[svn-commits] bebuild: tag 10.1.0-rc1 r349397 - /tags/10.1.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Dec 29 13:41:35 CST 2011
Author: bebuild
Date: Thu Dec 29 13:41:31 2011
New Revision: 349397
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=349397
Log:
Importing files for 10.1.0-rc1 release.
Added:
tags/10.1.0-rc1/.lastclean (with props)
tags/10.1.0-rc1/.version (with props)
tags/10.1.0-rc1/ChangeLog (with props)
Added: tags/10.1.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/10.1.0-rc1/.lastclean?view=auto&rev=349397
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+2011-12-29 15:14 +0000 [r349340] Matthew Jordan <mjordan at digium.com>
+
+
+ * main/rtp_engine.c, /: Handle AST_CONTROL_UPDATE_RTP_PEER frames
+ in local bridge loop Failing to handle
+ AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
+ causes the loop to exit prematurely. This causes a variety of
+ negative side effects, depending on when the loop exits. This
+ patch handles the frame by essentially swallowing the frame in
+ the local loop, as the current channel drivers expect the RTP
+ bridge to handle the frame, and, in the case of the local bridge
+ loop, no additional action is necessary. (issue ASTERISK-19040)
+ (issue ASTERISK-19128) (issue ASTERISK-17725) (issue
+ ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan
+ Schmidt Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1640/ ........ Merged
+ revisions 349339 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-28 21:33 +0000 [r349290] Sean Bright <sean at malleable.com>
+
+ * /, main/audiohook.c: Use ast_audiohook_write_list_empty to
+ determine if our lists are empty instead of duplicating that
+ logic. ........ Merged revisions 349289 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-28 19:00 +0000 [r349248-349250] Kevin P. Fleming <kpfleming at digium.com>
+
+ * utils: Tell Subversion to gnore the 'astdb2bdb' binary file if it
+ exists.
+
+ * main/dsp.c, res/res_fax.c, include/asterisk/dsp.h,
+ include/asterisk/res_fax.h, res/res_fax_spandsp.c: Improve T.38
+ gateway V.21 preamble detection. This commit removes the V.21
+ preamble detection code previously added to the generic DSP
+ implementation in Asterisk, and instead enhances the res_fax
+ module to be able to utilize V.21 preamble detection
+ functionality made available by FAX technology modules. This
+ commit also adds such support to res_fax_spandsp, which uses the
+ Spandsp modem tone detection code to do the V.21 preamble
+ detection. There should be no functional change here, other than
+ much more reliable V.21 preamble detection (and thus T.38 gateway
+ initiation).
+
+2011-12-27 20:53 +0000 [r349195] Matthew Jordan <mjordan at digium.com>
+
+ * /, res/res_timing_pthread.c, include/asterisk/module.h,
+ res/res_timing_dahdi.c, res/res_timing_timerfd.c,
+ res/res_musiconhold.c: Fix timing source dependency issues with
+ MOH Prior to this patch, res_musiconhold existed at the same
+ module priority level as the timing sources that it depends on.
+ This would cause a problem when music on hold was reloaded, as
+ the timing source could be changed after res_musiconhold was
+ processed. This patch adds a new module priority level,
+ AST_MODPRI_TIMING, that the various timing modules are now loaded
+ at. This now occurs before loading other resource modules, such
+ that the timing source is guaranteed to be set prior to resolving
+ the timing source dependencies. (closes issue ASTERISK-17474)
+ Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf,
+ Wes Van Tlghem, elguero, Thomas Arimont Patches:
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff
+ uploaded by elguero (License #5026)
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff
+ uploaded by elguero (License #5026)
+ asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by
+ elguero (License #5026) Review:
+ https://reviewboard.asterisk.org/r/1578/ ........ Merged
+ revisions 349194 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-27 17:17 +0000 [r349145] Sean Bright <sean at malleable.com>
+
+ * /, main/audiohook.c: Once an audiohook is attached to a channel,
+ we continue to transcode all of the frames, even after all of the
+ hooks are detached. This patch short-cicuits us out before we
+ transcode unnecessarily. ........ Merged revisions 349144 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-23 17:32 +0000 [r349045] Sean Bright <sean at malleable.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 349044 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec
+ 2011) | 18 lines In ChanSpy, don't create audiohooks that will
+ never be used. When ChanSpy is initialized it creates and
+ attaches 3 audiohooks: 1) Read audio off of the channel that we
+ are spying on 2) Write audio to the channel that we are spying on
+ 3) Write audio to the channel that is bridged to the channel that
+ we are spying on. The first is always necessary, but the others
+ are used only when specific options are passed to the ChanSpy
+ application (B, d, w, and W to be specific). When those flags are
+ not passed, neither of those audiohooks are ever sent frames, but
+ we still try to process the hooks for each voice frame that we
+ recieve on the channel. So in short - only create and attach
+ audiohooks that we actually need. ........
+
+2011-12-23 15:25 +0000 [r348993] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_dial.c, /: Fix missing doc tags found while fixing
+ ASTERISK-18689 Add missing <variable></variable> tags in app_dial
+ documentation. ........ Merged revisions 348992 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-23 02:30 +0000 [r348952] Richard Mudgett <rmudgett at digium.com>
+
+ * main/pbx.c, /, channels/chan_sip.c, include/asterisk/pbx.h: Fix
+ extension state callback references in chan_sip. Chan_sip gives a
+ dialog reference to the extension state callback and assumes that
+ when ast_extension_state_del() returns, the callback cannot
+ happen anymore. Chan_sip then reduces the dialog reference count
+ associated with the callback. Recent changes (ASTERISK-17760)
+ have resulted in the potential for the callback to happen after
+ ast_extension_state_del() has returned. For chan_sip, this could
+ be very bad because the dialog pointer could have already been
+ destroyed. * Added ast_extension_state_add_destroy() so chan_sip
+ can account for the sip_pvt reference given to the extension
+ state callback when the extension state callback is deleted. *
+ Fix pbx.c awkward statecbs handling in
+ ast_extension_state_add_destroy() and handle_statechange() now
+ that the struct ast_state_cb has a destructor to call. * Ensure
+ that ast_extension_state_add_destroy() will never return -1 or 0
+ for a successful registration. * Fixed pbx.c statecbs_cmp() to
+ compare the correct information. The passed in value to compare
+ is a change_cb function pointer not an object pointer. * Make
+ pbx.c ast_merge_contexts_and_delete() not perform callbacks with
+ AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
+ deadlocking when those locks are held during the callback. *
+ Removed unused lock declaration for the pbx.c store_hints list.
+ (closes issue ASTERISK-18844) Reported by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1635/ ........ Merged
+ revisions 348940 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-22 22:37 +0000 [r348846-348889] Matthew Jordan <mjordan at digium.com>
+
+ * cel/cel_pgsql.c, /: Fix for memory leaks / cleanup in cel_pgsql
+ There were a number of issues in cel_pgsql's pgsql_log method: *
+ If either sql or sql2 could not be allocated, the method would
+ return while the pgsql_lock was still locked * If the execution
+ of the log statement succeeded, the sql and sql2 structs were
+ never free'd * Reconnection successes were logged as ERRORs. In
+ general, the severity of several logging statements was reduced
+ (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
+ by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/
+ ........ Merged revisions 348888 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/say.c, main/file.c, main/app.c, apps/app_confbridge.c,
+ main/bridging.c: Add Asterisk TestSuite event hooks to support
+ ConfBridge testing This patch adds initial testsuite event hooks
+ so that ConfBridge tests can be executed in the Asterisk
+ TestSuite. (issue ASTERISK-19059)
+
+2011-12-22 20:17 +0000 [r348845] Terry Wilson <twilson at digium.com>
+
+ * /, include/asterisk/format_pref.h: Allow packetization vaules >
+ 127 According to the RTP packetization documentation, and the
+ maximum values listed in AST_FORMAT_LIST, we should support
+ values > that the signed char array that ast_codec_pref makes
+ available to store the value. All places in the code treat the
+ framing field as though it were an int array instaead of a char
+ array anyway, so this just fixes the type of the array. (closes
+ issue ASTERISK-18876) Review:
+ https://reviewboard.asterisk.org/r/1639/ ........ Merged
+ revisions 348833 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-21 20:13 +0000 [r348736-348793] Richard Mudgett <rmudgett at digium.com>
+
+ * codecs/speex: Make codecs/speex ignore *.i files also.
+
+ * apps/confbridge: Make apps/confbridge ignore *.i files also.
+
+ * /, channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS
+ number if it is blank. Some ISDN switches complain or block the
+ call if the RDNIS number is empty. * Made chan_iax2 not save a
+ RDNIS number into the ast_channel if the string is blank. This is
+ what other channel drivers do. (closes issue ASTERISK-17152)
+ Reported by: rmudgett ........ Merged revisions 348735 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-19 21:37 +0000 [r348648] Richard Mudgett <rmudgett at digium.com>
+
+ * /, configure, configure.ac: Fix crashes on other platforms caused
+ by interference from Darwin weak symbol support. Support weak
+ symbols on a platform specific basis. The Mac OS X (Darwin)
+ support must be isolated from the other platforms because it has
+ caused other platforms to crash. Several other platforms
+ including Linux have GCC versions that define the weak attribute.
+ However, this attribute is only setup for use in the code by
+ Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
+ Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged
+ revisions 348647 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-19 19:48 +0000 [r348605] Leif Madsen <lmadsen at digium.com>
+
+ * main/message.c: Update documentation for MESSAGE_SEND_STATUS
+ variable. (Closes issue ASTERISK-19056) Reported by: Yuri
+ Patches: 348360.diff uploaded by Yuri (license #5242)
+
+2011-12-18 18:28 +0000 [r348517] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, configs/sip.conf.sample: Correct two flaws in sip.conf.sample
+ related to AST-2011-013. * The sample file listed *two* values
+ for the 'nat' option as being the default. Only 'force_rport' is
+ the default. * The warning about having differing 'nat' settings
+ confusingly referred to both peers and users. ........ Merged
+ revisions 348515 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 348516 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-16 23:56 +0000 [r348311-348465] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c, /, main/features.c: Clean-up on isle five for
+ __ast_request_and_dial() and ast_call_forward(). * Add locking
+ when a channel inherits variables and datastores in
+ __ast_request_and_dial() and ast_call_forward(). Note: The
+ involved channels are not active so there was minimal potential
+ for problems. * Remove calls to ast_set_callerid() in
+ __ast_request_and_dial() and ast_call_forward() because the set
+ information is for the wrong direction. * Don't use C++ keywords
+ for variable names in ast_call_forward(). * Run the redirecting
+ interception macro if defined when forwarding a call in
+ ast_call_forward(). Note: Currently will never execute because
+ the only callers that supply a calling channel supply a hungup or
+ zombie channel. * Make feature_request_and_dial() put the
+ transferee into autoservice when it calls ast_call_forward() in
+ case a redirection interception macro is run. Note: Currently
+ will never happen because the caller channel (Party B) is always
+ hungup at this time. * Make feature_request_and_dial() ignore the
+ AST_CONTROL_PROCEEDING frame to silence a log message. ........
+ Merged revisions 348464 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/channel.c, /: Fix cut and past error in ast_call_forward().
+ (issue ASTERISK-18836) ........ Merged revisions 348401 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/channel.c, main/pbx.c, /, apps/app_authenticate.c,
+ funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h,
+ apps/app_followme.c, apps/app_queue.c, res/res_monitor.c: Fix
+ crash during CDR update. The ast_cdr_setcid() and
+ ast_cdr_update() were shown in ASTERISK-18836 to be called by
+ different threads for the same channel. The channel driver thread
+ and the PBX thread running dialplan. * Add lock protection around
+ CDR API calls that access an ast_channel pointer. (closes issue
+ ASTERISK-18836) Reported by: gpluser Review:
+ https://reviewboard.asterisk.org/r/1628/ ........ Merged
+ revisions 348362 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_parkandannounce.c, /: Fix ParkAndAnnounce to pass the
+ CallerID to the announcing channel. ParkAndAnnounce tried to pass
+ the CallerID to the announcing channel but the ID was wiped out
+ by the channel masquerade done when parking the call. * Save the
+ CallerID before parking the channel to pass it to the announcing
+ channel. * Fixed a minor memory leak in ParkAndAnnounce. *
+ Updated some ParkAndAnnounce log messages. ........ Merged
+ revisions 348310 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-14 22:34 +0000 [r348265] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_originate.c: Added support for all slin formats to
+ app_originate Previously, app_originate could not originate a
+ call into a non-8kHz conference bridge as the formats for
+ non-8kHz slin codecs were not applied to the created channel.
+ This patch adds all of the formats by default, such that if a
+ created channel has a codec that supports a higher sampling rate,
+ a translation path can be built between it and other channels.
+
+2011-12-14 22:05 +0000 [r348213] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, res/res_fax.c: Don't clear LOCALSTATIONID before sending or
+ receiving. The user may set that variable. ASTERISK-18921
+ ........ Merged revisions 348212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-14 21:58 +0000 [r348211] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_queue.c: Fixed Asterisk crash when function QUEUE_MEMBER
+ receives invalid input The function QUEUE_MEMBER has two required
+ parameters (queuename, option). It was only checking for the
+ presence of queuename. The patch checks for the existence of the
+ option parameter and provides better error logging when invalid
+ values are provided for the option parameter as well.
+
+2011-12-14 20:35 +0000 [r348155-348158] Jonathan Rose <jrose at digium.com>
+
+ * /, configs/features.conf.sample: Fix accidental use of tabs
+ instead of spaces from previous features.conf.sample change
+ ........ Merged revisions 348157 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configs/features.conf.sample: Document PARKINGSLOT variable in
+ features.conf.sample (issue ASTERISK-16239) ........ Merged
+ revisions 348154 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-13 23:06 +0000 [r348102] Richard Mudgett <rmudgett at digium.com>
+
+ * /, bridges/bridge_builtin_features.c, apps/app_followme.c: Fix
+ FollowMe CallerID on outgoing calls. The addition of the
+ Connected Line support changed how CallerID is passed to outgoing
+ calls. The FollowMe application was not updated to pass CallerID
+ to the outgoing calls. * Fix FollowMe CallerID on outgoing calls.
+ * Restructured findmeexec() to fix several memory leaks and
+ eliminate some duplicated code. * Made check the return value of
+ create_followme_number(). Putting a NULL into the numbers list is
+ bad if create_followme_number() fails. * Fixed a couple uses of
+ ast_strdupa() inside loops. * The changes to
+ bridge_builtin_features.c fix a similar CallerID issue with the
+ bridging API attended and blind transfers. (Not used at this
+ time.) (closes issue ASTERISK-17557) Reported by: hamlet505a
+ Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/1612/ ........ Merged
+ revisions 348101 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-13 15:20 +0000 [r348056] Stefan Schmidt <sst at sil.at>
+
+ * channels/chan_sip.c: Fix possible misshandling of an incoming SIP
+ response as a peer poke response. Also make sure peer has even
+ qualify enabled when handle a peer poke response. (closes issue
+ ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
+ UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
+ by: David Vossel ........ Merged revisions 348048 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-12 19:24 +0000 [r347996] Terry Wilson <twilson at digium.com>
+
+ * res/res_srtp.c, /: Add a separate buffer for SRTCP packets The
+ function ast_srtp_protect used a common buffer for both SRTP and
+ SRTCP packets. Since this function can be called from multiple
+ threads for the same SRTP session (scheduler for SRTCP and
+ channel for SRTP) it was possible for the packets to become
+ corrupted as the buffer was used by both threads simultaneously.
+ This patch adds a separate buffer for SRTCP packets to avoid the
+ problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
+ Collins) ........ Merged revisions 347995 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-12 18:13 +0000 [r347953-347955] Richard Mudgett <rmudgett at digium.com>
+
+ * configs/extensions.conf.sample, configs/iax.conf.sample,
+ configs/chan_dahdi.conf.sample, configs/chan_ooh323.conf.sample,
+ configs/vpb.conf.sample, configs/extensions.lua.sample,
+ configs/sip.conf.sample: Reverting -r347953 for ASTERISK-14122
+
+ * configs/extensions.conf.sample, configs/iax.conf.sample,
+ configs/chan_dahdi.conf.sample, configs/chan_ooh323.conf.sample,
+ configs/vpb.conf.sample, configs/extensions.lua.sample,
+ configs/sip.conf.sample: Update sample configs to put incoming
+ calls into context public. * Add warning about the SIP allowguest
+ option in context public. (closes issue ASTERISK-14122) Reported
+ by: Alec Davis Review: https://reviewboard.asterisk.org/r/719/
+
+2011-12-09 01:29 +0000 [r347812] Richard Mudgett <rmudgett at digium.com>
+
+ * main/pbx.c, /: Fix some parsing issues in
+ add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
+ potential sign extension issue. * Fix infinite loop in
+ add_exten_to_pattern_tree() handling of character set escape
+ handling. * Added buffer overflow checks in
+ add_exten_to_pattern_tree() character set collection. * Made
+ ignore empty character sets. * Added escape character handling to
+ end-of-range character in character sets. This has a slight
+ change in behavior if the end-of-range character is an escape
+ character. You must now escape it. * Fix potential sign extension
+ issue when expanding character set ranges. * Made remove
+ duplicated characters from character sets. The duplicate
+ characters lower extension matching priority and prevent
+ duplicate extension detection. * Fix escape character handling
+ when the escape character is trying to escape the end-of-string.
+ We could have continued processing characters after the end of
+ the exten string. We could have added the previous character to
+ the pattern matching tree incorrectly. (closes issue
+ ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions
+ 347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-08 21:31 +0000 [r347727] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: Fix regression when using tcpenable=no
+ and tlsenable=yes. The tlsenable settings are tucked away in
+ main/tcptls.c, so I missed them when resolving ASTERISK-18837.
+ This should resolve the test suite breakage of the sip tls tests.
+ Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
+ Jordan ........ Merged revisions 347718 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-08 20:43 +0000 [r347656] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_queue.c: Fix regressed behavior of queue set penalty to
+ work without specifying 'in <queuename>' r325483 caused a
+ regression in Asterisk 10+ that would make Asterisk segfault when
+ attempting to set penalty on an interface without specifying a
+ queue in the queue set penalty CLI command. In addition, no
+ attempt would be made whatsoever to perform the penalty setting
+ on all the queues in the core list with either the cli command or
+ the non-segfaulting ami equivalent. This patch fixes that and
+ also makes an attempt to document and rename some functions
+ required by this command to better represent what they actually
+ do. Oh yeah, and the use of this command without specifying a
+ specific queue actually works now. Review:
+ https://reviewboard.asterisk.org/r/1609/
+
+2011-12-08 17:53 +0000 [r347600] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/features.c: Mark channel running the h exten with the
+ soft-hangup flag. When a bridge is broken, ast_bridge_call()
+ might execute the h exten on the calling channel. However, that
+ channel may not have been the channel that broke the bridge by
+ hanging up. The channel executing the h exten must be in a hung
+ up state so things like AGI run in the correct mode. * Make sure
+ ast_bridge_call() marks the channel it is executing the h exten
+ on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
+ to match the pbx.c main dialplan execution loop when it executes
+ the h exten.) (closes issue ASTERISK-18811) Reported by: David
+ Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: David Hajek, rmudgett
+ ........ Merged revisions 347595 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-08 16:20 +0000 [r347532] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Don't crash on INFO automon request with
+ no channel AST-2011-014. When automon was enabled in
+ features.conf, it was possible to crash Asterisk by sending an
+ INFO request if no channel had been created yet. (closes issue
+ ASTERISK-18805) ........ Merged revisions 347530 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 347531 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-07 21:39 +0000 [r347439] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c, /: Update AMI Getvar and Setvar documentation
+ about supplying a channel name. (closes issue ASTERISK-18958)
+ Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license
+ #5621) patch uploaded by rmudgett ........ Merged revisions
+ 347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-07 20:27 +0000 [r347383] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_meetme.c: Fix: Meetme recording variables from
+ realtime DB use null entries over channel variables Meetme would
+ attempt to substitute the realtime values of RECORDING_FILE and
+ RECORDING_FORMAT from the meetme db entry instead of using the
+ channel variable set for those variables in spite of those
+ database entries being NULL or even lacking a column to represent
+ them. (closes issue ASTERISK-18873) Reported by: Byron Clark
+ Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
+ 6157) ........ Merged revisions 347369 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-07 20:12 +0000 [r347344] Terry Wilson <twilson at digium.com>
+
+ * Makefile, include/asterisk/paths.h, configs/asterisk.conf.sample,
+ build_tools/make_defaults_h, main/asterisk.c, main/db.c: Add
+ ASTSBINDIR to the list of configurable paths This patch also
+ makes astdb2sqlite3 and astcanary use the configured directory
+ instead of relying on $PATH. (closes issue ASTERISK-18959)
+ Review: https://reviewboard.asterisk.org/r/1613/
+
+2011-12-06 23:56 +0000 [r347293] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
+ signals case insensitive. (closes issue ASTERISK-18924) Reported
+ by: Kevin Taylor ........ Merged revisions 347292 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-06 21:53 +0000 [r347240] Jonathan Rose <jrose at digium.com>
+
+ * main/pbx.c, /: Documents CHANNEL(musicclass) taking priority over
+ m([x]) in waitExten If waitExten specifies a music class to use
+ with its music on hold option, it will use CHANNEL(musicclass)
+ instead if that channel variable has been set on the initiating
+ channel. This documents that behavior in the waitExten app so
+ that this can be known without checking the documentation of the
+ code in function local_ast_moh_start. (closes issue
+ ASTERISK-18804) ........ Merged revisions 347239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-06 19:42 +0000 [r347124-347167] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: Don't allow transport=tcp when
+ tcpenable=no. When tcpenable=no, sending to transport=tcp hosts
+ was still allowed. Resolving the source address wasn't possible
+ and yielded the string "(null)" in SIP messages. Fixed that and a
+ couple of not-so-correct log messages. (closes issue
+ ASTERISK-18837) Reported by: Andreas Topp Review:
+ https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan
+ ........ Merged revisions 347166 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_voicemail.c, /: Add regression tests for issue
+ ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
+ Reviewed by: Matt Jordan ........ Merged revisions 347131 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_voicemail.c, /: Move setting of voicemail zonetag and
+ locale up a bit. The voicemail [general] zonetag and locale
+ variables weren't loaded until after the mailboxes were
+ initialized. This caused the settings to be unset for those
+ mailboxes until a reload was performed. (closes issue
+ ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570
+ Reviewed by: Matt Jordan ........ Merged revisions 347111 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-06 17:24 +0000 [r347068] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Fixed crash from orphaned MWI
+ subscriptions in chan_sip This patch resolves the issue where MWI
+ subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.
+ When a peer is removed, either by pruning realtime SIP peers or
+ by unloading / loading chan_sip, the MWI subscriptions that were
+ orphaned would still be on the event engine list of valid
+ subscriptions but have a pointer to a peer that no longer was
+ valid. When an MWI event would occur, this would cause a seg
+ fault. (closes issue ASTERISK-18663) Reported by: Ross Beer
+ Tested by: Ross Beer, Matt Jordan Patches:
+ blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
+ Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged
+ revisions 347058 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-05 17:42 +0000 [r347007] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Restore call progress code for analog
+ ports. Extracting sig_analog from chan_dahdi lost call progress
+ detection functionality. * Fix analog ports from considering a
+ call answered immediately after dialing has completed if the
+ callprogress option is enabled. (closes issue ASTERISK-18841)
+ Reported by: Richard Miller Patches: chan_dahdi.diff (license
+ #5685) patch uploaded by Richard Miller (Modified by me)
+ sig_analog.c.diff (license #5685) patch uploaded by Richard
+ Miller (Modified by me) sig_analog.h.diff (license #5685) patch
+ uploaded by Richard Miller ........ Merged revisions 347006 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-05 15:02 +0000 [r346955] Jonathan Rose <jrose at digium.com>
+
+ * main/pbx.c, /: Resolve duplicate label used in multiple
+ priorities for the same extension. Prior to this patch, if labels
+ with the same name were used for different priorities in the same
+ extension, the new label would be accepted, but it would be
+ unusable since attempts to reach that label would just go to the
+ first one. Now pbx.c detects this, generates a warning in logs,
+ and culls the label before adding it to the dialplan. (closes
+ issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
+ pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........
+ Merged revisions 346954 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-05 14:46 +0000 [r346952] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_jabber.exports.in, /: Fix chan_jingle/gtalk load
+ regression introduced in r346087 Add missing symbol exports for
+ ast_aji_client_destroy and ast_aji_buddy_destroy for usage
+ outside res_jabber. Testing of these changes focused on
+ res_jabber itself, so this problem was missed. Reported-by:
+ Michael Spiceland ........ Merged revisions 346951 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-04 10:03 +0000 [r346900] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
+ domain ACL bypass. The code that allowed admins to create users
+ with domain-only uri's had stopped to work in 1.8 because of the
+ reqresp parser rewrites. This is fixed now: if you have a
+ [mydomain.com] sip user, you can register with useraddr
+ sip:mydomain.com. Note that in that case -- if you're using
+ domain ACLs (a configured domain list) -- mydomain.com must be in
+ the allow list as well. Reviewboard r1606 shows a list of
+ registration combinations and which SIP response codes are
+ returned. Review: https://reviewboard.asterisk.org/r/1533/
+ Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
+ issue ASTERISK-18741) ........ Merged revisions 346899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-02 23:27 +0000 [r346856] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: Update SIP MESSAGE To parsing to correctly
+ handle URI The previous patch (r346040) incorrectly parsed the
+ URI in the presence of a port, e.g., user at hostname:port would
+ fail as the port would be double appended to the SIP message.
+ This patch uses the parse_uri function to correctly parse the URI
+ into its username and hostname parts, and places them in the
+ correct fields in the sip_pvt structure. (issue ASTERISK-18903)
+ Review: https://reviewboard.asterisk.org/r/1597/
+
+2011-12-02 16:42 +0000 [r346763] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, /, channels/chan_h323.c: Merged revisions
+ 346762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7
+ lines process null frame pointer returned by
+ ast_rtp_instance_read correctly (closes issue ASTERISK-16697)
+ Reported by: under Patches: segfault.diff (License #5871) patch
+ uploaded by under ........
+
+2011-12-01 21:14 +0000 [r346701] Richard Mudgett <rmudgett at digium.com>
+
+ * main/stun.c, /, res/res_stun_monitor.c,
+ configs/res_stun_monitor.conf.sample, include/asterisk/stun.h:
+ Re-resolve the STUN address if a STUN poll fails for
+ res_stun_monitor. The STUN socket must remain open between polls
+ or the external address seen by the STUN server is likely to
+ change. However, if the STUN request poll fails then the STUN
+ server address needs to be re-resolved and the STUN socket needs
+ to be closed and reopened. * Re-resolve the STUN server address
+ and create a new socket if the STUN request poll fails. * Fix
+ ast_stun_request() return value consistency. * Fix
+ ast_stun_request() to check the received packet for expected
+ message type and transaction ID. * Fix ast_stun_request() to read
+ packets until timeout or an associated response packet is found.
+ The stun_purge_socket() hack is no longer required. * Reduce
+ ast_stun_request() error messages to debug output. * No longer
+ pass in the destination address to ast_stun_request() if the
+ socket is already bound or connected to the destination. (closes
+ issue ASTERISK-18327) Reported by: Wolfram Joost Tested by:
+ rmudgett Review: https://reviewboard.asterisk.org/r/1595/
+ ........ Merged revisions 346700 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-12-01 20:37 +0000 [r346565-346698] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
+ ringing. 183 Ringing isn't even a thing. 183 is actually a
+ session progress message. (closes issue ASTERISK-18925) Reported
+ by: Sebastian Denz Tested by: jrose Patches:
+ asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
+ Denz (License #6139) ........ Merged revisions 346697 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
+ r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) |
+ 18 lines Cleaning up chan_sip/tcptls file descriptor closing.
+ This patch attempts to eliminate various possible instances of
+ undefined behavior caused by invoking close/fclose in situations
+ where fclose may have already been issued on a
+ tcptls_session_instance and/or closing file descriptors that
+ don't have a valid index for fd (-1). Thanks for more than a
+ little help from wdoekes. (closes issue ASTERISK-18700) Reported
+ by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane
+ Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas
+ Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged
+ revisions 346564 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-30 19:37 +0000 [r346473] Leif Madsen <lmadsen at digium.com>
+
+ * /, configs/queues.conf.sample: Update queues.conf.sample
+ documentation. Update the documentation surrounding the use of
+ MONITOR_EXEC to make it more clear that it can be used for both
+ Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
+ Reported by: David Woolley Patches:
+ issue18817_mixmonitor_queues_doc.diff by Michael L. Young
+ (License #5026) ........ Merged revisions 346472 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-29 00:00 +0000 [r346349] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/message.h, main/message.c: Fixes memory leak in
+ message API. The ast_msg_get_var function did not properly
+ decrement the ref count of the var it retrieves. The way this is
+ implemented is a bit tricky, as we must decrement the var and
+ then return the var's value. As long as the documentation for the
+ function is followed, this will not result in a dangling pointer
+ as the ast_msg structure owns its own reference to the var while
+ it exists in the var container.
+
+2011-11-28 14:32 +0000 [r346293] Stefan Schmidt <sst at sil.at>
+
+ * res/res_rtp_asterisk.c, /: Fix regression that 'rtp/rtcp set
+ debup ip' only works when also a port was specified. (closes
+ issue ASTERISK-18693) Reported by: Davide Dal Fra Review:
+ https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
+ Doekes ........ Merged revisions 346292 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-23 22:58 +0000 [r346240] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/acl.h, /, channels/chan_skinny.c,
+ channels/chan_h323.c, main/acl.c, channels/chan_iax2.c: Fix calls
+ to ast_get_ip() not initializing the address family. ........
+ Merged revisions 346239 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-23 20:45 +0000 [r346145-346198] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
+ function. In r116240, get_msg_text() got an extra parameter to
+ fix the unwanted addition of trailing newlines to SIP MESSAGE
+ bodies. This caused all linefeeds to be trimmed, which isn't
+ right either. This is a stop-gap; the right fix is to return the
+ original SIP request body. Review:
+ https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan
+ ........ Merged revisions 346147 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, include/asterisk/strings.h: Fix ast_str_truncate signedness
+ warning and documentation. Review:
+ https://reviewboard.asterisk.org/r/1594 ........ Merged revisions
[... 19592 lines stripped ...]
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