[asterisk-commits] tag 1.2.10-netsec r37642 - in
/tags/1.2.10-netsec: .lastclean .version ChangeLog
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Jul 14 14:32:20 MST 2006
Author: kpfleming
Date: Fri Jul 14 16:32:20 2006
New Revision: 37642
URL: http://svn.digium.com/view/asterisk?rev=37642&view=rev
Log:
importing files for 1.2.10-netsec release
Added:
tags/1.2.10-netsec/.lastclean (with props)
tags/1.2.10-netsec/.version (with props)
tags/1.2.10-netsec/ChangeLog (with props)
Added: tags/1.2.10-netsec/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.2.10-netsec/.lastclean?rev=37642&view=auto
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--- tags/1.2.10-netsec/ChangeLog (added)
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+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
+ value of res before doing retries stuff. (issue #7140 reported by
+ tanischen)
+
+2006-05-18 12:13 +0000 [r28125] Olle Johansson <oej at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * channels/chan_sip.c: Issue #7074 - Problem with long contact
+ lines
+
+2006-05-02 19:39 +0000 [r24295] BJ Weschke <bweschke at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * Asterisk 1.2.7.1 released
+
+2006-04-13 17:40 +0000 [r19812] Kevin P. Fleming <kpfleming at 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 at mail.jeffandtilghman.com>
+
+ * pbx.c: Bug 6957 - variable names beginning with CALLERID weren't
+ substituted correctly
+
+2006-04-12 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.7 released
+
+2006-04-11 22:39 +0000 [r19394-19397] Tilghman Lesher <tilghman at 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 at 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 at mail.jeffandtilghman.com>
+
+ * asterisk.c: Bug 6097 - possible descriptor leak
+
+2006-04-11 21:50 +0000 [r19345-19348] Kevin P. Fleming <kpfleming at 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)
+
[... 1853 lines stripped ...]
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