[svn-commits] kpfleming: tag 1.2.11 r40846 - in /tags/1.2.11:
.lastclean .version ChangeLog
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Aug 22 11:56:44 MST 2006
Author: kpfleming
Date: Tue Aug 22 13:56:44 2006
New Revision: 40846
URL: http://svn.digium.com/view/asterisk?rev=40846&view=rev
Log:
importing files for 1.2.11 release
Added:
tags/1.2.11/.lastclean (with props)
tags/1.2.11/.version (with props)
tags/1.2.11/ChangeLog (with props)
Added: tags/1.2.11/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.2.11/.lastclean?rev=40846&view=auto
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--- tags/1.2.11/ChangeLog (added)
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+2006-08-22 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.11 released
+
+2006-08-22 02:59 +0000 [r40821] Tilghman Lesher <tilghman at 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 at 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 at mail.jeffandtilghman.com>
+
+ * CREDITS: Reformat to match the contribution style of other
+ contributors
+
+2006-08-20 04:49 +0000 [r40601] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * CREDITS: add explicit listing of anthm's contributions (issue
+ #7683)
+
+2006-08-08 17:04 +0000 [r39350] Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * Asterisk 1.2.10 released
+
+2006-07-14 13:31 +0000 [r37612] Tilghman Lesher <tilghman at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * app.c: GRRR no fprintf!
+
+2006-07-11 19:00 +0000 [r37378] Joshua Colp <jcolp at 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 at digium.com>
+
+ * channel.c: do masquerade-behind-proxy checking with better
+ control over locks
+
+2006-07-07 23:57 +0000 [r37307] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at mail.jeffandtilghman.com>
+
+ * funcs/func_db.c: Fix spelling/grammar (issue 7493)
+
+2006-07-05 15:31 +0000 [r36998] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * configs/indications.conf.sample: Add Venezuelan indications
+ (issue #7402 reported by palillo)
+
+2006-06-20 15:05 +0000 [r35121] Tilghman Lesher <tilghman at 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 at edvina.net>
+
+ * channels/chan_sip.c: Issue #6820 - Possible fix (already
+ implemented in trunk)
+
+2006-06-19 20:27 +0000 [r34911] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * Asterisk 1.2.9.1 released
+
+2006-06-06 16:02 +0000 [r32582] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * callerid.c: Bug 7268 - Callerid leaks memory on error
+
+2006-06-06 15:48 +0000 [r32566] Kevin P. Fleming <kpfleming at 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 at digium.com>
+
+ * Asterisk 1.2.9 released
+
+2006-06-05 19:53 +0000 [r32373] Kevin P. Fleming <kpfleming at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * rtp.c: silence a warning message that is not a warning
+
+2006-05-31 20:26 +0000 [r31127] Russell Bryant <russell at digium.com>
+
+ * channels/chan_zap.c: fix misplaced manager event (issue #6866,
+ flefoll)
+
+2006-05-30 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.8 released
+
+2006-05-30 14:55 +0000 [r30770] BJ Weschke <bweschke at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * channels/chan_iax2.c: fix the sourceaddress option (issue #7213,
+ alphaque)
+
+2006-05-23 18:16 +0000 [r29764] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: simplify/fix lock retry, and fix comment
+
+2006-05-23 17:17 +0000 [r29733] BJ Weschke <bweschke at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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
[... 2188 lines stripped ...]
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