[asterisk-commits] bebuild: tag 10.12.0-rc2 r378737 - /tags/10.12.0-rc2/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jan 9 13:50:15 CST 2013
Author: bebuild
Date: Wed Jan 9 13:50:11 2013
New Revision: 378737
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=378737
Log:
Importing files for 10.12.0-rc2 release.
Added:
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tags/10.12.0-rc2/.version (with props)
tags/10.12.0-rc2/ChangeLog (with props)
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+2013-01-09 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.12.0-rc2 Released.
+
+2013-01-08 23:55 +0000 [r378683-378689] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c: app_queue: Fix incorrect assertion. (issue
+ ASTERISK-16115)
+
+ * /, configs/queues.conf.sample, UPGRADE.txt, CHANGES,
+ apps/app_queue.c: app_queue: Fix multiple calls to a queue member
+ that is in only one queue. When ringinuse=no queue members can
+ receive more than one call if these calls happen at nearly the
+ same time. * Fix so a queue member does not receive more than one
+ call from a queue. NOTE: This fix does not prevent multiple calls
+ to a member if the member is in more than one queue. * Did some
+ refactoring to eliminate some code redundancy. (issue
+ ASTERISK-16115) Reported by: nik600 Patches:
+ jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Modified * Revert the -r341580 and -r341599
+ changes adding the queues.conf check_state_unknown option as it
+ was added in an attempt to fix this problem. The fix did not need
+ to be optional. The fix should not have tried to explicitly set
+ the device state. Setting the device state by something other
+ than the device introduces a race condition. I also could not see
+ how the change would be effective other than delaying the
+ app_queue code long enough for the device state to propagate to
+ app_queue. ........ Merged revisions 378663 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-01-02 17:40 +0000 [r378286-378320] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_calendar.c, include/asterisk/devicestate.h,
+ channels/chan_local.c, /, main/ccss.c, channels/chan_sip.c,
+ apps/app_meetme.c, channels/chan_agent.c, main/devicestate.c,
+ include/asterisk/channel.h, res/res_jabber.c, apps/app_queue.c,
+ channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
+ channels/chan_skinny.c, include/asterisk/event_defs.h,
+ main/features.c, main/event.c, apps/app_confbridge.c,
+ apps/confbridge/conf_state_empty.c, funcs/func_devstate.c:
+ Prevent exhaustion of system resources through exploitation of
+ event cache Asterisk maintains an internal cache for devices in
+ the event subsystem. The device state cache holds the state of
+ each device known to Asterisk, such that consumers of device
+ state information can query for the last known state for a
+ particular device, even if it is not part of an active call. The
+ concept of a device in Asterisk can include entities that do not
+ have a physical representation. One way that this occurred was
+ when anonymous calls are allowed in Asterisk. A device was
+ automatically created and stored in the cache for each anonymous
+ call that occurred; this was possible in the SIP and IAX2 channel
+ drivers and through channel drivers that utilized the
+ res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif).
+ These devices are never removed from the system, allowing
+ anonymous calls to potentially exhaust a system's resources. This
+ patch changes the event cache subsystem and device state
+ management to no longer cache devices that are not associated
+ with a physical entity. (issue ASTERISK-20175) Reported by:
+ Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore
+ patches: event-cachability-3.diff uploaded by jcolp (license
+ 5000) ........ Merged revisions 378303 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c, main/http.c, res/res_jabber.c,
+ channels/sip/include/sip.h: Resolve crashes due to large stack
+ allocations when using TCP Asterisk had several places where
+ messages received over various network transports may be copied
+ in a single stack allocation. In the case of TCP, since multiple
+ packets in a stream may be concatenated together, this can lead
+ to large allocations that overflow the stack. This patch modifies
+ those portions of Asterisk using TCP to either favor heap
+ allocations or use an upper bound to ensure that the stack will
+ not overflow: * For SIP, the allocation now has an upper limit *
+ For HTTP, the allocation is now a heap allocation instead of a
+ stack allocation * For XMPP (in res_jabber), the allocation has
+ been eliminated since it was unnecesary. Note that the HTTP
+ portion of this issue was independently found by Brandon Edwards
+ of Exodus Intelligence. (issue ASTERISK-20658) Reported by:
+ wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches:
+ ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license
+ 5049) issueA20658_http_postvars_use_malloc2.patch uploaded by
+ wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch
+ uploaded by wdoekes (license 5674) ........ Merged revisions
+ 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-31 14:43 +0000 [r378218] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Ensure chan_sip rejects encrypted streams
+ without crypto info This ensures that Asterisk rejects encrypted
+ media streams (RTP/SAVP audio and video) that are missing
+ cryptographic keys and ensures that the incoming SDP is
+ consistent with RFC4568 as far as having a crypto attribute
+ present for any SAVP streams. Review:
+ https://reviewboard.asterisk.org/r/2204/ ........ Merged
+ revisions 378217 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-18 17:38 +0000 [r378120] Kinsey Moore <kmoore at digium.com>
+
+ * main/channel.c, /: Add test events for time limit-related hangups
+ This patch adds hangup-related test events in order to support
+ testing of time-limited bridges. This aids in testing the S() and
+ L() bridge options. (issue SWP-4713) ........ Merged revisions
+ 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-17 23:08 +0000 [r378089-378093] Richard Mudgett <rmudgett at digium.com>
+
+ * main/loader.c, /: Fix potential double free when unloading a
+ module. ........ Merged revisions 378092 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_local.c, /: Make chan_local module references tied
+ to local_pvt lifetime. The chan_local module references were
+ manually tied to the existence of the ;1 and ;2 channel links. *
+ Made chan_local module references tied to the existence of the
+ local_pvt structure as well as automatically take care of the
+ module references. * Tweaked the wording of the local_fixup()
+ failure warning message to make sense. Review:
+ https://reviewboard.asterisk.org/r/2181/ ........ Merged
+ revisions 378088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-14 21:30 +0000 [r378037] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/app_queue.c: app_queue: Revert bad ringinuse=no patch.
+ With the option ringinuse=no set, the patch committed for
+ ASTERISK-16115 causes non-SIP queue members to never be called
+ because the device state is checked after a channel is created to
+ determine if the member is busy. These queue members always get
+ the "Member %s is busy, cannot dial" message. Most channel
+ drivers other than chan_sip use the default device state
+ handling. The default device-state state is considered in use or
+ unknown if the channel exists or not respectively. (closes issue
+ ASTERISK-20801) Reported by: rmudgett Patches:
+ jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621)
+ patch uploaded by rmudgett ........ Merged revisions 378036 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-13 20:52 +0000 [r377992] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/confbridge/conf_state.c,
+ apps/confbridge/include/confbridge.h,
+ include/asterisk/bridging.h, apps/app_confbridge.c,
+ apps/confbridge/conf_state_multi_marked.c: confbridge: Fix MOH on
+ simultaneous user entry to a new conference. When two users
+ entered a new conference simultaneously, one of the callers hears
+ MOH. This happened if two unmarked users entered simultaneously
+ and also if a waitmarked and a marked user entered
+ simultaneously. * Created a confbridge internal MOH API to
+ eliminate the inlined MOH handling code. Note that the conference
+ mixing bridge needs to be locked when actually starting/stopping
+ MOH because there is a small window between the conference join
+ unsuspend MOH and actually joining the mixing bridge. * Created
+ the concept of suspended MOH so it can be interrupted while
+ conference join announcements to the user and DTMF features can
+ operate. * Suspend any MOH until the user is about to actually
+ join the mixing bridge of the conference. This way any pre-join
+ file playback does not need to worry about MOH. * Made post-join
+ actions only play deferred entry announcement files. Changing the
+ user/conference state during that time is not protected or
+ controlled by the state machine. (closes issue ASTERISK-20606)
+ Reported by: Eugenia Belova Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2232/
+
+2012-12-13 13:48 +0000 [r377947] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Ensure Min-SE is included in outbound
+ INVITEs Asterisk now includes Min-SE in outbound INVITEs when the
+ value is not 90 (the default) and session timers are not
+ disabled. This has the effect of Asterisk following RFC4028 more
+ closely with regard to 422 responses and preventing situations in
+ which Asterisk would be forced to temporarily accept a call to
+ tear it down based on a Session-Expires below the locally
+ configured Min-SE. (issue SWP-5051) Review:
+ https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey
+ Moore Patch-by: Kinsey Moore ........ Merged revisions 377946
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-12 22:41 +0000 [r377923] Rusty Newton <rnewton at digium.com>
+
+ * /, sounds/Makefile: Incremented EXTRA_SOUNDS_VERSION in
+ sounds/Makefile to 1.4.12 for new Extra Sounds releases See
+ CHANGES-* files in English extra 1.4.12 tarballs for new sound
+ prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan
+ (closes AST-755) Reported by: John Bigelow ........ Merged
+ revisions 377922 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-11 21:57 +0000 [r377848-377882] Richard Mudgett <rmudgett at digium.com>
+
+ * main/timing.c, main/channel.c, main/data.c, main/stun.c, /,
+ main/file.c, main/http.c, main/aoc.c, main/image.c, main/cel.c:
+ Cleanup CLI commands on exit for several files. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ unregister-cli-multiple-all.patch (license #5909) patch uploaded
+ by Corey Farrell ........ Merged revisions 377881 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/udptl.c, /: Cleanup udptl on exit. * Cleanup CLI commands on
+ exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches:
+ udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by
+ Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377847 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-11 20:48 +0000 [r377842] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_clialiases.c, /: Fix crash that can occur if CLI
+ registration fails for an aliased command. A recent memory leak
+ fix in main/cli.c causes an ast_cli_entry's command field to be
+ freed and NULLed if ast_cli_register() fails. res_clialiases was
+ ignoring the return value of ast_cli_register() and was then
+ passing the NULL command off to a a hash function. This resulted
+ in a crash. The fix is not to ignore the erroneous return value.
+ If ast_cli_register() fails, then we do not continue trying to
+ process the current alias. ........ Merged revisions 377840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-11 20:42 +0000 [r377705-377838] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/taskprocessor.c: Cleanup taskprocessor on exit. * Cleanup
+ CLI commands on exit. * v10 only: Merged v1.8 -r374177 change to
+ taskprocessor.c missed in v10 -r374178. (issue ASTERISK-20649)
+ Reported by: Corey Farrell Patches:
+ taskprocessor-cleanup-1_8-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell taskprocessor-cleanup-10-only.patch
+ (license #5909) patch uploaded by Corey Farrell Modified ........
+ Merged revisions 377837 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/pbx.c, /: Cleanup pbx on exit. * Cleanup CLI commands on
+ exit. * Unreference hints and statecbs containers on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey
+ Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by
+ Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377806 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/logger.c: Cleanup logger on exit. * Cleanup CLI commands,
+ destroy verbosers and logchannels lists on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ logger-cleanup-all.patch (license #5909) patch uploaded by Corey
+ Farrell Modified ........ Merged revisions 377771 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/indications.c: Cleanup indications on exit. * Made
+ ast_unregister_indication_country() unlink the found tone zone
+ before selecting a new default_tone_zone to make it impossible to
+ select the tone zone being unregistered again. * Ringcadence is
+ no longer parsed twice in store_config_tone_zone(). * Cleanup CLI
+ commands and destroy default_tone_zone on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ indications-cleanup-all.patch (license #5909) patch uploaded by
+ Corey Farrell Modified ........ Merged revisions 377740 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/event.c: Cleanup event on exit. * Cleanup CLI commands on
+ exit. * v10 only: Merged v1.8 -r374177 change to event.c missed
+ in v10 -r374178. (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: event_shutdown-10-only.patch (license #5909)
+ patch uploaded by Corey Farrell event_shutdown-1_8-11-trunk.patch
+ (license #5909) patch uploaded by Corey Farrell ........ Merged
+ revisions 377708 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/dnsmgr.c, /: Cleanup dnsmgr on exit. * Cleanup dnsmgr thread
+ and CLI commands on exit. (issue ASTERISK-20649) Reported by:
+ Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909)
+ patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch
+ (license #5909) patch uploaded by Corey Farrell Modified ........
+ Merged revisions 377704 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-10 16:53 +0000 [r377624-377656] Kinsey Moore <kmoore at digium.com>
+
+ * /, res/res_fax.c: Ensure ReceiveFax provides a CED tone via T.38
+ When using res_fax_digium, the T.38 CED tone was not being
+ provided properly which would cause some incoming faxes to fail.
+ This was not an issue with res_fax_spandsp since it does not
+ strictly honor the send_ced flag and sends the CED tone whenever
+ receiving a T.38 fax. (closes issue FAX-343) Reported-by:
+ Benjamin Tietz Patch-by: Kinsey Moore ........ Merged revisions
+ 377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Handle Session-Expires less than local
+ Min-SE in 200 OK Ensure that a call is immediately torn down if a
+ Session-Expires value received in a 200 OK is less than the local
+ Min-SE. This also prevents Asterisk from allowing calls with
+ Session-Expires below the RFC4028-mandated minimum (90s). (closes
+ issue ASTERISK-20653) Review:
+ https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore
+ ........ Merged revisions 377623 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-10 06:41 +0000 [r377558-377592] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c, /: Fix codec mismatch Fix code to send
+ in both rx and tx open stream messages correct codecs. Found that
+ on phase 0/1 phones wrong codecs cause to no audio in some
+ situations. (issue ASTERISK-20183) ........ Merged revisions
+ 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_unistim.c, /: Fix crash on transfer initiated from
+ insreeen menu on Unistim phones. Removed CDR-related code that
+ moved to do_masquarade before. (closes issue ASTERISK-20417)
+ Reported by: Rudolf Migalin ........ Merged revisions 377557 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.12.0-rc1 Released.
+
+2012-12-10 01:39 +0000 [r377504-377510] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * main/xmldoc.c, /: Improve documentation by making all of the
+ colors used readable, no matter what the background color is.
+ Dark blue on a black background is unreadable, as is yellow on a
+ light background. This patch turns on the bright attribute for
+ colors when on a dark background and turns *off* the bright
+ attribute when the -W command line option is used (indicating a
+ _light_ background). This ensures that text is readable in both
+ cases. Patch by: tilghman Review:
+ https://reviewboard.asterisk.org/r/2224 ........ Merged revisions
+ 377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, addons/cdr_mysql.c: Remove some dead code and additionally
+ handle a case that wasn't handled. ........ Merged revisions
+ 377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-08 00:29 +0000 [r377399-377432] Richard Mudgett <rmudgett at digium.com>
+
+ * contrib/realtime/mysql/sippeers.sql, /: Fix order of SIP
+ allow/disallow in MySQL contrib script. Using the contrib
+ sippeers.sql script to create the sippeers MySQL table would
+ result in being unable to place calls if you set the disallow
+ value to all. (closes issue ASTERISK-20756) Reported by: Andre
+ Luis Patches: sippeers.patch patch uploaded by Andre Luis
+ ........ Merged revisions 377431 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/astmm.c: MALLOC_DEBUG: Only wait if we want atexit
+ allocation dumps. ........ Merged revisions 377398 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-07 21:58 +0000 [r377382] Kinsey Moore <kmoore at digium.com>
+
+ * codecs/codec_dahdi.c: codec_dahdi: Fix output of "transcoder
+ show" CLI command. In r306010 "Asterisk media architecture
+ conversion - no more format bitfields", the logic for
+ incrementing encoders and decoders when opening transcoder
+ channels was changed without making the corresponding change when
+ decrementing encoder / decoder channels. The result being that
+ when a channel was destroyed, codec_dahdi couldn't properly tell
+ if it was an encoder or decoder, and the default case is to
+ assume it was a decoder. This could result in negative numbers
+ for decoders in use like in: VOIP6*CLI> transcoder show 2/-2
+ encoders/decoders of 92 channels are in use. (closes issue
+ ASTERISK-19921) Patch-by: Shaun Ruffell
+
+2012-12-06 23:56 +0000 [r377354] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/confbridge/conf_config_parser.c, apps/app_confbridge.c:
+ confbridge: Fix some resource leaks on conference teardown. *
+ Made destroy_conference_bridge() destroy a missed ast_mutex_t and
+ ast_cond_t. * Made join_conference_bridge() init the
+ ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can
+ destroy them unconditionally. * Made join_conference_bridge()
+ abort if the new conference could not be added to the conferences
+ container. * Made leave_conference() discard any post-join
+ actions if join_conference_bridge() had to abort early. * Made
+ the join_conference_bridge() diagnostic messages better describe
+ what happened. * Renamed leave_conference_bridge() to
+ leave_conference() and made it only take a conference user
+ pointer. The conference pointer was redundant. * Made
+ conf_bridge_profile_copy() use struct copy instead of memcpy(). *
+ No need to lock the conference in start_conf_record_thread()
+ since all of the callers already have it locked.
+
+2012-12-05 16:57 +0000 [r377261] Jonathan Rose <jrose at digium.com>
+
+ * res/res_srtp.c, /: res_srtp: Fix a crash caused by srtp_dealloc
+ on an already dealloced session When srtp_create fails, the
+ session may be dealloced or just not alloced. At the same time
+ though, the session pointer might not be set to NULL in this
+ process and attempting to srtp_dealloc it again will cause a
+ segfault. This patch checks for failure of srtp_create and sets
+ the session pointer to NULL if it fails. (closes issue
+ ASTERISK-20499) Reported by: tootai Review:
+ https://reviewboard.asterisk.org/r/2228/ ........ Merged
+ revisions 377256 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-05 16:49 +0000 [r377258] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Fix a SIP request memory leak with TLS
+ connections. During the TLS re-work in chan_sip some TLS specific
+ code was moved into a separate function. This function operates
+ on a copy of the incoming SIP request. This copy was never
+ deinitialized causing a memory leak for each request processed.
+ This function is now given a SIP request structure which it can
+ use to copy the incoming request into. This reduces the amount of
+ memory allocations done since the internal allocated components
+ are reused between packets and also ensures the SIP request
+ structure is deinitialized when the TLS connection is torn down.
+ (closes issue ASTERISK-20763) Reported by: deti ........ Merged
+ revisions 377257 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-05 02:09 +0000 [r377038-377241] Richard Mudgett <rmudgett at digium.com>
+
+ * main/format.c: * Fix registering core show codecs/codec CLI
+ commands twice. * Fix registering atexit format_attr_shutdown()
+ more than once.
+
+ * apps/confbridge/conf_config_parser.c: confbridge: Fix several
+ small issues. * Made func_confbridge_helper() allow an empty
+ value when setting options. You previously could not
+ Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the
+ dialplan. * Made func_confbridge_helper() handle its datastore
+ better if multiple threads attempt to set the first CONFBRIDGE
+ option value on the channel. * Made the func_confbridge_helper()
+ only output one diagnostic message concerning the option. * Made
+ the bridge video_mode able to repeatedly change in the config
+ file and CONFBRIDGE dialplan function. The video_mode option
+ values are an enum and not independent of each other. * Made
+ handle_cli_confbridge_show_bridge_profile() better handle the
+ video_mode option. * Simplified datastore handling code in
+ conf_find_user_profile() and conf_find_bridge_profile(). * Made
+ parse_bridge(), parse_user(), and parse_menu() use var->file
+ instead of CONFBRIDGE_CONFIG because the var could have been from
+ an include file. (closes issue ASTERISK-20655) Reported by:
+ Birger "WIMPy" Harzenetter
+
+ * apps/app_confbridge.c: confbridge: Update online XML
+ documentation.
+
+ * /, main/asterisk.c: Cleanup ast_run_atexits() atexits list. *
+ Convert atexits list to a mutex instead of a rd/wr lock. The lock
+ is only write locked. * Move CLI verbose Asterisk ending message
+ to where AMI message is output in really_quit() to avoid further
+ surprises about using stuff already shutdown. (issue
+ ASTERISK-20649) Reported by: Corey Farrell ........ Merged
+ revisions 377165 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, include/asterisk/_private.h, main/stdtime/localtime.c,
+ main/asterisk.c: Cleanup core main on exit. * Cleanup time zones
+ on exit. * Make exit clean/unclean report consistent for AMI and
+ CLI in really_quit(). (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: core-cleanup-1_8-10.patch (license #5909) patch
+ uploaded by Corey Farrell core-cleanup-11-trunk.patch (license
+ #5909) patch uploaded by Corey Farrell Modified ........ Merged
+ revisions 377135 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/config.c: Cleanup config cache on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ config-cleanup-all.patch (license #5909) patch uploaded by Corey
+ Farrell ........ Merged revisions 377104 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/cli.c, /: Cleanup CLI resources on exit and CLI command
+ registration errors. (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch
+ uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
+ #5909) patch uploaded by Corey Farrell Modified ........ Merged
+ revisions 377073 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/cdr.c, /: Cleanup CDR resources on exit. * Simplify
+ do_reload() return handling since it never returned anything
+ other than 0. (issue ASTERISK-20649) Reported by: Corey Farrell
+ Patches: cdr-cleanup-1_8.patch (license #5909) patch uploaded by
+ Corey Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377069 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/ccss.c: Fix CCSS CLI commands and logger level not
+ unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell
+ Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by
+ Corey Farrell ........ Merged revisions 377037 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-30 21:33 +0000 [r376951] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/misdn/isdn_lib.c: chan_misdn: Fix sending
+ RELEASE_COMPLETE in response to SETUP. Fix sending a
+ RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
+ have a B channel available to assign to the call. (closes issue
+ ABE-2869) Reported by: Guenther Kelleter Patches:
+ setup-reject_2.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified ........ Merged revision 376949 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 376950 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-30 17:06 +0000 [r376920] Sean Bright <sean at malleable.com>
+
+ * /, funcs/func_volume.c: Minor spelling fix to the VOLUME
+ documentation. ........ Merged revisions 376919 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-30 16:23 +0000 [r376916] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Fix potential crashes during SIP attended
+ transfers. The principal behind this patch is simple. During a
+ transfer, we manipulate channels that are owned by a separate
+ thread than the one we currently are running in, so it makes
+ sense that we need to grab a reference to the channels so that
+ they cannot disappear out from under us. In the wild, crashes
+ were sometimes seen when the transferring party would hang up the
+ call before the transfer target answered the call. The most
+ common place to see the crash occur was when attempting to send a
+ connected line update to the transferer channel. (closes issue
+ ASTERISK-20226) Reported by Jared Smith Patches:
+ ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
+ Tested by: Jared Smith ........ Merged revisions 376901 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-29 22:58 +0000 [r376865-376869] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_local.c, /: chan_local: Fix local_pvt ref leak in
+ local_devicestate(). Regression introduced by ASTERISK-20390 fix.
+ (closes issue ASTERISK-20769) Reported by: rmudgett Tested by:
+ rmudgett ........ Merged revisions 376868 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)
+ ........ Merged revisions 376864 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-29 21:51 +0000 [r376835] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, channels/chan_sip.c: Improve Code Readability And Fix Setting
+ natdetected Flag For 1.8, 10, 11 and trunk we are are improving
+ the code readability. For 11 and trunk, auto nat detection was
+ added. The natdetected flag was being set to 1 when the host
+ address in the VIA header did not specifiy a port. This patch
+ fixes this by setting the port on the temporary sock address used
+ to SIP_STANDARD_PORT in order for the sock address comparison to
+ work properly. (closes issue ASTERISK-20724) Reported by: Michael
+ L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2206/ ........ Merged
+ revisions 376834 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-29 00:45 +0000 [r376759-376789] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/astmm.c, main/asterisk.c: Add MALLOC_DEBUG atexit
+ unreleased malloc memory summary. * Adds the following CLI
+ commands to control MALLOC_DEBUG reporting of unreleased malloc
+ memory when Asterisk is shut down. memory atexit list on memory
+ atexit list off memory atexit summary byline memory atexit
+ summary byfunc memory atexit summary byfile memory atexit summary
+ off * Made check all remaining allocated region blocks atexit for
+ fence violations. * Increased the allocated region hash table
+ size by about three times. It still isn't large enough
+ considering the number of malloced blocks Asterisk uses. * Made
+ CLI "memory show allocations anomalies" use
+ regions_check_all_fences(). Review:
+ https://reviewboard.asterisk.org/r/2196/ ........ Merged
+ revisions 376788 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/astmm.c: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI
+ "memory show allocations" misspelling of anomalies option. The
+ command will still accept the original misspelling. *
+ Miscellaneous tweaks to CLI "memory show allocations" command
+ output format. * Made CLI "memory show summary" summarize by line
+ number instead of by function if a filename is given. * Made CLI
+ "memory show summary" sort its output by filename or
+ function-name/line-number depending upon request. * Miscellaneous
+ tweaks to CLI "memory show summary" command output format.
+ ........ Merged revisions 376758 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-28 16:30 +0000 [r376726] Jonathan Rose <jrose at digium.com>
+
+ * main/manager.c, /: manager: Make challenge work with
+ allowmultiplelogin=no Prior to this patch, challenge would yield
+ a multiple logins error if used without providing the username
+ (which isn't really supposed to be an argument to challenge) if
+ allowmultiplelogin was set to no because allowmultiplelogin finds
+ a user with a zero length login name. This check is simply
+ disabled for the challenge action when the username is empty by
+ this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
+ Patches: challenge_action_nomultiplelogin.diff uploaded by
+ Jonathan Rose (license 6182) ........ Merged revisions 376725
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-27 23:58 +0000 [r376628-376689] Richard Mudgett <rmudgett at digium.com>
+
+ * main/pbx.c, /, UPGRADE.txt: Fix extension matching with the '-'
+ char. The '-' char is supposed to be ignored by the dialplan
+ extension matching. Unfortunately, it's treatment is not handled
+ consistently throughout the extension matching code. * Made the
+ old exten matching code consistently ignore '-' chars. * Made the
+ old exten matching code consistently handle case in the matching.
+ * Made ignore empty character sets. * Fixed ast_extension_cmp()
+ to return -1, 0, or 1 as documented. The only user of it in
+ pbx_lua.c was testing for -1. It was originally returning the
+ strcmp() value for less than which is not usually going to be -1.
+ * Fix character set sorting if the sets have the same number of
+ characters and start with the same character. Character set [0-9]
+ now sorts before [02-9a] as originally intended. * Updated some
+ extension label and priority already in use warnings to also
+ indicate if the extension is aliased. (closes issue
+ ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
+ Harzenetter Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2201/ ........ Merged
+ revisions 376688 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * addons/res_config_mysql.c, /, apps/app_celgenuserevent.c,
+ pbx/pbx_dundi.c: Remove unnecessary channel module references. *
+ Removed call to ast_module_user_hangup_all() in
+ res_config_mysql.c since it is effectively a noop. No channels
+ can attach a reference to that module. * Removed call to
+ ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
+ of unload_module() has already called it. * Removed redundant
+ channel module references in pbx_dundi.c. The registered dialplan
+ function callback dispatchers for the read/read2/write callbacks
+ already reference the module before calling. * pbx_dundi: Moved
+ unregistering CLI commands, DUNDi switch, and dialplan functions
+ to the first thing the unload_module() does. This will reduce the
+ chance of new channels using DUNDi services while the module is
+ being torn down. ........ Merged revisions 376657 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, include/asterisk/linkedlists.h: Made AST_LIST_REMOVE() simpler
+ and use better names. * Update doxygen of AST_LIST_REMOVE().
+ ........ Merged revisions 376627 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-22 23:56 +0000 [r376587] Matthew Jordan <mjordan at digium.com>
+
+ * main/lock.c, /, main/logger.c, include/asterisk/lock.h:
+ Re-initialize logmsgs mutex upon logger initialization to prevent
+ lock errors Similar to the patch that moved the fork earlier in
+ the startup sequence to prevent mutex errors in the recursive
+ mutex surrounding the read/write thread registration lock, this
+ patch re-initializes the logmsgs mutex. Part of the start up
+ sequence before forking the process into the background includes
+ reading asterisk.conf; this has to occur prior to the call to
+ daemon in order to read startup parameters. When reading in a
+ conf file, log statements can be generated. Since this can't be
+ avoided, the mutex instead is re-initialized to ensure a reset of
+ any thread tracking information. This patch also includes some
+ additional debugging to catch errors when locking or unlocking
+ the recursive mutex that surrounds locks when the DEBUG_THREADS
+ build option is enabled. DO_CRASH or THREAD_CRASH will cause an
+ abort() if a mutex error is detected. (issue ASTERISK-19463)
+ Reported by: mjordan Tesetd by: mjordan ........ Merged revisions
+ 376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-20 17:01 +0000 [r376522] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Add "Require:
+ timer" to 200 OK responses when appropriate. The method by which
+ the Require header is added to 200 responses is inspired by the
+ method that Olle Johansson uses in his darjeeling-prack branch.
+ (closes issue ASTERISK-20570) Reported by Matt Jordan, at the
+ behest of Olle Johansson Review:
+ https://reviewboard.asterisk.org/r/2172 ........ Merged revisions
+ 376521 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-19 19:44 +0000 [r376470] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c, main/security_events.c,
+ main/indications.c: Fix most leftover non-opaque ast_str uses.
+ Instead of calling str->str, one should use ast_str_buffer(str).
+ Same goes for str->used as ast_str_strlen(str) and str->len as
+ ast_str_size(str). Review:
+ https://reviewboard.asterisk.org/r/2198 ........ Merged revisions
+ 376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-18 20:18 +0000 [r376414-376431] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/utils.c, main/stdtime/localtime.c, main/asterisk.c:
+ Reorder startup sequence to prevent lockups when process is sent
+ to background Although it is very rare and timing dependent, the
+ potential exists for the call to 'daemon' to cause what appears
+ to be a deadlock in Asterisk during startup. This can occur when
+ a recursive mutex is obtained prior to the daemon call executing.
+ Since daemon uses fork to send the process into the background,
+ any threading primitives are unsafe to re-use after the call.
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