbox definition. (fixes BE-68)
+
+2006-09-25 21:14 +0000 [r43634] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue
+ 7824): 1) delete=yes was ignored 2) maxmessages was ignored
+
+2006-09-24 13:50 +0000 [r43552] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Check to see if the channel that is
+ activating the IAXPEER function is actually an IAX2 channel
+ before proceeding to process it to avoid crashing. (issue #8017,
+ reported by admott, fixed by myself)
+
+2006-09-22 21:53 +0000 [r43509] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_chanspy.c, channel.c: Yay another 'round of spy fixes!
+ This fixes a small logic flaw with the cleanup function and a
+ memory allocation issue. (issue #7960 reported by jojo & issue
+ #7999 reported by aster1) Special thanks to csum77 for letting me
+ into a box where this issue was happening.
+
+2006-09-21 17:01 +0000 [r43409-43420] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_rpt.c: Whitespace change... really just an excuse to
+ test repotools
+
+ * cdr/cdr_tds.c, cdr/Makefile: TDS 0.64 updates
+
+2006-09-20 05:08 +0000 [r43314] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_misdn.c, channels/chan_sip.c,
+ channels/chan_skinny.c: make some more functions static
+
+2006-09-19 16:21 +0000 [r43269] Matt O'Gorman <mogorman@digium.com>
+
+ * pbx/pbx_gtkconsole.c, apps/app_dial.c, channels/chan_sip.c,
+ apps/app_macro.c, asterisk.c, config.c, apps/app_queue.c, pbx.c:
+ fixes some verbose vs debug issues. patch from bug 2617
+
+2006-09-19 12:28 +0000 [r43248] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: cid is passed to a destructive function;
+ thus a copy is needed (issue 7961)
+
+2006-09-18 20:08 +0000 [r43220] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #7682 - don't add contacts to 4xx
+ responses. (Ugly fix, not proud at all)
+
+2006-09-18 15:30 +0000 [r43163] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_math.c: Add deprecation notice about app_math (issue
+ #7957 reported by k-egg)
+
+2006-09-18 15:05 +0000 [r43160] Steve Murphy <murf@digium.com>
+
+ * configs/zapata.conf.sample: Clarified what "callwaiting" does in
+ zapata.conf.
+
+2006-09-18 15:05 +0000 [r43159] Joshua Colp <jcolp@digium.com>
+
+ * configs/indications.conf.sample: Add number unobtainable tone for
+ New Zealand (issue #7969 reported by nic_bellamy)
+
+2006-09-17 13:54 +0000 [r43072] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_directory.c: Directory used the wrong context for
+ delivery of 0- and *- keypresses (according to Directory's own
+ documentation) - Issue 7965
+
+2006-09-16 07:57 +0000 [r43003-43019] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_iax2.c: When a realtime peer expires, reset the
+ ipaddress in the realtime database back to 0 (Issue 6656)
+
+ * apps/app_meetme.c: When the marked user enters the conference, we
+ should no longer timeout
+
+2006-09-14 22:16 +0000 [r42946] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_zap.c: Error message references wrong argument
+ (Issue 7951)
+
+2006-09-13 19:51 +0000 [r42892] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Backport bugfix patch from 7918 to 1.2 -
+ msg_cfg destroyed before used
+
+2006-09-11 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.12.1 released
+
+2006-09-11 21:47 +0000 [r42697-42783] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_meetme.c, apps/app_page.c: When paging, only wait 5
+ seconds for the marked user to enter the conference. After that,
+ assume the paging already completed by the time the channel
+ entered the conference and drop back out. (Issue 7275)
+
+ * configs/extensions.conf.sample, configs/alsa.conf.sample,
+ configs/zapata.conf.sample, configs/iax.conf.sample,
+ configs/osp.conf.sample, configs/dundi.conf.sample,
+ configs/enum.conf.sample, configs/vpb.conf.sample,
+ configs/cdr.conf.sample, configs/voicemail.conf.sample,
+ configs/phone.conf.sample, configs/misdn.conf.sample,
+ configs/sip.conf.sample, configs/skinny.conf.sample,
+ configs/features.conf.sample: Spelling/grammar fixes (Issue 7929)
+
+ * configs/voicemail.conf.sample: Two grammar issues (bug 7927)
+
+2006-09-09 20:24 +0000 [r42600] Joshua Colp <jcolp@digium.com>
+
+ * channel.c: Only truly consider the channel in the same format if
+ the format matches the raw format OR if a translation path
+ already exists to translate between them. (issue #7887 reported
+ by softins & issue #7803 reported by alvaro_palma_aste). Thanks
+ goes to stubert for giving me access to a box and showing me a
+ scenario where this occured.
+
+2006-09-09 12:14 +0000 [r42535] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: - Reset proper flag - Don't delete SIP
+ dialog prematurely Strangely enough imported from svn trunk...
+ It's confusing here in Greenland. (Committing from 36.000 feet
+ above Greenland, on the way to asterisk@von
+ http://www.pulver.com/asterisk )
+
+2006-09-08 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.12 released
+
+2006-09-08 18:50 +0000 [r42452] Joshua Colp <jcolp@digium.com>
+
+ * channel.c: Swap spies during masquerading
+
+2006-09-08 16:06 +0000 [r42421] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_authenticate.c: Jump logic was backwards: goto returns 0
+ if it succeeds, and we should jump if authentication fails. (Bug
+ #7907)
+
+2006-09-08 04:37 +0000 [r42402] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_local.c: Use ast_best_codec to set the read/write
+ format
+
+2006-09-07 23:12 +0000 [r42355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_record.c: Format vulnerability fix - allowing the user
+ to specify a format is not a good idea (Bug 7811)
+
+2006-09-07 16:30 +0000 [r42260] Joshua Colp <jcolp@digium.com>
+
+ * cdr.c: Let's use the same thing we use in other places to
+ calculate our time for ast_cond_timedwait (issue #7697 reported
+ by bn999)
+
+2006-09-07 02:14 +0000 [r42150-42200] Steve Murphy <murf@digium.com>
+
+ * logger.c: This should fix the problem reported in 7564: logger
+ config file errors getting lost because logging isn't configured
+ yet. The problem was that the code that exists to handle this
+ case was not getting reached, because other tests were causing an
+ early return from ast_log().
+
+ * Makefile: added hours,minutes,seconds .gsm files to the install
+ portion of the makefile, as per bug 7545
+
+2006-09-06 20:02 +0000 [r42148] Joshua Colp <jcolp@digium.com>
+
+ * res/res_agi.c: Don't close the second file descriptor if it's the
+ same as the first one, as it will have already been closed
+ elsewhere and could cause massive panic. (issue #7699 reported by
+ bn999)
+
+2006-09-06 18:16 +0000 [r42133] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_agent.c: Look ma! No more deadlocks! <sic> As
+ posted from #7458 and others similar to it in Mantis: p->app_lock
+ was a mutex really designed for use with agents not in callback
+ mode. That being the case, I've tried to code it so that when
+ callback mode is used, the app_lock mutex will not be
+ locked/unlocked at all. Please let me know how you make out - and
+ if you continue to deadlock now, please reproduce the deadlock
+ logging information and post to Mantis.
+
+2006-09-06 17:10 +0000 [r42110] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed pipe consuming bug when using
+ chanIsAvail (#7878), also moved a debug log to the very begining
+ of misdn_hangup.
+
+2006-09-06 15:55 +0000 [r42054-42086] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Make realtime regseconds work as people
+ expected (0 on registration expiration or release, and actual on
+ normal state) (issue #7684 reported by kshumard)
+
+ * include/asterisk/chanspy.h, apps/app_chanspy.c,
+ apps/app_mixmonitor.c, channel.c: Merge in last round of spy
+ fixes. This should hopefully eliminate all the issues people have
+ been seeing by distinctly separating what each component
+ (core/spy) is responsible for. Core is responsible for adding a
+ spy to a channel, feeding frames to the spy, removing the spy
+ from a channel, and telling the spy to stop. Spy is responsible
+ for reading frames in, and cleaning up after itself.
+
+2006-09-05 16:27 +0000 [r42014] Jason Parker <jparker@digium.com>
+
+ * configs/zapata.conf.sample: Small typo in zapata.conf.sample
+ Reported by ppyy in 7881
+
+2006-09-04 15:46 +0000 [r41989] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't kill the pvt before we have sent ACK
+ on CANCEL
+
+2006-09-03 17:38 +0000 [r41827-41882] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_queue.c: Make sure the forwarded channel inherits
+ variables appropriately when we receive a call forward in the
+ queue. (#7867 - raarts reported and patched)
+
+ * apps/app_queue.c: Don't keep trying the same member in certain
+ strategies when members of the queue are unavailable (#7278 -
+ diLLec reported and patched)
+
+ * apps/app_chanspy.c: Let's NOT spy on Zap/psuedo channels,
+ mmmmmmmmk?
+
+ * apps/app_queue.c: Setting a retry of 0 is generally not a good
+ idea and shouldn't be allowed. (#7574 - reported by regin)
+
+2006-09-01 22:49 +0000 [r41768] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only wipe the redirected audio & video
+ IP/port if it's specified, and trigger a reinvite.
+
+2006-09-01 17:35 +0000 [r41716] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c, include/asterisk/rtp.h, rtp.c: put in proper
+ fix for issue #7294 instead of the broken partial fix that was
+ committed, and thereby also fix issue #7438
+
+2006-09-01 16:33 +0000 [r41690-41691] Joshua Colp <jcolp@digium.com>
+
+ * channel.c: Finish up the last commit (was worse then originally
+ reported)
+
+ * channel.c: Don't treat an unexpected control subclass as voice
+ (issue #7858 reported by PCadach)
+
+2006-08-30 19:01 +0000 [r41423] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #7572 - Hangup when receiving a buggy
+ 487 response to an INVITE
+
+2006-08-30 18:59 +0000 [r41411] Russell Bryant <russell@digium.com>
+
+ * channels/chan_mgcp.c, channels/chan_phone.c,
+ channels/chan_local.c, channels/chan_misdn.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_features.c, channels/chan_h323.c,
+ channels/chan_iax2.c: Restore original functionality of 1.2 in
+ places where ANI was not set, but was changed to be set. The
+ original change was done to ensure that the behavior of the
+ "callerid" option in each channel driver was consistent, but it
+ caused an unexpected behavior change of CDR records for users, so
+ this change is being reverted in 1.2. (issue #7695)
+
+2006-08-30 17:58 +0000 [r41390] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/lock.h: Properly handle an ETIMEDOUT result from
+ pthread_cond_timedwait (issue #7318 reported by arkadia)
+
+2006-08-30 14:31 +0000 [r41334] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 7822 - don't use SRV lookups if it's
+ disabled.
+
+2006-08-29 13:33 +0000 [r41269] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_config.c: clean up last commit ... most notably, there is
+ no reason to do heap allocations here, and it also included a
+ potential memory leak
+
+2006-08-29 05:49 +0000 [r41239-41262] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_config.c: Fixes for bug 7813, via patch submitted by
+ stevens.
+
+ * doc/README.variables: Removed from the docs the mention of the !
+ and =~ operators, as these were knocked out of ast_expr2 because
+ they were new features. Let's hope I can keep them from getting
+ knocked out of the trunk, too!
+
+ * apps/app_macro.c: According to a note added to 7731 by
+ mneuhauser, this will repair a break caused by the last fix
+ (7731).
+
+2006-08-25 15:21 +0000 [r41066-41069] Matt Frederickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Don't send proceeding twice (#7800)
+
+2006-08-25 15:07 +0000 [r41065] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Text only - clarify the reason for entry
+ into authentication mode when the skipuser option is ignored
+
+2006-08-24 19:41 +0000 [r40994] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/linkedlists.h, channel.c, pbx.c: Fix a few
+ issues related to the handling of channel variables - in
+ pbx_builtin_serialize_variables(), the variable list traversal
+ would stop on a variables with empty name/values, which is not
+ appropriate - When removing the GROUP variables, use
+ AST_LIST_REMOVE_CURRENT instead of AST_LIST_REMOVE - During
+ masquerading, when copying the variables list from one channel to
+ the other, using AST_LIST_INSERT_TAIL is not valid for appending
+ a whole list. It leaves the tail pointer of the list invalid.
+ Introduce a new macro, AST_LIST_APPEND_LIST that appends a list
+ properly. (issue #7802, softins)
+
+2006-08-24 17:13 +0000 [r40971-40979] Joshua Colp <jcolp@digium.com>
+
+ * configs/zapata.conf.sample: Minor documentation fix to add the
+ 'dynamic' dialplan option from angler
+
+2006-08-23 16:05 +0000 [r40901] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * res/res_agi.c: Revert last change - breaks retrieval of builtin
+ variables
+
+2006-08-22 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.11 released
+
+2006-08-22 02:59 +0000 [r40821] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_random.c: Bug 7779 - Using initstate(3) means that we
+ cannot unload this module once loaded.
+
+2006-08-21 22:34 +0000 [r40798] Matt O'Gorman <mogorman@digium.com>
+
+ * asterisk.c: Move the load_modules call so that if a module needs
+ realtime support it will work, none do currently but a good move
+ none the less.
+
+2006-08-20 22:09 +0000 [r40692] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * CREDITS: Reformat to match the contribution style of other
+ contributors
+
+2006-08-20 04:49 +0000 [r40601] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Turn media level c= parsing on by default
+ (issue #7725 reported by psm)
+
+2006-08-19 01:03 +0000 [r40446] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c, apps/app_directory.c: Fix a bug with
+ app_voicemail when trying to use app_directory to leave messages
+ to another user (options 3, 5, 2). If the context/extension
+ didn't exist in the dialplan (and why should it have to?), it
+ would fail, saying that it's an "invalid extension". Fix was
+ different in svn trunk. (issue BE-71)
+
+2006-08-18 19:10 +0000 [r40310-40392] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/zapata.conf.sample: make a feeble attempt to avoid the
+ 'how do I enable my hardware echo canceler' questions
+
+ * channels/misdn_config.c (added), channels/chan_misdn_config.c
+ (removed): rename file per crichter's request
+
+2006-08-17 21:57 +0000 [r40306] Christian Richter <christian.richter@beronet.com>
+
+ * doc/README.misdn, channels/misdn/mISDN.patch (removed),
+ channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/fac.c (added), channels/misdn/Makefile,
+ channels/misdn/chan_misdn_config.h, channels/misdn/ie.c,
+ channels/misdn/fac.h (added), channels/misdn/portinfo.c
+ (removed), channels/misdn/isdn_lib_intern.h,
+ channels/chan_misdn_config.c, channels/misdn/isdn_msg_parser.c,
+ configs/misdn.conf.sample, channels/Makefile,
+ channels/misdn/isdn_lib.c: This rather small ;-) commit merges
+ the changes from my team branch 0.3.0 into t he 1.2 branch. These
+ changes include the new mISDN mqueue interface which makes it
+ possible to compile chan_misdn against the current cvs version of
+ mISDN/mISDNuser. These changes also contain various additions and
+ numerous bugfixes to chan_misdn . Each change is documented in
+ the commit logs in the team/crichter/0.3.0 branch.
+
+2006-08-17 16:36 +0000 [r40227] Russell Bryant <russell@digium.com>
+
+ * channel.c: revert bogus change to attempt to fix bug 7506 which
+ actually causes half of the channels not to get "Newchannel"
+ events at all (issue #7745)
+
+2006-08-17 16:22 +0000 [r40223-40225] Joshua Colp <jcolp@digium.com>
+
+ * funcs/func_cdr.c: Use the last CDR entry instead of the first CDR
+ entry for variable retrieving variables using the CDR dialplan
+ function. (issue #7689 reported by voipgate)
+
+ * apps/app_macro.c: Make app_macro compile again
+
+2006-08-17 16:07 +0000 [r40220] Steve Murphy <murf@digium.com>
+
+ * apps/app_macro.c: In app_macro, changed the previously changed
+ upper recursion depth limit to a variable, default of the
+ original val of 7. MACRO_RECURSION is a channel variable that
+ will override the limit, but until I can understand and fix why
+ this limit is neccessary, I am not advertising this variable in
+ the docs. This fix mirrors the changes made in r40200 in trunk.
+
+2006-08-16 18:57 +0000 [r40057] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_mgcp.c: don't allow AUEP responses to overflow the
+ stack during a string copy (reported by Mu Security)
+
+2006-08-15 22:49 +0000 [r39935] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: use pbx_builtin_getvar_helper() so that GET
+ VARIABLE can retrieve global variables (issue #7609)
+
+2006-08-15 22:13 +0000 [r39931] Steve Murphy <murf@digium.com>
+
+ * apps/app_macro.c: This revision fixes bug 7731, the inability for
+ macros to be called more than one level deep in the 'h'
+ extension. It also pushes up the limit of recursion depth from 7
+ to 20.
+
+2006-08-08 18:39 +0000 [r39379] Kevin P. Fleming <kpfleming@digium.com>
+
+ * CREDITS: add explicit listing of anthm's contributions (issue
+ #7683)
+
+2006-08-08 17:04 +0000 [r39350] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Increase the buffer size for the callid
+ (issue #7675, reported by pssatcs)
+
+2006-08-07 01:28 +0000 [r39081] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: Fix a crash reported to me by hads on IRC.
+ This crash would occur with the use of the
+ "distinctiveringaftercid" option. Also, on this user's system,
+ the crash would only occur when built without optimizations. This
+ is because the bug is that the code would write past the end of
+ an array that was allocated on the stack, and the structure of
+ the stack is different with or without optimizations enabled.
+
+2006-08-07 00:15 +0000 [r39056] Joshua Colp <jcolp@digium.com>
+
+ * channel.c: Reset our stream and vstream pointers back to NULL so
+ that any generator that uses them (file based MOH) will not try
+ to close them again. (issue #7668 reported by jmls)
+
+2006-08-05 09:01 +0000 [r38903-38982] Russell Bryant <russell@digium.com>
+
+ * channel.c: Always generate a Newstate event in ast_setstate()
+ instead of making it a Newchannel event if the state was
+ AST_STATE_DOWN. The Newchannel event will always be generated in
+ ast_request(), so this just causes a duplicated Newchannel event
+ in some cases. (issue #7506, repoted by capouch, fixed by me)
+
+ * apps/app_queue.c: remove duplicate queue log entry when the
+ caller exits on a timeout (issue #7616, ppyy)
+
+ * channels/chan_sip.c: don't advertise that this function can set a
+ SIP header when it can only do reads
+
+ * apps/app_dial.c: make sure the priv-callerintros directory exists
+ before trying to create a file there (issue #7659, patch by hads,
+ with some modifications by me)
+
+ * channels/chan_mgcp.c, channels/chan_vpb.c, channels/chan_phone.c,
+ channels/chan_misdn.c, channels/chan_zap.c, channels/chan_sip.c,
+ channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_modem.c, channels/chan_iax2.c: Fix an issue that
+ would cause a NewCallerID manager event to be generated before
+ the channel's NewChannel event. This was due to a somewhat recent
+ change that included using ast_set_callerid() where it wasn't
+ before. This function should not be used in the channel driver
+ "new" functions. (issue #7654, fixed by me) Also, fix a couple
+ minor bugs in usecount handling. chan_iax2 could have increased
+ the usecount but then returned an error. The place where chan_sip
+ increased the usecount did not call ast_update_usecount()
+
+ * channel.c: suppress a compiler warning about the usage of a
+ potentially uninitialized variable
+
+2006-08-03 19:54 +0000 [r38825] Joshua Colp <jcolp@digium.com>
+
+ * res/res_musiconhold.c: Treat the file as invalid if we have no
+ valid formats for it (issue #7643 reported by KNK)
+
+2006-08-03 05:22 +0000 [r38761] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 7648 - Checking wrong count for
+ plurality on new messages for Dutch language
+
+2006-08-02 19:29 +0000 [r38686-38731] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix brain-damage I introduced when trying to
+ fix the CANCEL/BYE sending mechanism for pending INVITES accept
+ unknown 1xx responses as 183 responses (as RFC3261 mandates we
+ should do)
+
+ * res/res_features.c, channel.c: ensure that the 'feature digit
+ timeout' value is taken into account when deciding how long the
+ bridge should run (this fixes a problem report where a digit
+ press that did not invoke a feature is never passed across the
+ bridge)
+
+2006-08-01 19:20 +0000 [r38654] Joshua Colp <jcolp@digium.com>
+
+ * res/res_musiconhold.c: Close the stream when file based MOH stop.
+ This won't get rid of their position in the file but it will
+ cause the translation path to be setup again. (issue #7634
+ reported by asimpson)
+
+2006-07-31 21:14 +0000 [r38611] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: don't reissue hangup requests for SIP
+ channels that have expired their RTP timeouts (one time is
+ enough) don't rescan the SIP private structure list too fast, it
+ can cause channels to not be able to hang up (issue #7495, and
+ probably others) use ast_softhangup_nolock() since we already
+ hold the channel's lock
+
+2006-07-31 17:09 +0000 [r38585] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Add missing code to bring transferee channel
+ out of MOH/autoservice under certain circumstance (issue #7611
+ reported by guillecabeza with minor mods by myself)
+
+2006-07-31 04:06 +0000 [r38546-38547] Russell Bryant <russell@digium.com>
+
+ * frame.c: one more small tweak for thread-safety of TRACE_FRAMES
+
+ * frame.c: Make the frame counting done with TRACE_FRAMES defined
+ thread-safe
+
+2006-07-29 23:18 +0000 [r38501] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: How many attempts does it take to make a SIP
+ URI parser that works well? I'm up to 5 personally. On to the
+ good stuff - parse the domain first, user second, and get rid of
+ port & options/params last. (issue #7616 reported by andrew)
+
+2006-07-28 18:49 +0000 [r38420] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Make a copy of the request URI in
+ check_user_full instead of modifying the one on the structure,
+ and also strip params properly from the user portion of the SIP
+ URI so as to preserve the domain (issue #7552 reported by dan42)
+
+2006-07-27 22:23 +0000 [r38347-38370] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_chanspy.c: use the enum that defines the option
+ arguments, so that the likelihood of mismatched option indexes is
+ reduced (which in this case was a bug, the volume argument was
+ not checked properly)
+
+ * channel.c: do a better job avoiding translation path
+ teardown/setup when not needed
+
+2006-07-27 04:25 +0000 [r38328] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix crash when using the "regexten" option
+ with MALLOC_DEBUG enabled. This was not reported in the bug
+ tracker but the same bug has been demonstrated in other places in
+ the code.
+
+2006-07-27 02:43 +0000 [r38310] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channel.c: don't do useless translation destroy/build when the
+ channel is already in the correct format
+
+2006-07-27 01:58 +0000 [r38288] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: fix a crash when MALLOC_DEBUG is enabled and
+ the regexten is enabled. The crash would occur when the extension
+ got removed. (fixes issue #7484)
+
+2006-07-26 15:26 +0000 [r38234] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Put default callerid into contact when the
+ one specified is either NULL or has a zero string length. (issue
+ #7590 reported by key2)
+
+2006-07-25 19:43 +0000 [r38200] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: This resolves a deadlock that a tech support
+ customer was getting frequently when his users would answer call
+ waiting. If another thread is currently holding the zt_pvt lock
+ for the first channel, unlock both channels and let asterisk
+ retry the native bridge, just like what is done for the second
+ channel directly below these changes.
+
+2006-07-24 17:05 +0000 [r38167] Steve Murphy <murf@digium.com>
+
+ * codecs/gsm/Makefile: This fixes a compile problem for s390 as
+ reported in bug 7253. Tested on both an s390 and non-s390
+ machine.
+
+2006-07-19 17:10 +0000 [r37949] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: ensure that global 'maxauthreq' is reset to
+ zero during 'reload'
+
+2006-07-18 00:41 +0000 [r37828-37856] Russell Bryant <russell@digium.com>
+
+ * frame.c: don't crash if the frame has no data, but has a src
+
+ * frame.c: if asked to duplicate a frame that has no data, don't
+ set the frame's data pointer past the end of the allocatted
+ buffer for the new frame
+
+2006-07-17 22:36 +0000 [r37765-37808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * formats/format_h263.c: Backport buffer increase to 1.2
+
+ * formats/format_h263.c: Overflow bad
+
+2006-07-15 23:29 +0000 [r37691] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * enum.c: Bug 7513 - ensure that each time we do a query, the
+ results are returned in the same logical order, so that when we
+ iterate over the list, we get all results, not some results
+ repeated, due to insufficient sorting.
+
+2006-07-14 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.10 released
+
+2006-07-14 13:31 +0000 [r37612] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_sms.c: Bug 7526 - previous commit broke app_sms
+
+2006-07-13 21:22 +0000 [r37571] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_voicemail.c: don't fail/abort if the message category
+ sound file cannot be played, just generate a warning message and
+ continue message playback
+
+2006-07-13 18:44 +0000 [r37546] Russell Bryant <russell@digium.com>
+
+ * rtp.c: yeah, ummm... This frame pointer should not be static.
+ This situation only exists in 1.2 (pointed out by Constantine
+ Filin on the asterisk-dev mailing list)
+
+2006-07-13 16:44 +0000 [r37531] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: report address of peer trying to subscribe
+ to unknown hint
+
+2006-07-13 15:45 +0000 [r37458-37516] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * doc/README.enum: Bug 7532 - Typo in enum example
+
+ * contrib/init.d/rc.mandrake.zaptel: Merge fixup for asterisk
+ startup script to zaptel startup script
+
+2006-07-12 15:53 +0000 [r37441-37442] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_voicemail.c: fix a weird case where a lock file could be
+ left (but would happen almost never)
+
+ * app.c: fix a case where ast_lock_path() could leave a
+ randomly-named lock file hanging around make ast_unlock_path
+ actually report when unlocking fails
+
+2006-07-12 15:23 +0000 [r37439] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Add support to have maxauthreq as a global
+ option
+
+2006-07-12 13:54 +0000 [r37417-37419] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c, utils.c, res/res_agi.c, apps/app_zapras.c,
+ asterisk.c, channels/chan_modem.c, channels/chan_iax2.c: remove
+ some more bad examples of using printf
+
+ * enum.c, pbx/pbx_config.c: get rid of some more printf's (although
+ most of these were ifdef-ed out)
+
+2006-07-12 03:55 +0000 [r37402] Matt O'Gorman <mogorman@digium.com>
+
+ * app.c: GRRR no fprintf!
+
+2006-07-11 19:00 +0000 [r37378] Joshua Colp <jcolp@digium.com>
+
+ * configs/iax.conf.sample, channels/chan_iax2.c: Add configuration
+ option for IAX2 users that will limit the amount of outstanding
+ AUTHREQs we are waiting for replies on.
+
+2006-07-10 21:01 +0000 [r37361] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channel.c: do masquerade-behind-proxy checking with better
+ control over locks
+
+2006-07-07 23:57 +0000 [r37307] Joshua Colp <jcolp@digium.com>
+
+ * rtp.c: Change message regarding marker bit forcing when SSRC
+ changes to be shown only during debug so it doesn't overload high
+ capacity systems
+
+2006-07-06 21:41 +0000 [r37224] Matt O'Gorman <mogorman@digium.com>
+
+ * channel.c: patch resolves issue with when to decide if its right
+ time to native bridge, feature redirect was not being checked.
+ patch from bug #7296
+
+2006-07-06 20:38 +0000 [r37212] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_agent.c: Don't do weird things on a callback agent
+ that has attempted logoff while still on the phone.
+
+2006-07-06 15:48 +0000 [r37173] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Instead of giving the scheduled item ID on a
+ peer expiration, give the time until they expire (issue #7455
+ reported by slavon)
+
+2006-07-06 13:47 +0000 [r37143] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * funcs/func_db.c: Fix spelling/grammar (issue 7493)
+
+2006-07-05 15:31 +0000 [r36998] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_oss.c: Spell extension correctly in documentation
+ for chan_oss dial (issue #7487 reported by flefoll)
+
+2006-07-04 14:45 +0000 [r36838-36911] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Tell clients based on old SIP standard that
+ we only support MD5 digest authentication...
+
+ * channels/chan_sip.c: issue #7470 - Need larger buffer for
+ record-route headers...
+
+2006-07-03 05:12 +0000 [r36697-36751] Russell Bryant <russell@digium.com>
+
+ * asterisk.c: fix a race condition that caused asterisk to log a
+ *ton* of warnings on mac osx about poll returning an error
+ because the polled file descriptor was bad.
+
+ * channels/chan_mgcp.c, channels/chan_phone.c,
+ channels/chan_local.c, channels/chan_misdn.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ channels/chan_agent.c, channels/chan_features.c,
+ channels/chan_h323.c, channels/chan_modem.c,
+ channels/chan_iax2.c: use ast_set_callerid to be more consistent
+ and to make sure that the "callerid" option in the conf files is
+ always handled the same way and sets ANI (issue #7285, gkloepfer)
+
+ * dsp.c: fix the build with BUSYDETECT_TONEONLY defined (issue
+ #7414)
+
+2006-06-30 14:05 +0000 [r36290-36377] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_directory.c: Bug 7349 - Directory did not work correctly
+ when USE_ODBC_STORAGE was defined.
+
+ * Makefile: Bug 7388 - compatibility changes for Solaris
+
+2006-06-29 07:19 +0000 [r36253-36254] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/queues.conf.sample: clarify documentation for
+ 'persistentmembers' setting
+
+ * configs/sip.conf.sample: add documentation for peer-specific
+ 'outboundproxy' setting
+
+2006-06-28 14:12 +0000 [r36187] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't delete scheduled item twice in
+ sip_destroy (already fixed in svn trunk)
+
+2006-06-26 17:10 +0000 [r36078] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: ensure that two SIP channels that exist at
+ the same moment will not have the same channel names (issue
+ #7245, different fix)
+
+2006-06-26 15:27 +0000 [r36043] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 6997 maybe, but anyway - don't
+ retransmit responses to NON-invite requests.
+
+2006-06-25 15:10 +0000 [r35915] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_sip.c: Bug 7425 - Size of buffer is passed in by
+ len
+
+2006-06-23 11:30 +0000 [r35669] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_queue.c: We should lock the queue before we go making
+ changes to member interface statuses.
+
+2006-06-21 19:25 +0000 [r35334] Joshua Colp <jcolp@digium.com>
+
+ * configs/indications.conf.sample: Add Venezuelan indications
+ (issue #7402 reported by palillo)
+
+2006-06-20 15:05 +0000 [r35121] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * stdtime/private.h: Bug 7398 - Solaris puts its zoneinfo files in
+ a nonstandard place
+
+2006-06-20 10:27 +0000 [r35058] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #6820 - Possible fix (already
+ implemented in trunk)
+
+2006-06-19 20:27 +0000 [r34911] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Call reset_user_pw upon changing the
+ password using externpass (issue #7395 reported by Ryan Cumming)
+
+2006-06-19 18:07 +0000 [r34875] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Issue 7357 - txt file left behind when
+ going to operator. Also, fix a possible file descriptor leak.
+
+2006-06-18 21:03 +0000 [r34627-34655] Russell Bryant <russell@digium.com>
+
+ * pbx.c: don't set state to BUSY if the channel is already in the
+ UP state (issue #7376, backported from trunk)
+
+ * configs/iax.conf.sample, channels/chan_iax2.c: don't store
+ multiple secrets delimited with semicolons for peers because this
+ is only valid for users. Instead, only keep the last specified
+ secret for a peer entry. Also, document how multiple secrets are
+ handled in the sample config. (Reported by PCadach on
+ #asterisk-bugs)
+
+2006-06-16 03:37 +0000 [r34400] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Zero out a declared structure so as to not
+ crash if it contains invalid data (reported by Qwell on
+ #asterisk-dev)
+
+2006-06-15 14:11 +0000 [r34306] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 7294 - patch by phsultan - Asterisk
+ sends Invite instead of BYE in some cases.
+
+2006-06-15 13:30 +0000 [r34274] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: don't use prefixed structure names for internal
+ structures don't use a plural structure name for a singular
+ object
+
+2006-06-15 12:40 +0000 [r34242] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: VoicemailMain exits on any key, when the
+ language is set to Italian, instead of properly handling the key
+ (issue 7353).
+
+2006-06-14 22:22 +0000 [r33841-34160] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: coding style cleanups on queue interface
+ handling code that was committed for the last release
+
+ * channels/chan_iax2.c: use existing dial string parser for strings
+ supplied to iax2_devicestate, because they can be complete dial
+ strings, not just device names
+
+ * include/asterisk/plc.h, jitterbuf.c, plc.c, apps/app_dumpchan.c,
+ apps/app_chanspy.c: clarify file headers that mention disclaimer
+ usage
+
+ * file.c: don't output 'no format found' when we _did_ find the
+ format but couldn't open the desired file for some other reason
+
+ * apps/app_mixmonitor.c: memory allocation optimizations
+
+2006-06-13 12:40 +0000 [r33753-33813] Russell Bryant <russell@digium.com>
+
+ * pbx.c: remove duplicate mutex_unlock
+
+ * apps/app_voicemail.c: fix various places where the code returns
+ without unlocking vmlock or destroying loaded configuration
+
+ * apps/app_festival.c: add a missing close of an open fd, destroy
+ of open config, and removal of the calling channel from the
+ localusers list
+
+ * asterisk.c: revert a change that caused more problems than it
+ fixed and fix the real problem in this code. fds was declared as
+ an array of zero size which caused some weird problems, some of
+ which would only be seen when compiling without optimizations.
+ (fixes issues #7071, #7326, and #7305)
+
+2006-06-12 21:34 +0000 [r33724] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/chanspy.h, apps/app_mixmonitor.c, channel.c:
+ Greatly simply the mixmonitor thread, and move channel reference
+ directly to spy structure so that the core can modify it.
+
+2006-06-12 20:40 +0000 [r33693] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: fix a place where a frame would be free'd twice
+
+2006-06-12 16:03 +0000 [r33638] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_local.c: only allow chan_local to masquerade the
+ outbound channel onto its owner, instead of the other way around
+ (this will ensure that group variables on the outbound channel are
+ preserved)
+
+2006-06-12 15:27 +0000 [r33615] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * res/res_agi.c: Move set priority up, because at this point in the
+ code, stdout is no longer the console. If we're unable to set
+ priority, the error goes to Asterisk as if it were an AGI command
+ (issue 7335).
+
+2006-06-11 21:21 +0000 [r33449-33548] Russell Bryant <russell@digium.com>
+
+ * pbx.c: fix another place where a frame does not get free'd
+
+ * apps/app_meetme.c: fix up five little places where frames would
+ not be free'd and remove an unnecessary mutex_unlock where there
+ is no way for it to be locked at that time
+
+ * apps/app_ices.c: fix a place that would leak a frame (all of
+ these fixes are in applications that call ast_read() on a channel
+ but have code paths in them that would not free the frame)
+
+ * apps/app_festival.c: fix a couple places that would leak a frame
+
+ * apps/app_alarmreceiver.c: fix two places that would cause a frame
+ to be leaked
+
+ * apps/app_url.c: fix a case where an HTML frame would be leaked
+
+ * apps/app_test.c: Free frames read from the channel when measuring
+ noise. This resulted in about 9 or 10 seconds of leaked frames in
+ both the TestClient and TestServer applications
+
+ * apps/app_zapbarge.c, apps/app_zapscan.c: backport a couple of
+ frame leak fixes from the trunk (revisions 33446, 33447)
+
+2006-06-09 18:52 +0000 [r33264-33300] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Allow the format outputted by meetme list to
+ be used for meetme commands (like kick) (issue #7322 reported by
+ darkskiez)
+
+ * channels/chan_iax2.c: Remove an unneeded double lock (issue #7310
+ reported by arkadia)
+
+ * apps/app_dial.c: Handle hangup during recording of screened name
+ (issue #7304 reported by kulldominique)
+
+ * apps/app_meetme.c: Add missing newlines (issue #7323 reported by
+ darkskiez)
+
+2006-06-09 15:53 +0000 [r33235] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Do not require a context on a domain=
+ setting
+
+2006-06-08 16:57 +0000 [r33036] Kevin P. Fleming <kpfleming@digium.com>
+
+ * frame.c: handle out-of-memory conditions properly in
+ ast_frisolate() (reported by Slav Kenov on asterisk-dev mailing
+ list)
+
+2006-06-07 17:53 +0000 [r32818] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: fix some broken code with
+ BRIDGE_OPTIMIZATION defined (issue #7292)
+
+2006-06-06 16:55 +0000 [r32605] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 7287 - A too short voicemail with
+ ODBC_STORAGE will cause the first voicemail to be deleted
+ erroneously
+
+2006-06-06 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.9.1 released
+
+2006-06-06 16:02 +0000 [r32582] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * callerid.c: Bug 7268 - Callerid leaks memory on error
+
+2006-06-06 15:48 +0000 [r32566] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: clean up yesterday's security fix to not
+ cause breakage when video mini frames are received
+
+2006-06-03 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.9 released
+
+2006-06-05 19:53 +0000 [r32373] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: ensure that the received number of bytes is
+ included in all IAX2 incoming frame analysis checks (fixes a
+ known vulnerability)
+
+2006-06-04 03:43 +0000 [r31921] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: return bridge exit logic to what it was before
+ i broke it :-(
+
+2006-06-03 17:02 +0000 [r31775] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: when using moh files mode, don't look for
+ a file past the number of files that have been loaded, or worse,
+ past the size of the files array
+
+2006-06-01 21:46 +0000 [r31321-31555] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_musiconhold.c: remove pointless forcing of the channel
+ into SLINEAR mode; the write format will be set later based on
+ the file that is chosen to be played to the channel
+
+ * include/asterisk/channel.h, channel.c: handle Zap transfers
+ behind chan_agent properly so the agent doesn't get stuck waiting
+ for the call to hang up
+
+ * configs/sip.conf.sample: remove a sample entry that never should
+ have been added (code to support it was not merged)
+
+2006-05-31 23:50 +0000 [r31194] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: if the connection to a FastAGI server fails
+ because of a timeout, log a more informative log message
+
+2006-05-31 22:26 +0000 [r31161] Kevin P. Fleming <kpfleming@digium.com>
+
+ * rtp.c: silence a warning message that is not a warning
+
+2006-05-31 20:26 +0000 [r31127] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: fix misplaced manager event (issue #6866,
+ flefoll)
+
+2006-05-30 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.8 released
+
+2006-05-30 14:55 +0000 [r30770] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_queue.c: Fix infinite loop scenario and add some sanity
+ checking to prevent segfault on a NULL parameter coming in (which
+ probably shouldn't happen, but just to be safe...)
+
+2006-05-26 17:09 +0000 [r30424-30546] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_queue.c: A new way to try and deal with deadlocks that
+ occur in app_queue at present. Using this approach, we only
+ manipulate the main queue mutexes when we get a dev state change
+ on a device that is actually a member of a queue. Backported from
+ /trunk for the "bug fix".
+
+2006-05-25 20:03 +0000 [r30373] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Don't play the enter sound twice when a person
+ joins a conference after the leader has joined it. (issue #6138
+ reported by shanermn)
+
+2006-05-25 17:39 +0000 [r30293-30296] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/gsm/Makefile: don't try to use -march=s390 when building
+ on S/390 systems (reported via asterisk-users mailing list)
+
+ * channels/chan_sip.c: allow SIPCHANINFO(peername) to work for
+ calls from users as well (issue #7215)
+
+2006-05-25 15:27 +0000 [r30239] Joshua Colp <jcolp@digium.com>
+
+ * configs/extensions.conf.sample: Get rid of an incorrect SIP dial
+ string in the sample extensions.conf - I even tried variations...
+ no go (issue #7222 reported by arkadia)
+
+2006-05-24 21:24 +0000 [r30069-30098] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: oops... make sure to stop processing a
+ request once we have sent an authentication challenge (issue
+ #7220)
+
+ * channels/chan_sip.c: don't send CANCEL on a pending INVITE if we
+ haven't received a provisional response yet... mark it pending
+ until the first response is received (issue #7079)
+
+2006-05-24 19:55 +0000 [r30037] Matt O'Gorman <mogorman@digium.com>
+
+ * apps/app_meetme.c: app_meetme used the ast_max_exten instead of
+ path_max solves bug 6822
+
+2006-05-24 19:44 +0000 [r30033-30035] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Merge branch for bug 6264 (Privacy option 2
+ returns dial-status ANSWER / option_priority_jumping not
+ respected) (reported by jkoopmann and branch by murf)
+
+ * logger.c: Fix deadlock caused by a race condition in the logger
+ when reloading (issue #7195 reported and fixed by softins)
+
+2006-05-24 16:59 +0000 [r29904-29973] Kevin P. Fleming <kpfleming@digium.com>
+
+ * res/res_agi.c: support video recording via AGI 'RECORD FILE'
+ command (issue #7068)
+
+ * apps/app_queue.c: fix various bugs related to exiting from queue
+ via keypress and moh handling (issue #6776, different fix)
+
+ * channels/chan_zap.c: respect 'usecallingpres' in zapata.conf even
+ if CLID has not been set for the channel (issue #7123)
+
+ * channels/chan_sip.c, configs/sip.conf.sample: add an option to
+ allow the admin to 'hide' SIP user/peer names from systems trying
+ to 'fish' names
+
+2006-05-23 21:44 +0000 [r29849] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: fix the sourceaddress option (issue #7213,
+ alphaque)
+
+2006-05-23 18:16 +0000 [r29764] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: simplify/fix lock retry, and fix comment
+
+2006-05-23 17:17 +0000 [r29733] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_sip.c: Sanity check code for an extended failure in
+ trying to obtain a channel lock that may have been obtained
+ elsewhere. Prevents the monitor thread of the SIP module from
+ going into an infinite loop, effectively, breaking SIP until you
+ restart Asterisk or the mutex is unlocked, whichever comes first.
+
+2006-05-23 17:15 +0000 [r29732] Kevin P. Fleming <kpfleming@digium.com>
+
+ * dnsmgr.c, res/res_features.c, include/asterisk/linkedlists.h,
+ include/asterisk/lock.h, apps/app_sql_postgres.c, pbx.c: backport
+ some mutex initialization and linked list handling fixes from
+ trunk
+
+2006-05-23 15:58 +0000 [r29696] BJ Weschke <bweschke@btwtech.com>
+
+ * res/res_features.c: Fix a potential leak and correct (hopefully)
+ a segfault under certain conditions. #6784 (vovan and perry
+ testing)
+
+2006-05-22 21:27 +0000 [r29464-29555] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_waitforsilence.c: Increase the silence threshold to 128
+ to "fix" it, so I'm told. (issue #6595 reported by davetroy fixed
+ by casper)
+
+ * res/res_features.c: Use the correct language when playing the
+ transfer sound (issue #7109 reported by casper)
+
+ * channels/chan_local.c: Preserve presentation bit when going
+ through chan_local (issue #7002 reported by acunningham)
+
+2006-05-22 14:59 +0000 [r29394-29398] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_meetme.c: Bug 7194 - spelling fix
+
+ * pbx.c: Bug 7196 - month range did not work
+
+2006-05-21 15:16 +0000 [r29196] BJ Weschke <bweschke@btwtech.com>
+
+ * res/res_features.c: When an application that is executed via
+ applicationmap and exits non-zero, make sure that we pass through
+ the correct return value from the application to make sure a
+ segfault doesn't occur by a bridge trying to continue when it
+ should not. Also, when executing applications via applicationmap,
+ make sure that the application is executed against the channel
+ whose DTMF caused it to be fired off in the first place. (part
+ 1/2 of #7090 - this is the only fix that will be applied to both
+ 1.2 and /trunk) acunningham and blitzrage on testing...
+
+2006-05-20 19:50 +0000 [r29052] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: fix the possibility of writing one byte past
+ the end of a buffer. (issue #7189, Mithraen)
+
+2006-05-20 02:35 +0000 [r28968] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: don't allow queue member devices to ring longer
+ than the total queue timeout (issue #6423, reported and patched
+ by bcnit)
+
+2006-05-20 02:31 +0000 [r28966] Russell Bryant <russell@digium.com>
+
+ * apps/app_sms.c: fix a case where code made assumptions about how
+ memory for variables is allocatted on the stack - this patch is
+ slightly different than the one that went in for the trunk
+
+2006-05-20 00:55 +0000 [r28794-28896] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: don't try to predict where the compiler
+ will place things on the stack... put them in the right place
+ explicitly (issues #7029 and #7100, maybe others)
+
+ * channels/chan_sip.c: use the specified 'subscribecontext' for a
+ peer rather than the context found via the target domain (domain
+ contexts are for calls, not for subscriptions) (issue #7122,
+ reported by raarts)
+
+2006-05-19 19:18 +0000 [r28754-28790] Russell Bryant <russell@digium.com>
+
+ * utils/smsq.c: fix the build of smsq with -Werror. I learned
+ something new about format strings from this patch! (issue #7141,
+ Mithraen)
+
+ * asterisk.c: This explicit poll is only needed on mac. In fact, it
+ breaks some systems such as some versions of Fedora, causing
+ 'asterisk -rx' to never exit. This has been tested on systems
+ showing the asterisk -rx problem, as well as other unaffected
+ versions of linux, mac osx 10.4, and FreeBSD 6. (issue #7071)
+
+2006-05-19 17:04 +0000 [r28627-28698] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_zap.c: Make the minidle option actually exist as
+ documented (issue #7159 reported by imran)
+
+ * apps/app_voicemail.c: When forwarding messages use the context
+ that the active voicemail user was found in. (issue #7010)
+
+ * enum.c: Backport of fix for issue #6654 that was fixed in trunk
+ but not here
+
+ * apps/app_queue.c: Treat paused queue members as unreachable
+ (issue #7127 reported by peterh)
+
+2006-05-18 20:43 +0000 [r28335-28384] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix up a few more places to find the SDP
+ properly (fallout from fix for #7124)
+
+ * channels/chan_sip.c: handle incoming multipart/mixed message
+ bodies in SIP and find the SDP, if present (issue #7124 reported
+ and patched by eborgstrom, but very different fix)
+
+ * enum.c: use unsigned counters for handling answer/IE lengths
+ while processing DNS results (issue #7174)
+
+ * channels/chan_sip.c: support 'inactive' tag for SDP media streams
+ (simple fix, proper fix will appear in 1.4 release) (issue #7130)
+
+2006-05-18 17:27 +0000 [r28257] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_hasnewvoicemail.c: Bug 7167 - HasNewVoicemail and
+ VMCOUNT() didn't work when USE_ODBC_STORAGE was defined
+
+2006-05-18 16:31 +0000 [r28169-28212] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Return -1 on error in ODBC messagecount and
+ 0 on success (issue #7133 reported by cfieldmtm)
+
+ * apps/app_voicemail.c: Fix endless looping message by checking
+ value of res before doing retries stuff. (issue #7140 reported by
+ tanischen)
+
+2006-05-18 12:13 +0000 [r28125] Olle Johansson <oej@edvina.net>
+
+ * apps/app_meetme.c: Video in meetme? Hmmm. Removed until we do
+ have some code for it.
+
+2006-05-17 22:34 +0000 [r27973] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Fix codec priority stuff during
+ authentication (issue #6194 reported by jkoopmann)
+
+2006-05-17 19:27 +0000 [r27927] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #7176 - Crash in expire_register (We
+ need to find out what's causing peer to be undefined, so this is
+ just a bandaid, not a real fix)
+
+2006-05-17 17:07 +0000 [r27767-27847] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Priority jumping not working on VoiceMail
+ app with new syntax (issue #7164 reported and fixed by
+ alvaro_palma_aste)
+
+ * apps/app_osplookup.c: OSPNext does not handle success/failure
+ correctly (issue #7147 reported and fixed by eborgstrom)
+
+2006-05-17 09:21 +0000 [r27723] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: chan_sip did not use the TRANSFER_CONTEXT
+ for transfers, like res_features. Now fixed.
+
+2006-05-17 02:19 +0000 [r27636] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 7125 - Fix race condition between
+ resequencing and leaving a message
+
+2006-05-16 23:31 +0000 [r27594] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c: Inherit channel variables during call forwards
+ when going through chan_local (issue #7095 reported by raarts)
+
+2006-05-16 20:05 +0000 [r27468] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channel.c: don't leak frames when deferring DTMF or dropping
+ duplicate ANSWER frames (issue #7041, slightly different fix,
+ reported/patched by clausf)
+
+2006-05-13 04:08 +0000 [r27093] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 7134 - File descriptor leak with ODBC
+ storage of voicemail
+
+2006-05-11 23:02 +0000 [r27051] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * funcs/func_logic.c: Bug 7086 - pbx_checkcondition substitution,
+ so that arbitrary strings are true (for regex)
+
+2006-05-11 09:05 +0000 [r26760-26773] Kevin P. Fleming <kpfleming@digium.com>
+
+ * rtp.c: backport fix from trunk for bug #6934, ensuring that RTP
+ mark bit is changed when SSRC changes
+
+ * channels/chan_sip.c: ensure that we send a response to REGISTER
+ requests that are successfully authenticated but contain invalid
+ Contact URIs
+
+2006-05-09 14:18 +0000 [r26050-26090] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_sip.c, doc/README.variables: Add the appropriate
+ jumping behavior that is the standard for 1.2.X to SIPGetHeader
+ that is now deprecated in /trunk. #7111 (blitzrage!!!)
+
+ * apps/app_voicemail.c: Correct memory leak in find_user_realtime
+ #7118 (fnordian)
+
+2006-05-08 15:09 +0000 [r25608] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 7103 - mikma - The header is named
+ "Require" - Don't reply to ACK (Not using patch against trunk)
+
+2006-05-08 14:12 +0000 [r25518-25563] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_agent.c: Don't show agents as available when they
+ are in wrap-up time. #6726 (ZX81)
+
+ * apps/app_queue.c: Make QueueStatusComplete event thread safe by
+ wrapping it inside the queue lock clause already there. #7013
+ (bziherl reporting)
+
+ * apps/app_queue.c: Don't recheck valid_exit() after getting the
+ result from say_position (which already checks it). Should
+ prevent another loop if the caller hits digits during the
+ position announcement. #6776 (tgj reporting)
+
+2006-05-08 11:16 +0000 [r25442] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Incorrect log statement when playing transfer
+ sounds (issue #7008 reported and fixed by nathan)
+
+2006-05-07 13:38 +0000 [r25288-25322] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_meetme.c: Fix playback behavior to exit correctly when
+ we receive a hangup during playback of the invalid pin message.
+ #7091 (AntD reporting)
+
+ * asterisk.c: Reset the value of ast_mainpid if we fork so future
+ remote unix connections display the correct PID. #7098 (tzafrir
+ reporting)
+
+2006-05-06 02:32 +0000 [r25015-25165] Russell Bryant <russell@digium.com>
+
+ * frame.c: fix a problem where the frame's data pointer is
+ overwritten by the newly allocated data buffer before the data
+ can be copied from it. This is in the ast_frisolate() function
+ which is rarely used. (issue #6732, stefankroon)
+
+ * channels/chan_zap.c: ensure that the appropriate manager events
+ are sent in all of the places where alarms are detected or
+ cleared (issue #6866, flefoll)
+
+ * channels/chan_h323.c: update chan_h323 to reflect the new
+ prototype for rtp_set_peer (issue #6560, casper) This was fixed a
+ couple months ago in the trunk, but never in 1.2.
+
+2006-05-05 20:44 +0000 [r25014] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_voicemail.c, include/asterisk/app.h, app.c: Voicemail
+ fixes along with an API change approved by russellb to fix the
+ bug(s). (jcollie and supczinskib) #7064
+
+2006-05-05 17:39 +0000 [r24837-24911] Russell Bryant <russell@digium.com>
+
+ * apps/app_while.c, apps/app_macro.c: use pbx_checkcondition()
+ instead of ast_true() to evaluate the condition for MacroIf and
+ WhileIf (issue #7086)
+
+2006-05-04 16:27 +0000 [r24706] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_queue.c: Bug 7023 - reload should not unpause members
+
+2006-05-04 11:17 +0000 [r24567-24669] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_verbose.c: Make sure that only the "|" is a recognized
+ delimiter for Verbose(), as the app documentation already
+ specifies. #7080 (alessiof reporting)
+
+ * apps/app_dial.c: Correct application documentation to make users
+ aware that certain options cannot be used in conjunction with
+ others. #6666 (chotaire)
+
+2006-05-03 18:31 +0000 [r24496] Russell Bryant <russell@digium.com>
+
+ * redhat/asterisk.spec: fix up "make rpm" - don't reference the
+ gzipped man page, because we don't store them compressed anymore
+ - add some files that currently were not listed (issue #6837)
+
+2006-05-03 12:39 +0000 [r24381] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #7074 - Problem with long contact
+ lines
+
+2006-05-02 19:39 +0000 [r24295] BJ Weschke <bweschke@btwtech.com>
+
+ * file.c: Make certain ast_stopstream() sets the channel's stream
+ members to NULL after closing them. #7067 (jcomellas)
+
+2006-05-02 02:12 +0000 [r24019-24097] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_privacy.c: Prompt does not request '#' to end input, so
+ the application should not require it
+
+ * apps/app_nbscat.c, apps/app_festival.c, apps/app_mp3.c,
+ apps/app_zapras.c, asterisk.c, apps/app_externalivr.c,
+ apps/app_ices.c, res/res_musiconhold.c,
+ include/asterisk/options.h: Bug 6864 - drop realtime priority on
+ ALL external processes
+
+2006-05-01 19:34 +0000 [r23985-23988] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_voicemail.c: Make sure that when someone 0's out while
+ recording a msg and then chooses to DELETE the recorded file, the
+ .txt file isn't left around by itself to cause problems later.
+ #7061 (dimitripietro reporting, blitzrage confirmed)
+
+2006-05-01 15:12 +0000 [r23951] Russell Bryant <russell@digium.com>
+
+ * pbx.c: add missing locking of the dialplan functions list in the
+ "show functions" CLI command
+
+2006-05-01 10:45 +0000 [r23305-23899] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_skel.c: fix this to actually compile so people can learn
+ from it
+
+ * cdr/cdr_sqlite.c: eliminate compiler warning
+
+ * channels/chan_iax2.c: remove a pointless comparison, since the
+ buffer is smaller than the length being checked for
+
+ * Makefile, editline/configure, cdr/Makefile, channels/Makefile,
+ db1-ast/Makefile: allow top-level OPTIMIZE setting to affect
+ builds in these subdirectories too
+
+ * Makefile: let the compiler determine whether hardware or software
+ floating point should be used (like we do in the editline
+ subdirectory)
+
+ * Makefile, apps/Makefile: remove extraneous -m64 flag that is not
+ needed remove old 'look' target which is no longer needed (these
+ are coming from Debian patches <G>)
+
+ * editline/makelist: ensure that the script output is correctly
+ generated when the system's character set does not use the
+ English lowercase/uppercase character groups
+
+ * Makefile: do installation in subdirs as a separate target (so
+ external modules can use the Makefile more easily) generate final
+ messages -after- running any post-install script that may be
+ present
+
+2006-04-28 16:40 +0000 [r23176] Russell Bryant <russell@digium.com>
+
+ * configs/zapata.conf.sample, configs/mgcp.conf.sample,
+ configs/sip.conf.sample: note that group assignments must be from
+ 0 to 63 (issue #7048)
+
+2006-04-27 19:11 +0000 [r22954] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_queue.c: Queue(somequeue,,,,) -> interpreted as
+ Queue(somequeue,,,,0) (issue #7044 reported nathan fixed by
+ jsmith - sort of)
+
+2006-04-27 16:12 +0000 [r22866] Matt Frederickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Fix buglet in channel reassignment on
+ SETUP_ACK
+
+2006-04-26 19:18 +0000 [r22596] Matt O'Gorman <mogorman@digium.com>
+
+ * apps/app_voicemail.c: do not allow for users to forward voicemail
+ to themselves, patch from 7001
+
+2006-04-21 22:39 +0000 [r22112-22113] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channel.c: Bug 7004 - release all threads waiting on a condition
+ prior to freeing it
+
+2006-04-19 21:10 +0000 [r21638] Kevin P. Fleming <kpfleming@digium.com>
+
+ * contrib/scripts/safe_asterisk.8, contrib/scripts/safe_asterisk:
+ support system-specific scripts in safe_asterisk, before starting
+ Asterisk proper
+
+2006-04-19 18:43 +0000 [r21597] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * cdr/cdr_odbc.c: Bug 6553 - plug memory leaks when ODBC connection
+ is down
+
+2006-04-18 23:31 +0000 [r21237] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx.c: properly handle brace-wrapped strings in variable/function
+ references in the dialplan
+
+2006-04-18 06:26 +0000 [r20966-21037] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_random.c: Bug 6984 - off by one error in Random()
+
+ * res/res_musiconhold.c: Bug 6544 - when we remove a music class,
+ the thread servicing it should die
+
+2006-04-14 17:21 +0000 [r20034-20037] Kevin P. Fleming <kpfleming@digium.com>
+
+ * sounds.txt: uncomment files that actually do exist (oops)
+
+ * sounds.txt: update text to match actual prompts being distributed
+ (thanks to Kinsey in the support department for reviewing all the
+ prompts!)
+
+2006-04-13 20:37 +0000 [r19891] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 6947 - Allow vm broadcasts to more than
+ 256 characters worth of mailboxes
+
+2006-04-13 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.7.1 released
+
+2006-04-13 17:40 +0000 [r19812] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_page.c: oops... let's not set a variable and then
+ immediately overwrite it while assuming its old value will
+ magically return
+
+2006-04-13 15:56 +0000 [r19768] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * pbx.c: Bug 6957 - variable names beginning with CALLERID weren't
+ substituted correctly
+
+2006-04-12 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.7 released
+
+2006-04-11 22:39 +0000 [r19394-19397] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_dial.c: Bug 6490 - telco intercept should report
+ NOANSWER instead of CHANUNAVAIL
+
+ * apps/app_voicemail.c: Bug 6061 - Fix ODBC storage of VM on PGSQL
+ and MSSQL
+
+2006-04-11 21:58 +0000 [r19353] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: don't create a 'voicemail' symlink in the sounds
+ directory; app_voicemail has not needed it since January of 2005
+ (issue #6613)
+
+2006-04-11 21:55 +0000 [r19351] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * asterisk.c: Bug 6097 - possible descriptor leak
+
+2006-04-11 21:50 +0000 [r19345-19348] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_page.c: don't call the originating device as part of the
+ Page() operation (issue #6932)
+
+ * channel.c: simplify spy queue flushing logic, and always force a
+ flush when one side gets full, even if the other side is not
+ empty (issue #6457)
+
+ * pbx/pbx_config.c: don't destroy the entire dialplan during
+ 'reload', just atomically replace it like 'extensions reload'
+ does (issue #6047)
+
+2006-04-11 20:46 +0000 [r19303] Joshua Colp <joshnet@nbnet.nb.ca>
+
+ * include/asterisk/linkedlists.h: Minor linked lists bug fix. When
+ you're dealing with swapping entries around a lot it can cause a
+ seg fault.
+
+2006-04-11 20:11 +0000 [r19301] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_dial.c: handle call time limit properly when warning is
+ requested _after_ call would hae already ended (issue #6356)
+
+2006-04-11 01:05 +0000 [r18866-19008] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_voicemail.c, app.c: When using the silence detector in
+ ast_play_and_record() and ast_play_and_prepend(), the truncation
+ code never gets called to remove the detected silence, because
+ the value of res is zero when control gets to that point. #6903
+ w/some mods (softins)
+
+ * res/res_features.c: Don't say that we can pass an 'exten'
+ argument in the documentation of Park() when we really cannot.
+ #6902 (opsys)
+
+2006-04-08 19:20 +0000 [r18436-18494] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 6914 - .txt file fails to rename on
+ operator out
+
+ * formats/format_jpeg.c: Bug 6913 - fix for possible buffer
+ overflow
+
+2006-04-07 14:16 +0000 [r18250-18260] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: - Fix cause codes - Add cause code for
+ incompatible formats
+
+ * channels/chan_sip.c: - Fix possible minor memory leak in chan_sip
+ - Return proper cause code on memory allocation error
+
+2006-04-06 22:15 +0000 [r18087-18089] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_meetme.c: fix typo
+
+ * apps/app_meetme.c: small fix... don't try to check conference
+ details if it couldn't be created or found
+
+ * apps/app_meetme.c: don't try to support 'i' or 'r' options if
+ chan_zap is not loaded, and warn the user when they attempt to
+ use them (issue #6675) update application help text to more
+ clearly define when Zaptel and chan_zap are required
+
+2006-04-06 17:24 +0000 [r17945] Russell Bryant <russell@digium.com>
+
+ * apps/app_alarmreceiver.c: move continue out of block that checks
+ verbose level (issue #6880)
+
+2006-04-06 17:00 +0000 [r17702-17905] Joshua Colp <joshnet@nbnet.nb.ca>
+
+ * pbx.c: Unlock channel on failure so that ast_mutex_destroy
+ doesn't throw a fit (issue #6647 reported by casper)
+
+2006-04-05 06:50 +0000 [r17335-17489] Olle Johansson <oej@edvina.net>
+
+ * CREDITS, enum.c: Issue #6654: Enum crash on ADDRESS record,
+ possibly bad record, but still a crash
+
+ * channels/chan_zap.c: Issue #6878 - Unhide DNDstate manager events
+ (thanks casper)
+
+ * apps/app_queue.c: Issue #6882 - move "res=-1" out of verbosity
+ block, minor code cleanups (casper)
+
+2006-04-04 15:24 +0000 [r17283] Matt O'Gorman <mogorman@digium.com>
+
+ * apps/app_senddtmf.c: Adds documentation to show what the w flag.
+ Patch from Ian Kinner at Digium.
+
+2006-04-03 20:38 +0000 [r17074-17150] Olle Johansson <oej@edvina.net>
+
+ * configs/features.conf.sample: Issue 6870 - document that parking
+ lots need to be numeric
+
+ * channels/chan_sip.c: Issue #6848 take two - Use the tag provided
+ by the SUBSCRIBE request when sending NOTIFY
+
+ * channels/chan_sip.c: Ugly patch to avoid hangup causes in
+ non-final responses
+
+2006-03-31 19:11 +0000 [r16744-16771] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: move a NULL check to before the first time
+ the pointer is dereferenced (issue #6832)
+
+ * channels/chan_iax2.c: fix the situation where bindport is
+ specified but bindaddr is not (issue #6616)
+
+2006-03-31 18:24 +0000 [r16742] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx.c: ensure that hint watchers (subscribers) cannot be added or
+ removed while the dialplan is being modified
+
+2006-03-30 22:56 +0000 [r16579-16581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_sip.c: Bug 6853 - Manager fixes: 1) extra ActionID,
+ 2) missing colon
+
+ * asterisk.c: Bug 6849 - trivial typo fix
+
+2006-03-30 21:44 +0000 [r16534-16559] Joshua Colp <joshnet@nbnet.nb.ca>
+
+ * codecs/gsm/Makefile: Add another check for 64-bit goodness (issue
+ #6850 reported by evilbunny)
+
+ * res/res_musiconhold.c: Do not exceed the array size for maximum
+ allowed moh files. (issue #6842)
+
+2006-03-30 01:34 +0000 [r16303-16346] Olle Johansson <oej@edvina.net>
+
+ * res/res_features.c: Set initial value on adsipark
+
+ * apps/app_groupcount.c: Typo fix.
+
+ * configs/extensions.conf.sample: Typo (Issue 6839, casper)
+
+2006-03-29 19:11 +0000 [r16082-16192] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk/pbx.h, apps/app_stack.c, pbx.c: Bug 6830 - Let
+ GosubIf work with the same conditions as a GotoIf (change in API
+ approved by Russell)
+
+ * pbx.c: Bug 6835 - Updates to GotoIf help text
+
+2006-03-29 04:15 +0000 [r16008] Russell Bryant <russell@digium.com>
+
+ * strcompat.c: tell unsetenv for solaris to return the result of
+ the setenv call
+
+2006-03-29 00:58 +0000 [r15898] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #6823 - Portability issue with the
+ registration port number patch from yesterday. Be compatible with
+ more systems than OS/X :-) Thanks Rizzo for the advice.
+
+2006-03-29 00:32 +0000 [r15896] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/linkedlists.h: ensure that list traversal loops
+ which skip entries properly update the 'previous entry' pointer
+ so when entries _are_ removed the list does not get damaged
+
+2006-03-28 20:22 +0000 [r15703-15743] Russell Bryant <russell@digium.com>
+
+ * agi/Makefile, strcompat.c, astmm.c: backport astmm + sparc fixes
+ from the trunk
+
+ * channels/chan_iax2.c: fix Bus Error on sparc (issue #6354)
+
+2006-03-28 19:07 +0000 [r15699] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Fix breakage of NAT support for peers with
+ qualify=yes. Thanks Damin for access to your system, sorry folks.
+
+2006-03-28 18:09 +0000 [r15658] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_ael.c: fix the order in which for loops are expanded
+ (issue #6810)
+
+2006-03-28 17:48 +0000 [r15615] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * contrib/init.d/rc.redhat.asterisk: Bug 6815 - Adding quotes to
+ make bash happy
+
+2006-03-27 23:45 +0000 [r15366-15381] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #6736 - Use flags for OPTION messages.
+ Thanks Casper!
+
+ * channels/chan_sip.c: Issue #6597 - sip show registry shows
+ incorrect port
+
+ * channels/chan_sip.c: Issue #6409 - Use "s" extension when there's
+ no username in the URI
+
+2006-03-26 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.6 released
+
+2006-03-25 05:07 +0000 [r14821-14868] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * contrib/init.d/rc.redhat.asterisk: Bug 6601 - More configuration
+ abilities for the RH init script
+
+ * apps/app_voicemail.c: Fix incorrect size of zeroing (left over
+ from when maxmsg was hardcoded at 100)
+
+ * apps/app_voicemail.c: Bug 6783 - When context is specified,
+ voicemail should look for mailboxes in that context
+
+2006-03-24 14:48 +0000 [r14704] Russell Bryant <russell@digium.com>
+
+ * image.c: use the correct variable in an error message (issue
+ #6791)
+
+2006-03-24 04:53 +0000 [r14610-14659] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_voicemail.c: Fix a typo in the app description
+
+ * include/asterisk/sched.h: Doxygen comment typo corrections
+
+2006-03-23 21:51 +0000 [r14523] Joshua Colp <joshnet@nbnet.nb.ca>
+
+ * res/res_features.c: Issue #6764 - Return BUSY signal when other
+ party is busy at Attended Transfer (Reported by mnachev)
+
+2006-03-23 21:44 +0000 [r14522] Matt Frederickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Fix SETUP_ACK handling so that we change
+ channels if so requested
+
+2006-03-23 20:43 +0000 [r14467] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_meetme.c: Bug #5884 - fix a possible race state in
+ app_meetme when a channel has gone away and we are reading
+ continuously for more frames. (mneuhauser)
+
+2006-03-23 20:13 +0000 [r14462] Russell Bryant <russell@digium.com>
+
+ * apps/app_readfile.c: don't crash when asked to read from a file
+ that doesn't exist (issue #6786)
+
+2006-03-22 22:18 +0000 [r14191-14276] Joshua Colp <joshnet@nbnet.nb.ca>
+
+ * apps/app_voicemail.c: Fix a minor code issue
+
+ * apps/app_voicemail.c: Issue #6781 - Verbose levels not enforced
+ in app_voicemail (Reported by flobi)
+
+ * include/asterisk/cdr.h, cdr.c: Issue #5918 - Disposition showing
+ FAILED even though call is answered successfully (Reported by
+ tracinet)
+
+ * pbx.c: Issue #6780 - ast_pbx_outgoing_cdr_failed description fix.
+ (Reported and fixed by casper)
+
+2006-03-22 09:10 +0000 [r14140] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #6766 - fix ;user=phone functionality.
+ (Reported by alein, fix by russell - thanks!)
+
+2006-03-21 18:59 +0000 [r13814-13964] Russell Bryant <russell@digium.com>
+
+ * configs/features.conf.sample: add a note explaining how to set
+ the DYNAMIC_FEATURES variable to allow the use of custom features
+ (issue #6747)
+
+ * res/res_features.c: fix crash when using the ParkAndAnnounce
+ application. When using this application, there will be no peer
+ channel to play the parking announcement to. (issue #6756)
+
+ * funcs/func_strings.c: fix REGEX on strings that contain quotes
+ (issue #6678)
+
+ * sounds.txt: fix spelling of whiskey
+
+ * apps/app_meetme.c: don't add conference participant if the user
+ hangs up while recording their name (issue #6661)
+
+ * sample.call: re-add the Account parameter to the sample call file
+ since it's not really deprecated since the CDR function is no
+ longer built in
+
+2006-03-21 06:24 +0000 [r13707-13748] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 6714 - Workaround to avoid retrieving
+ incomplete voicemail message
+
+ * editline/term.c: Do away with some warnings and fix some
+ indentation
+
+2006-03-20 17:36 +0000 [r13634] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_iax2.c: Do not overwrite ANI if it's set by IE
+ (sendani=yes in the peer)
+
+2006-03-19 09:59 +0000 [r13550] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c: revert the change made in revision 12927 in
+ favor of keeping the original behavior of the option. The
+ documentation has now been updated to reflect the actual
+ behavior. (issue #6523)
+
+2006-03-19 09:25 +0000 [r13547] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Reset global_rtautoclear at sip reload
+
+2006-03-16 20:05 +0000 [r13279] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * ast_expr2.y, ast_expr2.c: Bug 6737 - Fix compile warning on OS X
+
+2006-03-16 17:58 +0000 [r13239] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Issue #6690 - clarify progressinband
+ default setting
+
+2006-03-16 17:42 +0000 [r13237] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: always use the callerid signalling method
+ set in the zt_pvt strucutre as opposed to the last one read from
+ the config file (issue #6734, with mods)
+
+2006-03-16 06:56 +0000 [r13197] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: To quote giant developers: "Oops". Thanks,
+ Tony!
+
+2006-03-15 22:16 +0000 [r13161] Russell Bryant <russell@digium.com>
+
+ * cdr.c: - remove some calculations that will always result in 0 -
+ if a CDR was never started, don't try to calculate a duration and
+ consider it failed
+
+2006-03-15 13:01 +0000 [r13026] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #6728: Remove parameters to Event:
+ header on SUBSCRIBE requests
+
+2006-03-14 18:41 +0000 [r12925-12927] Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c: when using the G() option to Dial, fix sending
+ the called channel to 1 priority beyond what was specified (issue
+ #6523)
+
+ * apps/app_queue.c: fix a problem with not loading realtime queue
+ members by always reloading a realtime queue from the database
+ even if it is found in the list (issue #6680)
+
+2006-03-12 19:26 +0000 [r12646] Russell Bryant <russell@digium.com>
+
+ * pbx.c: add locking to protect the list of global dialplan
+ variables
+
+2006-03-12 17:57 +0000 [r12577] Russell Bryant <russell@digium.com>
+
+ * codecs/gsm/Makefile: fix build on parisc (issue #6704)
+
+2006-03-10 12:13 +0000 [r12477-12495] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #5937 - Make sure SIP CANCEL's are
+ re-transmitted
+
+ * channels/chan_sip.c: Issue #6576 - SIP_CODEC not used for early
+ media (reported by gpapadop73)
+
+2006-03-08 10:51 +0000 [r12458] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue #6657 - Ignore 183 session progress
+ without SDP
+
+2006-03-07 00:05 +0000 [r12161-12195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_sip.c: Bug 6020 - Race condition where packet could
+ be lost if first packet on list is acked
+
+ * editline/np/vis.c, editline/readline.c: Bug 6664 - More fixes for
+ Solaris
+
+2006-03-06 14:23 +0000 [r12036-12072] Olle Johansson <oej@edvina.net>
+
+ * channel.c: Revert earlier change
+
+ * channel.c: Fix for astmm compilation
+
+2006-03-06 02:32 +0000 [r11946] Russell Bryant <russell@digium.com>
+
+ * configs/zapata.conf.sample: fix a typo in the description of the
+ ringtimeout option
+
+2006-03-05 12:40 +0000 [r11849] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Clear page2 flags at reload too
+
+2006-03-04 11:45 +0000 [r11778] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_mixmonitor.c: Substitute variables in the post_process
+ string (if it exists) before those variables could possibly
+ disappear (channel hangup) #6462
+
+2006-03-03 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.5 released
+
+2006-03-03 00:38 +0000 [r11607-11635] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * Makefile: Bug 6638 - Use POSIX command for Solaris
+
+ * build_tools/make_build_h: Bug 6638 - Change from a historic BSD
+ command to a POSIX command for determining username
+
+ * asterisk.c: Bug 6637 - Fixes for Solaris
+
+ * Makefile: If debugging, the frame pointer is helpful
+
+2006-03-02 19:05 +0000 [r11528-11561] Russell Bryant <russell@digium.com>
+
+ * res/res_monitor.c: fix inaccurate ack message to ChangeMonitor
+ action (issue #6630)
+
+ * asterisk.sgml: make the terminology used in the synopsis match
+ the option description
+
+ * asterisk.sgml: add the -L option to the synopsis on the man page
+
+2006-03-01 17:41 +0000 [r11479-11503] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * cdr/cdr_manager.c, cdr/cdr_tds.c, res/res_config_odbc.c,
+ include/asterisk/cdr.h, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, cdr.c:
+ Bug 6615 - Fix 64bit conversion errors by using a long int
+
+ * build_tools/make_svn_branch_name: Bug 6618 - Solaris
+ compatibility fix
+
+2006-02-28 19:46 +0000 [r11382-11410] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: fix the output that indicates whether
+ qualify smoothing is on or not (issue #6608)
+
+ * asterisk.c: adjust the keys directory when astvarlibdir is
+ specified in asterisk.conf (issue #6602)
+
+ * res/res_agi.c: add a missing newline in the agi app description
+ (thanks wunderkin!)
+
+2006-02-27 15:20 +0000 [r11250-11281] Russell Bryant <russell@digium.com>
+
+ * cli.c: don't try to print the help text for a CLI command when
+ RESULT_SHOWUSAGE is returned if there is no help text available
+ (issue #6604)
+
+ * channels/chan_sip.c: fix finding realtime peers that are not
+ dynamic by ip address (issue #6093)
+
+ * channel.c: don't hang up the channel if its state is set to UP
+ before we return from ast_call (issue #6569)
+
+2006-02-26 16:26 +0000 [r11165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk/logger.h, logger.c: Bug 5950 - reenable queue
+ log rotation; also, eliminate redundant code
+
+2006-02-25 19:54 +0000 [r11120] Matt Frederickson <creslin@digium.com>
+
+ * translate.c: Backport of fix to translation optimizations. Thanks
+ again file!
+
+2006-02-25 05:08 +0000 [r11058-11089] Kevin P. Fleming <kpfleming@digium.com>
+
+ * translate.c: factor the number of translation steps required into
+ translation path decisions, so that equal cost paths that require
+ fewer translations are preferred
+
+ * translate.c: reformat code to fit guidelines remember which
+ translation paths are multi-step paths
+
+ * channel.c: ensure that spy frame queueing is able to deal with
+ translation failing for any reason (issue #6546)
+
+2006-02-23 23:06 +0000 [r10952] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * Makefile: set PWD properly
+
+2006-02-23 14:57 +0000 [r10736-10863] Kevin P. Fleming <kpfleming@digium.com>
+
+ * dnsmgr.c, include/asterisk/linkedlists.h: backport list handling
+ fix from trunk (solves memory leak problem in cdr variables and
+ device state watchers) remove unused variable to silence
+ compiler warning
+
+ * configs/iax.conf.sample: add comment warning people about trying
+ to use hostnames/IPs in the sample config
+
+2006-02-20 23:01 +0000 [r10577] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * app.c: Would be nice to tell people to look in the right file to
+ increase a constant
+
+2006-02-20 06:17 +0000 [r10511-10535] Mark Spencer <markster@digium.com>
+
+ * channels/chan_sip.c: Handle ACKing properly (remove gratuitous
+ -1)
+
+ * channels/chan_iax2.c: Fix numerous places in jitter buffer where
+ freed memory is referenced
+
+2006-02-19 18:29 +0000 [r10462-10487] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * formats/format_sln.c: Okay, fseek doesn't return an offset
+
+ * apps/app_voicemail.c: Fix possible lack of initialization of
+ useadsi
+
+ * formats/format_sln.c: Bug 6539 - Division by two negates error
+ flag
+
+2006-02-18 00:17 +0000 [r10409] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * app.c: Bug 6529 - memory leak in ast_play_and_prepend
+
+2006-02-17 01:55 +0000 [r10301-10368] Russell Bryant <russell@digium.com>
+
+ * jitterbuf.c: fix incorrent index calculation for jitterbuffer
+ history (issue #6517)
+
+ * apps/app_voicemail.c: when executing the Directory application
+ from voicemail and a context is not specified, use the "default"
+ context, not the channel's current context (issue #6507)
+
+2006-02-15 01:21 +0000 [r10108-10137] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_agent.c: ensure that agents logged in via the
+ manager interface are stored in the persistence database (related
+ to issue #6301)
+
+ * funcs/func_enum.c: handle longer ENUM lookup results (issue
+ #6476)
+
+ * res/res_agi.c: ensure that FastAGI launcher can handle system
+ call interruption (issue #6449)
+
+2006-02-14 20:56 +0000 [r10021] Matt O'Gorman <mogorman@digium.com>
+
+ * apps/app_meetme.c: bug fix from 6485 with musiconhold not being
+ turned off by app_meetme
+
+2006-02-14 20:20 +0000 [r10018] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_queue.c: don't double-increment abandon counter for
+ calls that are hung up while dialing members (issue #6289)
+
+2006-02-14 19:11 +0000 [r9990] Mark Spencer <markster@digium.com>
+
+ * apps/app_meetme.c: Fix stopstream in menus (bug #6137)
+
+2006-02-14 18:50 +0000 [r9961-9964] BJ Weschke <bweschke@btwtech.com>
+
+ * asterisk.c: #ifdef the include too.
+
+ * asterisk.c: #ifdef'd the prctl fix to only try and compile on
+ linux systems. Thanks rizzo for pointing this out.
+
+2006-02-14 18:30 +0000 [r9953-9958] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: when answering INVITE, don't send codecs the
+ peer didn't offer (issue #6052)
+
+ * rtp.c: revert yesterday's temporary fix for issue #6052
+
+2006-02-14 04:45 +0000 [r9861-9870] BJ Weschke <bweschke@btwtech.com>
+
+ * asterisk.c: Fixed my silly backport error from r9861
+
+ * asterisk.c: Merged changes from r9844 from /trunk. Make sure that
+ PR_SET_DUMPABLE is set to make certain that we still dump core if
+ Asterisk has setuid'd to run as non-root.
+
+2006-02-14 00:46 +0000 [r9818] Kevin P. Fleming <kpfleming@digium.com>
+
+ * rtp.c: don't try to use peer's dynamic codec numbers, it leads to
+ duplication (issue #6052)
+
+2006-02-13 17:37 +0000 [r9756] Josh Roberson <josh@asteriasgi.com>
+
+ * apps/app_meetme.c: Don't set the formats before we stop
+ indications. (issue #6380)
+
+2006-02-11 19:23 +0000 [r9581-9609] Russell Bryant <russell@digium.com>
+
+ * channels/chan_mgcp.c, channels/chan_sip.c, pbx/pbx_dundi.c,
+ channels/chan_iax2.c: fix memory leak from not destroying the
+ scheduler context on module unload
+
+ * apps/app_page.c: fix due to CDR changes
+
+ * manager.c, pbx/pbx_spool.c, include/asterisk/channel.h,
+ include/asterisk/pbx.h, include/asterisk/manager.h, channel.c,
+ pbx.c: now that CDR is a loadable module, don't depend on it
+ elsewhere (issue #6460)
+
+2006-02-11 15:22 +0000 [r9528] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c, cdr.c: clean up my mess from thread-starting
+ change
+
+2006-02-11 06:29 +0000 [r9493] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_sip.c: kpfleming's fix from r9472 backported to 1.2
+
+2006-02-10 20:38 +0000 [r9404] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_mgcp.c, dnsmgr.c, channels/chan_sip.c,
+ devicestate.c, channels/chan_modem.c, cdr.c: don't create monitor
+ threads in detached mode, when we need to be able to
+ pthread_join() them later if the module is unloaded (solve
+ crash-on-unload problem for these channel modules)
+
+2006-02-09 21:10 +0000 [r9323-9326] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Revert behavior change from previous commit
+ (fixes only)
+
+ * apps/app_voicemail.c: Backport 5929 to 1.2
+
+2006-02-09 02:31 +0000 [r9246-9262] Russell Bryant <russell@digium.com>
+
+ * apps/Makefile: add another location for postgresql headers (issue
+ #6419)
+
+ * channels/chan_iax2.c: reload peercontext on iax2 reload (issue
+ #6442)
+
+2006-02-08 22:34 +0000 [r9233] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * cdr/Makefile: Leave it to RH/CentOS to put the freetds headers in
+ a completely nonstandard location.
+
+2006-02-08 22:12 +0000 [r9232] Matt O'Gorman <mogorman@digium.com>
+
+ * logger.c, channels/chan_oss.c: Make logger report
+ error,warning,notice if logger.conf not found, also updated
+ chan_oss to give correct error message if its config file is not
+ found.
+
+2006-02-05 17:10 +0000 [r9156] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_macro.c: Bug 6176 - Fix race condition
+
+2006-02-02 18:37 +0000 [r9086] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: don't override ASTERISKVERSIONNUM to 000000 for non-svn
+ builds
+
+2006-02-02 16:12 +0000 [r9073] Matt Frederickson <creslin@digium.com>
+
+ * res/res_odbc.c: Fix for (#6309), potential (highly unlikely)
+ memory leak in res_odbc
+
+2006-01-30 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.4 Released
+
+2006-01-30 17:08 +0000 [r8905] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: disable buggy PRI user-user code until it
+ can be fixed
+
+2006-01-28 13:52 +0000 [r8808] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 6182 - Don't remove scheduled event
+ until it's really done. (reported by malverian)
+
+2006-01-27 08:02 +0000 [r8785] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 6362 - Register without Contact: and
+ Expires: fails (reporter: op)
+
+2006-01-27 00:52 +0000 [r8758] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * ast_expr2.h, ast_expr2f.c, ast_expr2.c: Bug 6072 - Revisions to
+ the source bison and flex files don't auto-regenerate these files
+
+2006-01-26 19:42 +0000 [r8729] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: fix problem with dtmf on e&m (issue #6364)
+
+2006-01-26 14:39 +0000 [r8710] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 5898: Registrations does not get
+ deleted if there's an active SIP dialog
+
+2006-01-25 19:14 +0000 [r8666-8677] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: don't call ast_update_realtime with
+ uninitialized variables if we get a registration with an expirey
+ of 0 seconds (issue #6173)
+
+ * channels/chan_features.c: fix memory leak (inspired by issue
+ #6351)
+
+2006-01-25 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.3 Released
+
+2006-01-25 09:46 +0000 [r8632] Olle Johansson <oej@edvina.net>
+
+ * channel.c: Issue #6439 - the "timebomb" bug. Patch by Markster
+ over GPRS
+
+2006-01-25 05:38 +0000 [r8619] Russell Bryant <russell@digium.com>
+
+ * utils/astman.c: don't leak almost 200 bytes for each new channel
+ (issue #6330)
+
+2006-01-25 01:50 +0000 [r8608] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_dial.c: ensure hangup cause code is handled properly
+ when channel does not return a frame (issue #6346)
+
+2006-01-24 22:55 +0000 [r8600] Russell Bryant <russell@digium.com>
+
+ * asterisk.c: completely arbitrary whitespace change for testing
+ something with svnmerge ...
+
+2006-01-24 22:32 +0000 [r8588] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channel.c: ensure that channel cannot become zombie after we
+ check but before we try to start indications
+
+2006-01-24 20:37 +0000 [r8573] Matt Frederickson <creslin@digium.com>
+
+ * channels/chan_zap.c: Backport fix for #6229, hangup on polarity
+ reversal
+
+2006-01-24 19:21 +0000 [r8537-8562] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Issue 6114: Don't hangup on BYE/ALSO with no
+ channel.
+
+ * channels/chan_sip.c: Issue #6308 - never send response to ACK.
+ (Reported by whiskerp)
+
+2006-01-22 19:03 +0000 [r8437-8445] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: fix memory leak from not freeing the queue
+ member list when freeing an old queue
+
+ * channel.c: fix MixMonitor crash (issue #6321, probably others)
+
+2006-01-22 15:13 +0000 [r8433] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_sip.c: Bug fix: Correct some scenarios where
+ CALL_LIMIT could not be getting adjusted properly allowing
+ chan_sip to send calls when it really shouldn't. Bug #6111
+
+2006-01-22 08:52 +0000 [r8429] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_sip.c: Bug 6281 - Cannot set more than a single
+ header with SIPAddHeader
+
+2006-01-22 02:05 +0000 [r8412-8418] Russell Bryant <russell@digium.com>
+
+ * pbx.c: add a modified fix to prevent writing outside of the
+ provided workspace when calculating a substring (issue #6271)
+
+ * pbx.c: temporarily revert substring fix pending the result of the
+ discussion in issue #6271
+
+ * pbx.c: prevent the possibility of writing outside of the
+ available workspace (issue #6271)
+
+2006-01-21 18:29 +0000 [r8394] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_queue.c: Bug 5936 - AddQueueMember fails on realtime
+ queue, if queue not yet loaded
+
+2006-01-20 18:34 +0000 [r8347] Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: fix invalid value of prev_q (issue #6302)
+
+2006-01-20 01:00 +0000 [r8320] Matt O'Gorman <mogorman@digium.com>
+
+ * channels/chan_iax2.c: solved problem with delayreject and iax
+ trunking bug 4291
+
+2006-01-19 19:40 +0000 [r8281] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Enable "musicclass" setting for sip peers as
+ per the config sample.
+
+2006-01-19 19:14 +0000 [r8276] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * ast_expr2.y, ast_expr2.fl: Bug 6072 - Memory leaks in the
+ expression parser
+
+2006-01-19 04:56 +0000 [r8232-8242] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: fix Message-Account header to use the ip
+ address if the fromdomain isn't set (issue #6278)
+
+ * apps/app_milliwatt.c: fix a seg fault due to assuming that space
+ gets allocatted on the stack in the same order that we declare
+ the variables (issue #6290)
+
+2006-01-18 21:02 +0000 [r8194] Matt O'Gorman <mogorman@digium.com>
+
+ * apps/app_meetme.c: Solves issue with the login proccess in meetme
+ patch from 6136
+
+2006-01-18 02:49 +0000 [r8173] Russell Bryant <russell@digium.com>
+
+ * ChangeLog (removed): remove ChangeLog from the 1.2 branch. It
+ will only be present in the tags.
+
+2006-01-18 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.2.2 Released
+
+2006-01-18 00:47 +0000 [r8140-8162] Matt O'Gorman <mogorman@digium.com>
+
+ * loader.c: Changed order of autoload so that pbx_ comes before
+ channels, and in doing so cause bug 6002 to not be an issue
+
+ * apps/app_festival.c: Stop any generators running on a channel
+ when festival is called as described in 5996
+
+2006-01-17 18:29 +0000 [r8134] Matt Frederickson <creslin@digium.com>
+
+ * res/res_features.c: Backport of fix for #6094
+
+2006-01-17 16:55 +0000 [r8124] Matt O'Gorman <mogorman@digium.com>
+
+ * logger.c: Fixed code ordering of logger_init and queue_log_init
+ bug 6263
+
+2006-01-17 13:11 +0000 [r8112-8122] Kevin P. Fleming <kpfleming@digium.com>
+
+ * asterisk.c: update CLI copyright notice
+
+ * asterisk.c: do rlimit check _after_ reading config file, in case
+ 'dumpcore' is specified there
+
+2006-01-14 19:06 +0000 [r8074] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * funcs/func_strings.c: Bug 6238 - Fix segfault when delimiter not
+ specified
+
+2006-01-13 06:07 +0000 [r8047] Russell Bryant <russell@digium.com>
+
+ * channels/chan_agent.c: fix spelling errors (issue #6227)
+
+2006-01-12 06:14 +0000 [r7999] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c, configs/voicemail.conf.sample: Bug 6211 -
+ Add option deletevoicemail as equivalent to option delete for
+ Realtime
+
+2006-01-11 19:08 +0000 [r7965-7986] Russell Bryant <russell@digium.com>
+
+ * channels/chan_agent.c: move variable to correct scope (issue
+ #6197)
+
+ * apps/app_voicemail.c: fix temp greetings with ODBC storage (issue
+ #6078)
+
+ * channels/chan_sip.c: fix mem leak on module unload (issue #6190)
+
+ * app.c: don't override an error condition that occurred when
+ acting on the primary channel when stopping the autoservice on
+ the peer channel. (from issue #6087)
+
+ * translate.c: lock list of translators *before* recalculating the
+ translation matrix
+
+2006-01-11 04:38 +0000 [r7963] Matt O'Gorman <mogorman@digium.com>
+
+ * channel.c: Minor typo refrenced in 6191
+
+2006-01-11 04:19 +0000 [r7957-7960] Russell Bryant <russell@digium.com>
+
+ * pbx.c: fix locking error - lock instead of unlock
+
+ * apps/app_dial.c: fix a little typo
+
+2006-01-11 01:30 +0000 [r7955] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 6192 - behave correctly when mailbox is
+ specified as argument
+
+2006-01-10 08:48 +0000 [r7939] Olle Johansson <oej@edvina.net>
+
+ * doc/README.cdr: - Adding reference to README.tds - Reformatting
+ table
+
+2006-01-09 22:48 +0000 [r7917] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: re-initialize _all_ sequence numbers when
+ transfer completes
+
+2006-01-09 22:07 +0000 [r7915] Russell Bryant <russell@digium.com>
+
+ * file.c: add missing unlock (issue #6112)
+
+2006-01-09 20:08 +0000 [r7904-7908] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * pbx/pbx_spool.c: Bug 6157 - Memory leak
+
+ * doc/README.variables: Update variable documentation to match the
+ code
+
+2006-01-09 18:11 +0000 [r7898-7900] Kevin P. Fleming <kpfleming@digium.com>
+
+ * asterisk.c: commit user/group-related changes from trunk
+
+ * db.c: backport fix from revision 7856 of trunk
+
+ * apps/app_voicemail.c: fix breakage introduced in revision 7871
+
+2006-01-09 05:11 +0000 [r7870-7871] Russell Bryant <russell@digium.com>
+
+ * apps/app_voicemail.c: fix seg fault when using greek syntax in
+ VoicemMailMain (issue #6142)
+
+ * manager.c: backport fix for unnecessary unlock (issue #6171)
+
+2006-01-07 07:27 +0000 [r7848] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * pbx/pbx_spool.c: Bug 6156 - catch all threading errors, not just
+ simple failure
+
+2006-01-06 00:34 +0000 [r7831] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * pbx/pbx_config.c: Dumb error messages - "Context 'context'
+ already included in 'in' context"
+
+2006-01-06 00:21 +0000 [r7829] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_agent.c: update agent persistence when an agent
+ gets logged off by autologoff
+
+2006-01-05 23:53 +0000 [r7827] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk/strings.h: Bug 6076 - Fix documentation of
+ ast_trim_blank return value
+
+2006-01-05 23:49 +0000 [r7825] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channel.c: eliminate rounding errors that caused call time limits
+ to be inaccurate (issue #5913) round 'time left' reported during
+ call limit warnings up to sound more accurate
+
+2006-01-05 23:07 +0000 [r7823] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * res/res_features.c: Bug 6081 - fix for memory leak, formatting
+ fixes
+
+2006-01-05 20:52 +0000 [r7819] Kevin P. Fleming <kpfleming@digium.com>
+
+ * formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that
+ variable is initialized
+
+2006-01-05 09:13 +0000 [r7812] Olle Johansson <oej@edvina.net>
+
+ * res/res_features.c: Fix copyright of changed file
+
+2006-01-05 00:58 +0000 [r7799-7809] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_agent.c: send device state updates for auto-logoff
+ of agents as well
+
+ * formats/format_pcm.c, formats/format_pcm_alaw.c: doh... fseek()
+ has no useful return value
+
+ * formats/format_pcm.c, formats/format_pcm_alaw.c: use proper
+ fwrite() parameters and return value
+
+ * formats/format_pcm.c, formats/format_pcm_alaw.c: return properly
+ after extending file
+
+ * formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that
+ ulaw/alaw sound files are filled with silence when extended (not
+ zeroes)
+
+ * channel.c: make monitoring more tolerant of peers that deliver
+ frames in bursts
+
+2006-01-04 21:46 +0000 [r7792-7795] Olle Johansson <oej@edvina.net>
+
+ * res/res_features.c: Issue #5980: Removing extra CR+LF in manager
+ events - needs port to trunk
+
+ * channels/chan_sip.c: Fixing typo in XML for video updates.
+
+2006-01-04 07:06 +0000 [r7773] Russell Bryant <russell@digium.com>
+
+ * funcs/func_moh.c: use a more correct way of determining the size
+ of the destination buffer
+
+2006-01-04 05:27 +0000 [r7771] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_privacy.c: Fix the 'if' clause to be true under the
+ right conditions. Bug #6126
+
+2006-01-03 20:22 +0000 [r7746] Kevin P. Fleming <kpfleming@digium.com>
+
+ * ast_expr.y (removed): remove unused 'old' expression parser
+
+2006-01-03 18:15 +0000 [r7743] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_stack.c: Bug 6121 - typo in application description
+
+2006-01-03 17:24 +0000 [r7736-7740] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/chanspy.h, apps/app_chanspy.c,
+ apps/app_mixmonitor.c, channel.c: revert incorrect fix for bug
+ #6048 from revision 7709 put in correct (simpler) fix add doxygen
+ docs for channel spy 'state' values
+
+ * channels/chan_sip.c: backport rport scanning fix from trunk (bug
+ #6071)
+
+ * ast_expr2f.c, ast_expr2.fl: don't leak memory for (most)
+ expression evaluations
+
+2006-01-02 07:31 +0000 [r7709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_mixmonitor.c: Bug 6084 - MixMonitor after a 'cli stop
+ monitor' deadlocks
+
+2006-01-02 02:04 +0000 [r7706] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_sip.c, channels/chan_iax2.c: Fix compiler warnings.
+
+2005-12-30 14:54 +0000 [r7677] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channel.c: Bug 6091 - Fix race condition around uniqueid
+
+2005-12-28 17:35 +0000 [r7663-7665] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: fix memory leak in build_rpid (issue #6070)
+
+ * apps/app_chanspy.c: backport fix for permissions of created
+ recordings (issue #6067)
+
+2005-12-27 00:07 +0000 [r7641] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: backport fix to ensure that DSP is never
+ enabled on pseudo channels
+
+2005-12-26 20:32 +0000 [r7637] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * cdr/cdr_tds.c: Remove copy of code in libc, preferring code in
+ utils.c (public domain code)
+
+2005-12-26 18:19 +0000 [r7634] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c, channels/chan_agent.c, apps/app_sms.c,
+ asterisk.c, config.c, pbx/pbx_dundi.c, apps/app_externalivr.c,
+ apps/app_queue.c, channels/chan_iax2.c, cli.c,
+ apps/app_chanspy.c, res/res_monitor.c: cast time_t to an int in
+ printf/scanf (issue #5635)
+
+2005-12-23 06:38 +0000 [r7608] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_hasnewvoicemail.c: Bug 6051 - VMCOUNT should work as
+ documented and count all, not quit after finding 1
+
+2005-12-23 03:01 +0000 [r7606] Kevin P. Fleming <kpfleming@digium.com>
+
+ * asterisk.c: add license reference to copyright notice displayed
+ when CLI session begins add 'show warranty' and 'show license'
+ CLI commands (still need a complete list of non-GPL components
+ included in Asterisk)
+
+2005-12-23 00:00 +0000 [r7605] BJ Weschke <bweschke@btwtech.com>
+
+ * apps/app_waitforsilence.c: Another app documentation tweak.
+
+2005-12-22 22:04 +0000 [r7601] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 6050 SQL requires the use of single
+ ticks to delimit values, not quotes
+
+2005-12-22 20:36 +0000 [r7595-7599] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: revert changes to
+ videosupport to allow per-peer setting, since it isn't quite
+ complete and there is not an obvious fix at this point
+
+ * channels/chan_sip.c: remove stray unlock (issue #5955)
+
+2005-12-21 22:23 +0000 [r7586] Josh Roberson <josh@asteriasgi.com>
+
+ * channels/chan_sip.c: Actually put in the per-peer settings for
+ sip video, as they didn't make it in at astricon somehow, and
+ I've been too busy up until now to redo it.
+
+2005-12-21 20:01 +0000 [r7582] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * channels/chan_alsa.c: Allow a chan_alsa that failed to open sound
+ devices to be unloaded.
+
+2005-12-21 19:53 +0000 [r7580] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_agent.c: Bug #6040 - Documentation correction
+
+2005-12-21 19:23 +0000 [r7577] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * pbx/pbx_ael.c: Bug 5777 - Remove parentheses on Goto in AEL, so
+ that it parses correctly
+
+2005-12-20 20:21 +0000 [r7550-7557] Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: check array bounds when parsing arguments to AGI
+ (issue #5868)
+
+ * channels/chan_iax2.c: backport fix for reloading peer context
+ (issue #6007)
+
+ * apps/app_directed_pickup.c: backport fix for segfault on directed
+ pickup when no CDR is available (issue #5998)
+
+2005-12-20 12:58 +0000 [r7546] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_meetme.c: backport fix for larger-than-20ms-frames from
+ trunk (bug #5697)
+
+2005-12-19 23:47 +0000 [r7529] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: I messed up and accidently committed this to
+ the trunk first ... - add note on required values of sip_methods
+ struct - remove duplicate function prototype - remove duplicate
+ ast_mutex_lock (issue #6025)
+
+2005-12-19 19:06 +0000 [r7521-7523] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * file.c: Bug 5988 - record append option not working
+
+ * cdr.c: Bug 6026 - segfault for the sequence NoCDR(),
+ SetAMAFlags()
+
+2005-12-17 18:55 +0000 [r7517-7519] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * doc/README.ael: Document that curley braces must be on the same
+ line as the keyword.
+
+ * apps/app_chanspy.c: Bug 6009 - off by one error
+
+2005-12-17 03:59 +0000 [r7510-7515] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: Max-Forwards headers must only be present on
+ requests, not responses
+
+ * channels/chan_sip.c: forcibly expire previous subscriptions from
+ a peer when they resubscribe (keeps them from building up and
+ waiting for expiration, and stops us sending unwanted NOTIFY
+ messages to devices)
+
+ * build_tools/make_svn_branch_name: fix some buglet when building
+ team branch version strings
+
+2005-12-17 01:02 +0000 [r7508] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk/linkedlists.h: We want to check the previous
+ value, not the current value (which was just changed).
+
+2005-12-16 00:49 +0000 [r7497] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_cut.c: First field is truncated
+
+2005-12-15 10:52 +0000 [r7490] Christian Richter <christian.richter@beronet.com>
+
+ * doc/README.misdn, channels/misdn/mISDNuser.patch (added),
+ channels/misdn/isdn_lib_intern.h, channels/misdn/mISDN.patch
+ (added), channels/misdn/isdn_lib.h, channels/chan_misdn.c,
+ channels/misdn/Makefile, channels/misdn/chan_misdn_config.h,
+ channels/misdn/ie.c, channels/chan_misdn_config.c,
+ channels/misdn/isdn_msg_parser.c, channels/Makefile,
+ channels/misdn/isdn_lib.c: * Added mISDN/mISDNuser Echo cancel
+ Patch * Fixed Makefiles so that chan_misdn can be compiled again
+ * added some hints, that mISDN cannot be compiled against gcc-4,
+ SMP, Spinlock Debug * fixed some Minor issues in chan_misdn,
+ regarding Type Of Number and Presentation
+
+2005-12-15 02:51 +0000 [r7482] BJ Weschke <bweschke@btwtech.com>
+
+ * channel.c: Bug #6003 - Don't free the channel structure until
+ after having sent the manager event.
+
+2005-12-13 18:54 +0000 [r7435-7470] Kevin P. Fleming <kpfleming@digium.com>
+
+ * doc/README.variables: clarify substring documentation
+
+ * utils.c: correct broken math in tvfix() for timestamp values over
+ one million
+
+ * apps/app_dial.c: restore ability of caller to hangup calls that
+ are still ringing (issue #5839)
+
+ * channels/chan_sip.c, pbx.c: ensure that hangups while incoming
+ calls are in early state are handled properly (issue #5919)
+
+ * channels/chan_agent.c: only report AGENT_IDLE for callback mode
+ agents when they are actually idle (issue #5902)
+
+ * app.c: use the stream's current point when pausing/unpausing,
+ instead of elapsed time (which doesn't work when the stream has
+ been skipped forward or backward) (issue #5897)
+
+ * apps/app_externalivr.c: set all the child file descriptors to
+ non-blocking so that we don't hang if the child fails to send a
+ newline-terminated command or error message
+
+2005-12-12 17:19 +0000 [r7433] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk/linkedlists.h: Typo
+
+2005-12-11 06:08 +0000 [r7430] Russell Bryant <russell@digium.com>
+
+ * utils/astman.c: silence a couple of compiler warnings about
+ pointer signedness
+
+2005-12-11 01:26 +0000 [r7427-7429] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * include/asterisk/linkedlists.h: Bug 5965 - major bug in
+ AST_LIST_REMOVE
+
+ * apps/app_voicemail.c: Bug 5967
+
+2005-12-10 18:10 +0000 [r7425] BJ Weschke <bweschke@btwtech.com>
+
+ * channels/chan_zap.c: Bug #5877 Make sure the digit string from
+ E&M wink DNIS collection is properly null terminated as it grows.
+
+2005-12-08 23:45 +0000 [r7404-7406] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 5960
+
+ * configs/res_odbc.conf.sample: Documenting two keywords that were
+ previously missing
+
+2005-12-08 01:05 +0000 [r7382-7386] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx.c: initialize the buffer before using it...
+
+ * pbx.c: ensure that hints are allowed to use global variable
+ references
+
+2005-12-06 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.1 Released
+
+2005-12-05 06:47 +0000 [r7335-7340] Russell Bryant <russell@digium.com>
+
+ * Makefile: remove ASTERISKVERSIONNUM from the version string given
+ to doxygen
+
+ * apps/app_queue.c: don't delete dynamic queue members when
+ reloading the static members from a realtime database (issue
+ #5922)
+
+ * channels/chan_sip.c: fix the order of arguments to an error
+ message (issue #5927)
+
+2005-12-04 18:03 +0000 [r7329] Kevin P. Fleming <kpfleming@digium.com>
+
+ * build_tools/make_svn_branch_name: use a more efficient way to get
+ the revision number, that will also report if the working copy
+ contains uncommitted modifications
+
+2005-12-03 19:55 +0000 [r7310] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 5925: check for "Unknown", as that's
+ what app_voicemail puts into the field for Unknown callerid Also,
+ remove useless res checks (initialized to 0; never set)
+
+2005-12-03 01:24 +0000 [r7299] Olle Johansson <oej@edvina.net>
+
+ * configs/sip.conf.sample: Documenting the default registerattempts
+ setting as 0, continue hammering the server for ever and ever ;-)
+
+2005-12-02 21:12 +0000 [r7285] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
+
+ * contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.mandrake.asterisk,
+ contrib/init.d/rc.redhat.asterisk,
+ contrib/init.d/rc.gentoo.asterisk,
+ contrib/init.d/rc.mandrake.zaptel,
+ contrib/init.d/rc.slackware.asterisk: Turn on executable bits for
+ startup scripts, and fix bash var interpolation for Mandrake
+
+2005-12-02 00:52 +0000 [r7275] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Bug #5907. Improve SIP INFO DTMF debugging
+ output. (1.2 & Trunk)
+
+2005-12-02 00:51 +0000 [r7266-7274] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_page.c, pbx.c: inherit channel variables into channels
+ created by Page() application (issue #5888)
+
+ * apps/app_voicemail.c, configs/voicemail.conf.sample, UPGRADE.txt:
+ allow previous context-searching behavior to be used if desired
+ (issue #5899)
+
+ * apps/app_voicemail.c: properly handle password changes when
+ mailbox is last line of config file and not followed by a newline
+ (issue #5870) reformat password changing code to conform to
+ coding guidelines (issue #5870)
+
+ * channels/chan_agent.c: protect agent_bridgedchannel() from
+ segfaulting when there is no bridged channel (issue #5879)
+
+ * channels/chan_local.c: allow variables to exist on both 'halves'
+ of the Local channel (issue #5810)
+
+ * apps/app_festival.c: don't block waiting for the Festival server
+ forever when it goes away (issue #5882)
+
+ * channel.c: ensure channel's scheduling context is freed (issue
+ #5788)
+
+ * Makefile, patches (removed): Makefile 'update' target now
+ supports updating from Subversion repositories (issue #5875)
+ remove support for 'patches' subdirectory, it's no longer useful
+
+2005-12-01 23:18 +0000 [r7261-7265] Olle Johansson <oej@edvina.net>
+
+ * doc/README.misdn: Changing bug report address to the Asterisk
+ issue tracker
+
+ * doc/README.jitterbuffer, doc/README.realtime: Removing references
+ to 1.1dev, replacing with 1.2, in documentation files.
+
+ * doc/README.misdn: Fixing some spelling errors, as well as
+ changing "cvs" to "subversion" in misdn documentation.
+
+2005-12-01 19:25 +0000 [r7257] Kevin P. Fleming <kpfleming@digium.com>
+
+ * build_tools/make_svn_branch_name: ensure that 'svn info' output
+ is in the expected language for the script to parse (issue #5880)
+
+2005-12-01 02:33 +0000 [r7228-7251] Russell Bryant <russell@digium.com>
+
+ * apps/app_externalivr.c: use ast_app_separate_args to split
+ arguments (issue #5686)
+
+ * apps/app_queue.c: fix queue weight feature - compare member
+ interfaces instead of pointers to the members, since each queue
+ has its own list of members. (issue #5863)
+
+ * build_tools/make_svn_branch_name: use '=' instead of '==' for
+ string comparisons. /bin/bash is ok with this, but /bin/sh is
+ not. (issue #5885)
+
+ * redhat/asterisk (removed), Makefile: remove outdated redhat init
+ script and provide the updated one in 'make rpm' (issue #5786)
+
+ * contrib/init.d/rc.debian.asterisk,
+ contrib/init.d/rc.redhat.asterisk: Comment out LD_ASSUME_KERNEL
+ by default. Print error messages if the asterisk executable or
+ the asterisk configuration directory are not found. (issue #5785,
+ #5708)
+
+ * apps/app_dial.c: fix DIALEDTIME when call has not been answered
+ (issue #5862)
+
+ * rtp.c: do not allow an rtp message with zero type (issue #5749)
+
+ * pbx.c: fix hint case sensitivity (issue #5856)
+
+ * configs/sip.conf.sample: add description of the "fromdomain"
+ option (issue #5874)
+
+2005-11-30 03:52 +0000 [r7227] Josh Roberson <josh@asteriasgi.com>
+
+ * apps/app_voicemail.c, UPGRADE.txt, ChangeLog: backport fix from
+ trunk
+
+2005-11-30 03:37 +0000 [r7219-7226] Kevin P. Fleming <kpfleming@digium.com>
+
+ * doc/cdr.txt, doc/CODING-GUIDELINES, include/asterisk.h,
+ doc/README.mp3: remove remaining CVS references
+
+ * channel.c: port memory leak fix from rev 7223 in trunk
+
+ * include/asterisk/lock.h: do the multiple-lock check for cond_wait
+ properly...
+
+2005-11-29 06:12 +0000 [r7216-7218] Russell Bryant <russell@digium.com>
+
+ * apps/app_cut.c: print an error message if invalid arguments are
+ specified
+
+ * apps/app_skel.c: fix a couple of typos and a buglet
+
+2005-11-29 01:25 +0000 [r7199-7213] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/lock.h: if the lock protected a pthread_cond is
+ held recursively, warn before waiting onthe condition
+
+ * Makefile, build_tools/make_svn_branch_name (added): port version
+ string computation from trunk
+
+ * / (added): branch renames remove unneeded branches
+
+2005-11-29 Josh Roberson <josh@asteriasgi.com>
+
+ * apps/app_voicemail.c: Only look in 'default' context when no context defined to VoiceMailMain(). (issue #5887)
+
+2005-11-25 Russell Bryant <russell@digium.com>
+
+ * apps/app_dial.c: Properly duplicate the string for ANI (issue #5850)
+
+2005-11-23 Russell Bryant <russell@digium.com>
+
+ * configs/voicemail.conf.sample: Add note to indicate that #include should not be used for this file. (issue #5828)
+
+ * indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
+ * configs/indications.conf.sample: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
+ * include/asterisk/indications.h: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
+ * res/res_indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826)
+
+ * apps/app_voicemail.c: Remove left over "yay!" debugging message. (issue #5829)
+
+2005-11-21 Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_cut.c: remove unnecessary include that causes spurious rebuilding
+
+ * channels/chan_sip.c (build_peer): ensure that case changes made to peer names are not ignored during reload operations
+ (build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified
+
+ * channels/chan_iax2.c (build_peer and build_user): ensure that case changes made to peer/user names are not ignored during reload operations
+ (build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified
+
+2005-11-21 Russell Bryant <russell@digium.com>
+
+ * Makefile: Revert previous change for Darwin.
+
+ * apps/app_osplookup.c: Properly populate the number of results. (issue #5789)
+
+ * Makefile: Don't hard-code that poll functionality needs to be provided on Darwin.
+ * apps/Makefile: Fix incorrect portion of the patch to fix 'make install' on Solaris.
+
+ * channels/chan_iax2.c (iax2_getpeername): Return non-zero to indicate that a peer was found when using realtime (issue #5815)
+
+2005-11-20 Russell Bryant <russell@digium.com>
+
+ * Makefile apps/Makefile: Fix 'make install' for Solaris. (issue #5775)
+
+ * apps/app_record.c: Don't leak a frame if writing it to the file fails. (issue #5787)
+
+ * Makefile: Create the monitor spool directory when the other spool directories are created.
+
+ * channels/chan_sip.c channels/chan_iax2.c: Change warning messages about the number of scheduled events happening all at once to debug messages. (issue #5794)
+
+ * pbx/pbx_spool.c: Fix crash when a value is not specified with a variable on a Set: line in a call file. (issue #5806)
+
+ * apps/app_meetme.c: Fix the 'X' option to the MeetMe application. (issue #5773)
+
+ * apps/app_voicemail.c: Correct the use of a mailbox entered by the calling party instead of indicated as an argument to the Voicemail application. (issue #5774)
+
+ * apps/app_controlplayback.c: Fix logic in checking for success when jumping to priority n+101.
+ * apps/app_md5.c: Fix logic in checking for success when jumping to priority n+101.
+
+ * apps/app_hasnewvoicemail.c: Fix a typo in the application description. Also, fix the logic in checking for success when jumping to priority n+101. (issue #5795)
+
+ * UPGRADE.txt: Add a note on a second way that the IAX2 channel naming convention has changed. (issue #5792)
+ * channels/chan_iax2.c: Fix alignment of the output for the "iax2 show peer <peer>" CLI command (issue #5792)
+
+ * channels/Makefile: Re-add chan_oss to the default build. (issue #5799)
+
+ * res/res_musiconhold.c: Fix incorrect argument for the buffer size to an ast_copy_string call (issue #5803)
+
+ * funcs/func_enum.c: Shorten the module description (issue #5791)
+
+2005-11-17 Russell Bryant <russell@digium.com>
+
+ * Makefile: Fix the output of Makefile generated variables to doxygen
+
+ * channels/chan_sip.c: Add missing carriage return and line feed to the SDP line indicating that we don't support VAD (issue #5780)
+
+2005-11-16 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.0 released.
+
+2005-11-16 Jeremy McNamara <jj@nufone.net>
+
+ * apps/app_voicemail.c (load_config): do not terminate asterisk if no voicemail config file
+ * channels/chan_skinny: Don't register channel type until ready, code formatting updates
+
+2005-11-16 Josh Roberson <josh@asteriasgi.com>
+
+ * Makefile: Update to fix non-responsive remote console on Darwin (OSX)(issue #5757)
+
+2005-11-16 Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/Makefile: don't build chan_modem and sub-modules by default
+ * configs/modules.conf.sample: explicitly 'noload' chan_modem.so and submodules, in case old versions exist
+
+ * res/Makefile: issue mpg123 not-installed warning at 'make install' time, not 'make'
+
+ * apps/app_forkcdr.c (forkcdr_exec): issue warning (and don't segfault) if ForkCDR is called on a channel that doesn't have a CDR (issue #5763)
+
+ * channel.c (ast_queue_hangup): ensure that the channel lock is held before changing its fields... (issue #5770)
+
+ * res/res_musiconhold.c: don't spit out incorrect log messages (and leak memory) during reload (issue #5766)
+
+ * channels/chan_sip.c (process_sdp): don't pass video codec number into ast_getformatname(), it is not valid input for that function (issue #5764)
+
+ * pbx/pbx_ael.c (match_assignment): properly parse equal signs surrounded by whitespace (issue #5761)
+
+ * doc/README.realtime: document the limitations of using FreeTDS with Realtime (issue #5767)
+
+2005-11-15 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: use -g3 for compiler to include macro information for debugger
+
+ * astmm.c (__ast_vasprintf): don't re-use the ap list without copying it; that's not safe on some platforms (issue #5035)
+
+ * doc/README.backtrace: add note about properly building Asterisk to be able to produce backtraces; wrap text and remove DOS line endings
+
+ * channels/chan_sip.c (add_codec_to_sdp): add 'annexb=no' to G.729A SDP (issue #5539)
+
+ * channels/chan_alsa.c (alsa_hangup): handle autohangup properly (issue #5672)
+
+ * channels/chan_misdn.c (and other files): various fixes (issue #5739)
+
+ * channels/chan_sip.c (handle_request_info): properly forward 'flash' events received via SIP INFO (issue #5751, different patch)
+
+ * apps/app_disa.c (disa_exec): don't duplicate constant strings when not needed
+
+ * apps/app_playback.c (playback_exec): use correct logic tests for options (issue #5752)
+
+ * apps/app_disa.c (disa_exec): use standard arg parsing routines (issue #5736)
+
+2005-11-15 Russell Bryant <russell@digium.com>
+
+ * manager.c: Don't crash on a SetVar action if the channel name is not set, or variable's value is not set (issue #5760)
+
+ * doc/README.variables: Add application exit status variables
+
+2005-11-14 Josh Roberson <josh@asteriasgi.com>
+
+ * manager.c: Fix crash on variable passing from AMI originate (issue #5737)
+
+2005-11-14 Russell Bryant <russell@digium.com>
+
+ * many files: Merge doxygen documentation updates. (issue #5605)
+
+ * apps/app_dial.c: Fix typo in RetryDial description.
+
+2005-11-12 Russell Bryant <russell@digium.com>
+
+ * channels/chan_oss.c: Fix a typo in an error message.
+
+2005-11-11 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.0-rc2 released.
+
+2005-11-11 Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c (thread_safe_rand): ensure that threads don't get the same random number (issue #5712)
+
+ * apps/app_voicemail.c (forward_message): correct bugs in message forwarding (issue #5718)
+ (copy_message): use correct path for locking (issue #5704)
+
+ * apps/app_dial.c (wait_for_answer): correct flag copying for automon feature (issue #5720)
+
+ * channels/chan_iax2.c: correct comment
+
+ * apps/app_voicemail.c (close_mailbox): correct previous commit (issue #5663)
+ (vm_change_password): fix password change writing (issue #5721)
+
+ * channels/chan_sip.c (transmit_invite): remove useless debug message; don't try to add OSP tokens to OPTIONS pings
+
+ * apps/app_voicemail.c (close_mailbox): properly remove deleted messages at mailbox close time (issue #5663)
+
+2005-11-11 Mark Spencer <markster@digium.com>
+
+ * channels/chan_zap.c (zt_bridge): only enable/disable DTMF detection on SUB_REAL channels
+
+2005-11-10 Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: ensure that system headers that provide basic types are included first (issue #5713)
+
+2005-11-11 Russell Bryant <russell@digium.com>
+
+ * many files in apps/: Clean up application descriptions. Clarify some wording and make sure they wrap at 80 characters.
+
+2005-11-10 Mark Spencer <markster@digium.com>
+
+ * rtp.c (ast_rtp_raw_write): use unsigned int for return value from calc_txstamp() (issue #5595)
+ (calc_txstamp): never return a value that was less than zero before being turned into 'unsigned int' (issue #5595)
+
+2005-11-10 Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/chanspy.h: move spy-related stuff into separate header so chan_h323 can build (issue #5590)
+
+ * include/asterisk/linkedlists.h (AST_LIST_HEAD_SET_NOLOCK): properly initialize tail pointer when list head is directly set (issue #5669)
+
+ * app.c (ast_app_parse_options): ok, so we aren't all perfect... let's make the while loop actually work properly here (issue #5684)
+
+ * apps/app_disa.c (disa_exec): correct password file parsing (issue #5676)
+
+ * apps/app_meetme.c (conf_run): don't restrict admin users from joining a locked conference (issue #5680)
+
+ * channels/chan_misdn.c: include stdio.h (issue #5671)
+ * channels/chan_misdn_config.c: fix prototype warning (issue #5671)
+
+ * pbx.c: remove apps that were deprecated before 1.0 was released (issue #5673)
+
+ * apps/app_striplsd.c, apps/app_substring.c: remove apps that were deprecated before 1.0 was released (issue #5673)
+
+ * include/asterisk/lock.h (PTHREAD_MUTEX_RECURSIVE_NP): work around header problems on Cygwin (issue #5668)
+
+ * pbx/pbx_ael.c: handle switch default cases inside macros properly (issue #5354)
+
+ * configs/voicemail.conf.sample (format): add strong warning about changing format list when mailboxes contain messages (issue #5689)
+
+ * many files: ensure that system headers are included before Asterisk headers (issue #5693)
+
+ * channels/chan_iax2.c (complete_iax2_show_peer): don't return from function without releasing lock (issue #5685)
+
+ * channels/iax2-provision.c (iax_provision_reload): don't leak memory (issue #5700)
+
+ * pbx/pbx_ael.c (handle_macro): don't leak memory (issue #5701)
+ (handle_context): ditto
+
+ * res/res_features.c (load_config): properly initialize referenced variable (issue #5703)
+
+ * apps/app_queue.c (rqm_exec): correct segfault problem (issue #5705)
+ (aqm_exec): ditto
+
+ * app.c (ast_app_parse_options): don't increment 's' until after checking for NULL (related to issue #5630)
+
+ * apps/app_rpt.c: solve a memory leak (config structure was not freed) (issue #5706)
+
+2005-11-10 Russell Bryant <russell@digium.com>
+
+ * app.c (ast_app_separate_args): Don't consider the open parenthesis as part of the arguments to an option. (issue #5630)
+
+ * many files: Change all references to ast_separate_app_args to ast_app_separate_args
+
+ * many files in apps/: Clean up some application descriptions. Make sure all descriptions in changed files are wrapped at 80 characters.
+
+2005-11-09 Russell Bryant <russell@digium.com>
+
+ * pbx.c: Clean up descriptions of built-in dialplan applications. Changes include clearer wording and not referring to return values.
+
+2005-11-09 Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c (update_registry): don't complain about unspecifed registration expiration intervals, just use the minimum
+
+2005-11-08 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.0-rc1 released.
+
+ * include/asterisk/file.h: add test to ensure that stdio.h is included before this file (issue #5658)
+
+ * res/res_odbc.c (odbc_prepare_and_execute): add new API call for use to properly handle prepared statements across server disconnects (issue #5563)
+
+ * pbx.c (pbx_substitute_variables_helper_full): use already-substituted buffer for parsing variable name (issue #5664)
+
+ * channels/chan_zap.c (zt_request): return AST_CAUSE_CONGESTION when a group-channel is requested and the group exists but all channels are busy (issue #3360, related fix)
+ * channels/chan_iax2.c (create_addr): treat UNREACHABLE as AST_CAUSE_UNREGISTERED so that it will generate CHANUNAVAIL from app_dial (issue #3360)
+
+ * res/res_features.c (ast_bridge_call_thread_launch): set SCHED_RR separately from thread creation, so it won't fail when running as non-root (issue #5601, different fix)
+
+ * pbx.c (pbx_builtin_pushvar_helper): add new API function for setting variables that can exist multiple times (issue #2720)
+ * apps/Makefile (APPS): add app_stack (issue #2720)
+ * apps/app_stack.c: new applications (issue #2720)
+
+ * apps/app_meetme.c: fix two audio delay problems related to using non-Zap channels in conferences (issues #3599 and #4252)
+ * configs/meetme.conf.sample: add documentation of new 'audiobuffers' setting to control buffering on incoming audio from non-Zap channels
+
+ * channels/chan_local.c (local_call): move channel variables from incoming to outgoing instead of inheriting them (issue #5604)
+
+ * many files: add explicit include of stdio.h (issue #5650)
+
+2005-11-07 Kevin P. Fleming <kpfleming@digium.com>
+
+ * UPGRADE.txt (Parking): add note about new parking behavior (issue #5532)
+
+ * many files: more Cygwin compatibility, and proper getloadavg() prototype/macro (issue #5569)
+
+ * include/asterisk/lock.h (__ast_pthread_mutex_lock): correct build with DETECT_DEADLOCKS defined (issue #5570)
+
+2005-11-07 Russell Bryant <russell@digium.com>
+
+ * apps/app_queue.c: upgrade to new arg/option API and implement priority jumping control (issue #5580)
+ * many files: Add missing include of stdio.h, and remove some duplicate and unused header includes
+
+ * include/asterisk/app.h: Increment the arg_index in the options structure to fix applicaiton options that have arguments to them
+
+2005-11-07 Kevin P. Fleming <kpfleming@digium.com>
+
+ * cryptostub.c: include necessary headers
+ * include/asterisk/crypto.h: don't include unnecessary headers
+
+ * manager.c (action_setvar): add support for setting global variables (issue #5571)
+
+ * Makefile: correct cross-compilation issue introduced in Cygwin patches (issue #5572)
+
+ * apps/app_voicemail.c: upgrade to new arg/option API and implement priority jumping control (issue #5649)
+
+ * asterisk.c (main): setpriority() failure is not a reason to stop the process (issue #5581)
+
+ * say.c (ast_say_date_with_format_da): say hours properly (issue #5576)
+
+ * manager.c (astman_get_variables): restore old multiple-variable behavior for "Variable" header (issue #5585)
+
+ * many files: don't check for NULL before calling ast_strlen_zero, it can do it itself (issue #5648)
+
+ * pbx.c (handle_show_hints): use proper state-to-string function for hint state (issue #5583)
+
+ * rtp.c: use unsigned format for debug packet output (issue #5595)
+
+ * asterisk.c (main): force a dnsmgr background refresh after all other modules are initialized (issue #5599)
+ * dnsmgr.c: add ability to start a background refresh on demand (issue #5599)
+
+ * apps/app_dial.c (HANDLE_CAUSE): set CDR disposition to match cause code (issue #5602)
+
+ * asterisk.c: support 'runuser' and 'rungroup' options in asterisk.conf (issue #5621)
+
+ * res/Makefile, apps/Makefile, channels/Makefile, Makefile: support WITHOUT_ZAPTEL define to forcibly avoid building Zaptel support (issue #5634)
+
+ * Makefile: various fixes (issue #5633)
+
+ * apps/app_osplookup.c: upgrade to new arg/option API and implement priority jumping control
+
+ * channels/chan_misdn.c: various fixes (issue #5639)
+ * channels/misdn/isdn_lib.c: various fixes (issue #5639)
+
+ * apps/app_playback.c: upgrade to new arg/option API and implement priority jumping control
+
+ * apps/app_privacy.c: upgrade to new arg/option API and implement priority jumping control
+
+ * apps/app_sendtext.c: upgrade to new arg/option API and implement priority jumping control
+
+ * apps/app_transfer.c: upgrade to new arg/option API and implement priority jumping control
+
+ * apps/app_txtcidname.c: upgrade to new arg/option API and implement priority jumping control
+
+ * Makefile: restore function of 'dont-optimize'
+
+ * config.c (config_text_file_load): don't generate log message when stat() fails
+
+ * many files: clean up application documentation to not refer to return values, since they cannot be used in the dialplan (work done by Neil Lewis)
+
+2005-11-06 Russell Bryant <russell@digium.com>
+
+ * many files: alphabetize options in applicaiton descriptions
+
+ * channels/chan_iax2.c: Use an enum to define iax peer/user flags as well as the pvt structure state. Use the ast_flags macros for checking or setting the state.
+
+ * sounds.txt: Add missing words from the description of the vm-opts prompt
+
+ * apps/app_externalivr.c: Add a space that fixes building on older versions of gcc
+
+ * many files: Add doxygen updates to categorize modules into groups. Convert a lot of comments over to doxygen style. Add some text giving a basic overview of channels.
+
+ * many files: Update applications to add an exit status variable, make priority jumping optional, and use new args parsing macros
+
+ * pbx.c cdr.c res/res_features.c apps/app_dial.c include/asterisk/cdr.h: Convert some built-in applications to use new args parsing macros. Change ast_cdr_reset to take a pointer to an ast_flags structure instead of an integer for flags.
+
+ * channels/chan_agent.c: Don't loop forever on an invalid options string
+
+ * apps/app_disa.c apps/app_forkcdr.c: Fix to use correct arguments to ast_cdr_reset
+
+2005-11-05 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: don't rebuild asterisk/build.h unless the asterisk binary is going to be relinked for some other reason (stops spurious recompile/link every time 'make' is issued); clean up variable substitutions to use consistent syntax
+ * asterisk.c: don't include asterisk/build.h (it's unnecessary)
+ * cli.c: don't include asterisk/build.h, use extern references to buildinfo.c
+ * buildinfo.c: new file to hold version info strings
+
+2005-11-04 Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_mixmonitor.c (mixmonitor_exec): correct app name in an error message
+
+2005-11-04 Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Create a function that stores a peer's status in a given buffer. Use this function in "iax2 show peers" and "iax2 show peer <peername>". Also, add the peer's status as an option to the IAXPEER dialplan function.
+
+2005-11-04 Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/compiler.h: don't try to use always_inline on old compilers
+
+2005-11-03 Russell Bryant <russell@digium.com>
+
+ * res/res_agi.c: initialize buffer for result so that the contents are always valid in the response to GET FULL VARIABLE
+
+2005-11-03 Kevin P. Fleming <kpfleming@digium.com>
+
+ * doc/README.variables: document DYNAMIC_FEATURES
+
+ * res/res_features.c (ast_bridge_call): remove unused variables
+
+ * apps/app_dial.c (dial_exec_full): simplify options and flag usage
+
+ * include/asterisk/app.h: re-work application arg/option parsing APIs for consistent naming, add doxygen docs for option API
+ * many files: update to new APIs
+
+2005-11-02 Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_dial.c (dial_exec_full): convert to use API calls for argument/option parsing
+
+ * include/asterisk/channel.h: add doxygen docs for silence generator APIs
+
+ * channel.c (ast_channel_bridge): simplify native-bridge return logic, remove 'unsuccessful' message since it causes too many questions :-)
+
+2005-11-01 Kevin P. Fleming <kpfleming@digium.com>
+
+ * stdtime/localtime.c: fix build failure on uClibc systems (issue #5558)
+ * devicestate.c: same
+
+ * many files: make chan_misdn actually build (issue #5566)
+
+ * many files: more Cygwin build system support (issue #4678)
+
+ * apps/app_parkandannounce.c (parkandannounce_exec): supply parent channel to ast_request_and_dial so channel variables can be inherited (issue #5564)
+ * include/asterisk/channel.h: add parent_channel field
+ * channel.c (__ast_request_and_dial): use parent_channel field to inherit variables into new channel
+
+ * apps/app_cut.c (cut_internal): use ast_app_separate_args() instead of open code (issue #5560)
+
+ * apps/app_mixmonitor.c (launch_monitor_thread): ast_strlen_zero can handle NULL input (issue #5561)
+ (mixmonitor_exec): same
+
+ * res/res_features.c (ast_feature_request_and_dial): ensure that channel variables are inherited from the channel placing the call (issue #5499)
+
+ * utils.c (getloadavg): change to using _BSD_SOURCE as the indicator for whether this function is present or not (issue #5549)
+
+ * include/asterisk/utils.h (ast_slinear_saturated_add): force to be inlined whenever possible
+ (ast_slinear_saturated_multiply): same
+ (ast_slinear_saturated_divide): same
+ (inaddrcmp): same
+ * include/asterisk/strings.h (ast_strlen_zero): force to be inlined whenever possible
+ * include/asterisk/compiler.h (force_inline): add macro to force inlining of functions
+
+ * app.c (ast_play_and_record): use ast_silence_generator during recording if requested
+ * asterisk.c: add global option to enable silence-during-record (issue #5135)
+ * channel.c (silence_generator_alloc): new
+ (silence_generator_release): new
+ (silence_generator_generate): new
+ (ast_channel_start_silence_generator): new API call to start generating silence on a channel
+ (ast_channel_stop_silence_generator): parallel call to stop silence generation
+ * apps/app_record.c (record_exec): use ast_silence_generator during recording if requested
+
+2005-11-01 Kevin P. Fleming <kpfleming@digium.com>
+
+ * Asterisk 1.2.0-beta2 released.
+