[asterisk-commits] tag 1.2.9 r32403 - in /tags/1.2.9: .lastclean
.version ChangeLog
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Jun 5 13:16:44 MST 2006
Author: kpfleming
Date: Mon Jun 5 15:16:44 2006
New Revision: 32403
URL: http://svn.digium.com/view/asterisk?rev=32403&view=rev
Log:
importing files for 1.2.9 release
Added:
tags/1.2.9/.lastclean (with props)
tags/1.2.9/.version (with props)
tags/1.2.9/ChangeLog (with props)
Added: tags/1.2.9/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.2.9/.lastclean?rev=32403&view=auto
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--- tags/1.2.9/ChangeLog (added)
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+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)
+
+2006-04-11 20:46 +0000 [r19303] Joshua Colp <joshnet at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * Asterisk 1.2.6 released
+
+2006-03-25 05:07 +0000 [r14821-14868] Tilghman Lesher <tilghman at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * channels/chan_sip.c: Reset global_rtautoclear at sip reload
+
+2006-03-16 20:05 +0000 [r13279] Tilghman Lesher <tilghman at 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 at edvina.net>
+
+ * configs/sip.conf.sample: Issue #6690 - clarify progressinband
+ default setting
+
+2006-03-16 17:42 +0000 [r13237] Russell Bryant <russell at 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 at edvina.net>
+
+ * channels/chan_sip.c: To quote giant developers: "Oops". Thanks,
+ Tony!
+
+2006-03-15 22:16 +0000 [r13161] Russell Bryant <russell at 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 at 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 at 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 at digium.com>
+
+ * pbx.c: add locking to protect the list of global dialplan
+ variables
+
+2006-03-12 17:57 +0000 [r12577] Russell Bryant <russell at digium.com>
+
+ * codecs/gsm/Makefile: fix build on parisc (issue #6704)
+
+2006-03-10 12:13 +0000 [r12477-12495] Olle Johansson <oej at 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 at 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 at 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 at edvina.net>
+
+ * channel.c: Revert earlier change
+
+ * channel.c: Fix for astmm compilation
+
+2006-03-06 02:32 +0000 [r11946] Russell Bryant <russell at 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 at edvina.net>
+
+ * channels/chan_sip.c: Clear page2 flags at reload too
+
+2006-03-04 11:45 +0000 [r11778] BJ Weschke <bweschke at 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 at digium.com>
+
+ * Asterisk 1.2.5 released
+
[... 1463 lines stripped ...]
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