[svn-commits] bebuild: tag 11.8.0-rc1 r405532 - /tags/11.8.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jan 14 13:37:51 CST 2014
Author: bebuild
Date: Tue Jan 14 13:37:48 2014
New Revision: 405532
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=405532
Log:
Importing files for 11.8.0-rc1 release.
Added:
tags/11.8.0-rc1/.lastclean (with props)
tags/11.8.0-rc1/.version (with props)
tags/11.8.0-rc1/ChangeLog (with props)
Added: tags/11.8.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/11.8.0-rc1/.lastclean?view=auto&rev=405532
==============================================================================
--- tags/11.8.0-rc1/.lastclean (added)
+++ tags/11.8.0-rc1/.lastclean Tue Jan 14 13:37:48 2014
@@ -1,0 +1,1 @@
+40
Propchange: tags/11.8.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/11.8.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/11.8.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/11.8.0-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/11.8.0-rc1/.version?view=auto&rev=405532
==============================================================================
--- tags/11.8.0-rc1/.version (added)
+++ tags/11.8.0-rc1/.version Tue Jan 14 13:37:48 2014
@@ -1,0 +1,1 @@
+11.8.0-rc1
Propchange: tags/11.8.0-rc1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/11.8.0-rc1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/11.8.0-rc1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/11.8.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/11.8.0-rc1/ChangeLog?view=auto&rev=405532
==============================================================================
--- tags/11.8.0-rc1/ChangeLog (added)
+++ tags/11.8.0-rc1/ChangeLog Tue Jan 14 13:37:48 2014
@@ -1,0 +1,28308 @@
+2014-01-14 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.8.0-rc1 Released.
+
+2014-01-14 18:43 +0000 [r405434-405487] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: No BYE message sent after
+ INVITE with Replaces Setting channel state DOWN is an unnecessary
+ step that was only being done in handle_invite_replaces(). This
+ changes that by removing the call and reducing locking. (closes
+ issue ASTERISK-23010) Reported by: Ryan Tilton Review:
+ https://reviewboard.asterisk.org/r/3116/ ........ Merged
+ revisions 405486 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: chan_sip: fix Local From tag on outbound
+ register regression In ASTERISK-12117, an improvement to insure
+ consistant local from tags on outbound registrations resulted in
+ an undesirable behavior - caused by leftover unexpired sip_pvt
+ dialogs (with the previous cseq number), resulting in many
+ uncessary REGISTER requests. Instead of significant rework of
+ transmit_register(), this change deletes the dialogs after a 200
+ OK response indiciating a successful registration, keeping the
+ old dialogs from interfering with normal operation. (closes issue
+ ASTERISK-22946) Reported by: Stephan Eisvogel Review:
+ https://reviewboard.asterisk.org/r/3109/ ........ Merged
+ revisions 405433 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-14 17:26 +0000 [r405431] Richard Mudgett <rmudgett at digium.com>
+
+ * main/asterisk.c, configs/logger.conf.sample, main/cli.c,
+ include/asterisk/logger.h, main/pbx.c, main/manager.c,
+ funcs/func_timeout.c, apps/app_dumpchan.c, main/logger.c,
+ UPGRADE.txt, apps/app_verbose.c: verbosity: Fix performance of
+ console verbose messages. The per console verbose level feature
+ as previously implemented caused a large performance penalty. The
+ fix required some minor incompatibilities if the new rasterisk is
+ used to connect to an earlier version. If the new rasterisk
+ connects to an older Asterisk version then the root console
+ verbose level is always affected by the "core set verbose"
+ command of the remote console even though it may appear to only
+ affect the current console. If an older version of rasterisk
+ connects to the new version then the "core set verbose" command
+ will have no effect. * Fixed the verbose performance by not
+ generating a verbose message if nothing is going to use it and
+ then filtered any generated verbose messages before actually
+ sending them to the remote consoles. * Split the "core set debug"
+ and "core set verbose" CLI commands to remove the per module
+ verbose support that cannot work with the per console verbose
+ level. * Added a silent option to the "core set verbose" command.
+ * Fixed "core set debug off" tab completion. * Made "core show
+ settings" list the current console verbosity in addition to the
+ root console verbosity. * Changed the default verbose level of
+ the 'verbose' setting in the logger.conf [logfiles] section. The
+ default is now to once again follow the current root console
+ level. As a result, using the AMI Command action with "core set
+ verbose" could again set the root console verbose level and
+ affect the verbose level logged. (closes issue AST-1252) Reported
+ by: Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/3114/
+
+2014-01-14 15:32 +0000 [r405362-405380] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Hangup transferer/transferee when
+ transfer to Parking fails When performing a SIP transfer to a
+ Park extension, if the Park fails, chan_sip will currently not
+ hang up either the transferer or the transfer target. This
+ results in the channels being orphaned with no thread to service
+ frames, resulting in stuck channels. This patch immediately hangs
+ up the two channels if a Park fails. (closes issue
+ ASTERISK-22834) Reported by: rsw686 Tested by: rsw686 (closes
+ issue ASTERISK-23047) Reported by: Tommy Thompson Tested by:
+ Tommy Thomspon Review: https://reviewboard.asterisk.org/r/3107
+
+ * res/Makefile: res/Makefile: alias dist-clean to distclean A 'make
+ distclean' is equivalent to 'make dist-clean' in the top most
+ Makefile. This patch updates the res/Makefile to recognize both
+ distclean and dist-clean. Note that this is needed for removing
+ build.mak, which can run into problems if the source file of
+ Asterisk or its path is changed after build.mak is generated.
+ (issue ASTERISK-22480) Reported by: Matt Jordan
+
+2014-01-10 17:50 +0000 [r405281] Richard Mudgett <rmudgett at digium.com>
+
+ * main/logger.c: Logging callid: Fix some sizeof() references per
+ coding guidelines.
+
+2014-01-09 16:49 +0000 [r405234] Kevin Harwell <kharwell at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fails to resume WebRTC
+ call from hold In ast_rtp_ice_start if the ice session create
+ check list failed, start check was never initiated and
+ ice_started was never set to true. Upon re-entering the function
+ (for instance, [un]hold) it would try to create the check list
+ again with duplicate remote candidates. Fixed so that if the
+ create check list fails the necessary data structures are
+ properly re-initialized for any subsequent retries. Note, it was
+ decided to not stop ice support (by calling ast_rtp_ice_stop) on
+ a check list failure because it possible things might still work.
+ However, a debug message was added to help with any future
+ troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis
+ ValentinaviÄius Patches: works_on_my_machine.patch uploaded by
+ xytis (license 6558)
+
+2014-01-09 15:41 +0000 [r405215] Matthew Jordan <mjordan at digium.com>
+
+ * apps/confbridge/conf_state_multi_marked.c, apps/app_confbridge.c:
+ app_confbridge: Fix crash caused when waitmarked/marked users
+ leave together When waitmarked users join a ConfBridge, the
+ conference state is transitioned from EMPTY -> INACTIVE. In this
+ state, the users are maintined in a waiting users list. When a
+ marked user joins, the ConfBridge conference transitions from
+ INACTIVE -> MULTI_MARKED, and all users are put onto the active
+ list of users. This process works correctly. When the marked user
+ leaves, if they are the last marked user, the MULTI_MARKED state
+ does the following: (1) It plays back a message to the bridge
+ stating that the leader has left the conference. This requires an
+ unlocking of the bridge. (2) It moves waitmarked users back to
+ the waiting list (3) It transitions to the appropriate state: in
+ this case, INACTIVE However, because it plays the prompt back to
+ the bridge before moving the users and before finishing the state
+ transition, this creates a race condition: with the bridge
+ unlocked, waitmarked users who leave the conference (or are
+ kicked from it) can cause a state transition of the bridge to
+ another state before the conference is transitioned to the
+ INACTIVE state. This causes the state machine to get a bit wonky,
+ often leading to a crash when the MULTI_MARKED state attempts to
+ conclude its processing. This patch fixes this problem: (1) It
+ prevents kicked users from being kicked again. That's just a
+ nicety. (2) More importantly, it fixes the race condition by only
+ playing the prompt once the state has transitioned correctly to
+ INACTIVE. If waitmarked users sneak out during the prompt being
+ played, no harm no foul. Review:
+ https://reviewboard.asterisk.org/r/3108/ (closes issue AST-1258)
+ Reported by: Steve Pitts
+
+2014-01-09 14:12 +0000 [r405161] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * apps/app_dumpchan.c, /: "Minimun" typo. ........ Merged revisions
+ 405160 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-08 16:17 +0000 [r405081-405091] Kinsey Moore <kmoore at digium.com>
+
+ * /, configure, configure.ac, pbx/pbx_lua.c: pbx_lua: Add support
+ for Lua 5.2 This adds support for Lua 5.2 in pbx_lua which is
+ available on newer operating systems. (closes issue
+ ASTERISK-23011) Review: https://reviewboard.asterisk.org/r/3075/
+ Reported by: George Joseph Patch by: George Joseph ........
+ Merged revisions 405090 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * UPGRADE.txt, /: UPGRADE: Add a note about non-functionality Add a
+ note that the "retry on 403 response to REGISTER" for chan_sip is
+ non-functional in the versions in which it was first introduced.
+ ........ Merged revisions 405088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Add the missing part of r400140 When the
+ patch to add retry-on-forbidden-response was committed, part of
+ the patch for chan_sip was not committed which caused the feature
+ to be entirely nonfunctional. This corrects the code in question.
+ (closes issue ASTERISK-17138) Review:
+ https://reviewboard.asterisk.org/r/2874 ........ Merged revisions
+ 405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-03 22:24 +0000 [r404888] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * main/asterisk.c, /: asterisk.c: suppress live_dangerously warning
+ on rasterisk Even since the fixes of AST-2013-007, Asterisk
+ prints the following warning on startup if the user decided to
+ live dangerously: Privilege escalation protection disabled! See
+ https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This
+ message is intended for the logs and interactive startup. No need
+ for it to appear on a remote console. This commit removes it from
+ there. (closes issue ASTERISK-23084) Review:
+ https://reviewboard.asterisk.org/r/3101/ ........ Merged
+ revisions 404861 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-03 21:58 +0000 [r404773-404858] Kevin Harwell <kharwell at digium.com>
+
+ * cel/cel_pgsql.c, /: cel_pgsql: module not correctly reloading
+ Upon reload the module unconditionally "unloaded" the module
+ (freeing memory and setting pointers to NULL) and then when
+ attempting a "load" if the config file had not changed then
+ nothing would be reinitialized. By moving the "unload" to occur
+ conditionally (reload only) after an attempted configuration
+ load, but before module "loading" alleviates the issue. The
+ module now loads/unloads/reloads correctly. (closes issue
+ ASTERISK-22871) Reported by: Matteo ........ Merged revisions
+ 404857 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /: chan_dahdi: dahdi show channels slices
+ PRI channel dnid on output dahdi show channels output slices the
+ callerid (which is dnid copied over on PRI channels). If the
+ channel naming structures look like: 'DAHDI/i1/1408409XXXX-6'
+ then the output slices 1408409XXXX down to 1408409XXX. This patch
+ just opens it up to 15 chars so you can see the whole thing.
+ (closes issue ASTERISK-22918) Reported by: outtolunc Patches:
+ svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc
+ (license 5198) ........ Merged revisions 404784 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_meetme.c: app_meetme: compiler warning Fixed a
+ compiler warning (errors in 'dev-mode') given by gcc version
+ 4.8.1. The one in app_meetme involved the
+ 'sizeof-pointer-memaccess' (see:
+ http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. Fixed so it
+ would no longer issue a warning and can compile again in
+ 'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/
+ ........ Merged revisions 404742 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-01-02 19:35 +0000 [r404675] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * /, funcs/func_strings.c: func_strings: use memmove to prevent
+ overlapping memory on strcpy When calling REPLACE() with an empty
+ replace-char argument, strcpy is used to overwrite the the
+ matching <find-char>. However as the src and dest arguments to
+ strcpy must not overlap, it causes other parts of the string to
+ be overwritten with adjacent characters and the result is
+ mangled. Patch replaces call to strcpy with memmove and adds a
+ test suite case for REPLACE. (closes issue ASTERISK-22910)
+ Reported by: Gareth Palmer Review:
+ https://reviewboard.asterisk.org/r/3083/ Patches:
+ func_strings.patch uploaded by Gareth Palmer (license 5169)
+ ........ Merged revisions 404674 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-31 21:26 +0000 [r404579-404604] Kevin Harwell <kharwell at digium.com>
+
+ * /, cel/cel_pgsql.c: cel_pgsql: deadlock on unload and
+ core_event_dispatcher A deadlock can happen between a thread
+ unloading or reloading the cel_pgsql module and the
+ core_event_dispatcher taskprocessor thread. Description of what
+ is happening: Thread 1 (for example, a netconsole thread): a
+ "module reload cel_pgsql" is launched the thread enter the
+ "my_unload_module" function (cel_pgsql.c) the thread acquire the
+ write lock on psql_columns the thread enter the
+ "ast_event_unsubscribe" function (event.c) the thread try to
+ acquire the write lock on ast_event_subs[sub->type] Thread 2
+ (core_event_dispatcher taskprocessor thread): the taskprocessor
+ pop a CEL event the thread enter the "handle_event" function
+ (event.c) the thread acquire the read lock on
+ ast_event_subs[sub->type] the thread callback the "pgsql_log"
+ function (cel_pgsql.c), since it's a subscriber of CEL events the
+ thread try to acquire a read lock on psql_columns (closes issue
+ ASTERISK-22854) Reported by: Etienne Lessard Patches:
+ cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license
+ 6394) ........ Merged revisions 404603 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/channel.c: channels.c: core show channeltypes slicing 'core
+ show channeltypes' type column is being sliced, resulting in
+ incomplete type names. (closes issue ASTERISK-22919) Reported by:
+ outtolunc Patches: svn_channel.c.format_15.diff.txt uploaded by
+ outtolunc (license 5198)
+
+2013-12-20 21:15 +0000 [r404457] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * /, main/say.c: say.c: correct time for polish In
+ ast_say_date_with_format_pl(), change ast_say_number() to use
+ tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported
+ by: Robert Mordec Review:
+ https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch
+ uploaded by veilen (license 6555) ........ Merged revisions
+ 404456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-19 16:57 +0000 [r404344-404351] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * res/res_fax.c: res_fax.c: crash on framehook with no dsp in fax
+ detect In fax_detect_framehook() a null pointer reference can
+ occur where a voice frame is processed but no dsp is attached to
+ the fax detection structure. The code block that rejects frames
+ that detection cannot be processed on is checking for dsp but
+ falls through when it should instead return, as this change
+ implements. (closes issue ASTERISK-22942) Reported by: adomjan
+ Review: https://reviewboard.asterisk.org/r/3076/
+
+ * main/db.c: astdb: crash in sqlite3 during shutdown When Asterisk
+ is shut down, the astdb_atexit() function releases (finalize) the
+ previously initiated (prepared) SQL statements in sqlite3.
+ Another thread making a subsequent request can cause a crash in
+ sqlite3. This patch eliminates that issue by resetting the
+ statement pointer after it is released/cleared. The sqlite3 code
+ detects the null pointer, and aborts the operation cleanly.
+ (closes issue AST-1265) Reported by: Alexander Hömig (closes
+ issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter
+ Review: https://reviewboard.asterisk.org/r/3078/
+
+2013-12-19 08:15 +0000 [r404318] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooGkClient.h, addons/ooh323c/src/oochannels.c:
+ Handle temporary failures on gk registration Introduce new
+ 'stopped' state for gk client and restart gk client on failures
+ Remove ooh323 stack command lock as it is not need now. (closes
+ issue ASTERISK-21960) Reported by: Dmitry Melekhov Patches:
+ ASTERISK-21960.patch ASTERISK-21960-stacklockup-2.patch Tested
+ by: Dmitry Melekhov
+
+2013-12-18 22:34 +0000 [r404275] Jason Parker <jparker at digium.com>
+
+ * main/manager.c: Add AMI event for presence state. Review:
+ https://reviewboard.asterisk.org/r/3039/
+
+2013-12-18 20:19 +0000 [r404219] Richard Mudgett <rmudgett at digium.com>
+
+ * addons/ooh323c/src/ooTimer.c, /: ooh323c: Fix gcc 4.6.3 compiler
+ warnings. ........ Merged revisions 404212 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-18 11:59 +0000 [r404136] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_calendar.c: res_calendar: Protect channel when adding
+ datastore. This change adds a missing channel lock when adding a
+ datastore to a channel. ........ Merged revisions 404135 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-18 00:28 +0000 [r404045-404087] Rusty Newton <rnewton at digium.com>
+
+ * /, funcs/func_strings.c: func_strings: Documentation fix for
+ QUOTE() Example output was inaccurate. (issue ASTERISK-22970)
+ (closes issue ASTERISK-22970) Reported by: Gareth Palmer Patches:
+ func_strings.patch uploaded by Gareth Palmer (license 5169)
+ ........ Merged revisions 404081 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/rtp_engine.c, channels/chan_iax2.c, apps/app_chanspy.c,
+ apps/app_mixmonitor.c, include/asterisk/test.h, main/channel.c:
+ Several components: fixing Typos in comments and code,
+ "avaliable" instead of "available" (issue ASTERISK-23021) (closes
+ issue ASTERISK-23021) Reported by: Jeremy Lainé Tested by: Rusty
+ Newton Patches: available.patch uploaded by Jeremy Lainé (license
+ 6561)
+
+2013-12-16 17:14 +0000 [r403917] David M. Lee <dlee at digium.com>
+
+ * funcs/func_realtime.c, main/pbx.c, main/tcptls.c,
+ funcs/func_db.c, /, README-SERIOUSLY.bestpractices.txt,
+ configs/asterisk.conf.sample, funcs/func_shell.c,
+ funcs/func_env.c, funcs/func_lock.c, UPGRADE.txt,
+ include/asterisk/pbx.h, main/asterisk.c: security: Inhibit
+ execution of privilege escalating functions This patch allows
+ individual dialplan functions to be marked as 'dangerous', to
+ inhibit their execution from external sources. A 'dangerous'
+ function is one which results in a privilege escalation. For
+ example, if one were to read the channel variable SHELL(rm -rf /)
+ Bad Things(TM) could happen; even if the external source has only
+ read permissions. Execution from external sources may be enabled
+ by setting 'live_dangerously' to 'yes' in the [options] section
+ of asterisk.conf. Although doing so is not recommended. (closes
+ issue ASTERISK-22905) Review:
+ http://reviewboard.digium.internal/r/432/ ........ Merged
+ revisions 403913 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-16 15:55 +0000 [r403855-403863] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/pbx.c, /: pbx.c: put copy of ast_exten.data on stack to
+ prevent memory corruption During dialplan execution in
+ pbx_extension_helper(), the contexts global read lock prevents
+ link list corruption, but was released with a pointer to the
+ ast_exten and data later used in variable substitution. Instead,
+ this patch removes pbx_substitute_variables() and locates a copy
+ of the ast_exten data on the stack before releasing the lock,
+ where ast_exten could get free'd by another thread performing a
+ module reload. (issue AST-1179) Reported by: Thomas Arimont
+ (issue AST-1246) Reported by: Alexander Hömig Review:
+ https://reviewboard.asterisk.org/r/3055/ ........ Merged
+ revisions 403862 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_sms.c: app_sms: BufferOverflow when receiving odd
+ length 16 bit message This patch prevents an infinite loop
+ overwriting memory when a message is received into the
+ unpacksms16() function, where the length of the message is an odd
+ number of bytes. (closes issue ASTERISK-22590) Reported by: Jan
+ Juergens Tested by: Jan Juergens ........ Merged revisions 403853
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-11 19:14 +0000 [r403635] Russell Bryant <russell at russellbryant.com>
+
+ * /, channels/chan_sip.c: Reset peer outboundproxy on sip.conf
+ reload If you set a peer's outboundproxy and then removed it from
+ the config, this would not get picked up in a config reload. This
+ patch fixes that by resetting it in set_peer_defaults(). Closes
+ ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/
+ ........ Merged revisions 403634 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-09 03:11 +0000 [r403450] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_fax_spandsp.c, /: res_fax_spandsp: Always init T.38
+ session to avoid crashes during state change Prior to this patch,
+ res_fax_spandsp was conservative with how it initialized the
+ spandsp T.38 context. It would only initialize it if the driver
+ thought the current state was a T.38 fax. While this works fine
+ in nominal situations, in certain off nominal situations,
+ res_fax_spandsp can believe that a T.38 fax will not occur when
+ in fact one has started. In particular, this was discovered when
+ res_fax would fall back to audio after timing out on a T.38
+ upgrade. The SIP channel driver would continue to retry the
+ re-INVITE and - if the remote end responded after res_fax timed
+ out with a 200 OK - a T.38 frame would be delivered to the
+ res_fax stack when it no longer expected it. As it turns out,
+ there does not appear to be any downside to always initializing
+ the T.38 context, other than the actual memory allocation. Since
+ that avoids this off nominal situation (and others which are
+ equally likely hard to predict), this is the safest way to avoid
+ this problem. Much thanks to Torrey as well for providing a
+ scenario that reproduces this issue. (closes issue
+ ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey
+ Searle patches: always-init-t38.patch uploaded by awinters
+ (License 6477) A_PARTY.xml uploaded by tsearle (License 5334)
+ ........ Merged revisions 403449 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-12-02 17:55 +0000 [r403288] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: Check and reject non-digits e164 values on
+ peers and general sections in ooh323.conf Regenerate e164
+ endpoint list on reload ooh323 (issue ASTERISK-22901) Reported
+ by: Cyril CONSTANTIN Patches: ASTERISK-22901.patch
+
+2013-11-22 17:11 +0000 [r403015] Joshua Colp <jcolp at digium.com>
+
+ * /, main/translate.c: translate: Move freeing of frame to after it
+ is used. When translating from one format to another it is
+ possible to inform the translation function that the source frame
+ should be freed. This was previously done immediately but shortly
+ afterwards the frame that was freed was accessed and used again.
+ This change moves code around a bit so that the frame is now
+ freed after it has been completely used. (closes issue
+ ASTERISK-22788) Reported by: Corey Farrell Patches:
+ translate-access-after-free-11up.patch uploaded by coreyfarrell
+ (license 5909) translate-access-after-free-1.8.patch uploaded by
+ coreyfarrell (license 5909) ........ Merged revisions 403014 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-11-12 15:00 +0000 [r402709] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_dahdi.c, /: chan_dahdi: Fix crash during caller ID
+ read Asterisk will sometimes core dump during caller id read on
+ analog channels due to a negative return value from the read() in
+ my_get_callerid that slips through as a negative length argument
+ to callerid_feed() if the errno returned by DAHDI is ELAST. This
+ change ensures that the negative return is treated properly even
+ when it is ELAST. (closes issue ASTERISK-22746) Reported by:
+ Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch
+ uploaded by Michael Walton (License 6502) ........ Merged
+ revisions 402708 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-11-11 19:26 +0000 [r402686] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_confbridge.c: Get rid of some inaccurate comments. I'm
+ doing some unrelated work in app_confbridge and finding these
+ "invalid pin" comments to be annoying. Get out!
+
+2013-11-11 15:35 +0000 [r402646] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_queue.c: app_queue: Honor penalty limits of 0 In the
+ current app_queue code from 1.8 up to trunk the upper and lower
+ penalties can be set to 0 but the value is interpreted to be
+ disabled instead of actually setting limits. This is especially
+ evident if min and max limits are set to 0 and members with
+ penalties of 0 and 1 are in the queue since the member with
+ penalty 1 will still receive calls. This patch adjusts the
+ special disabled value to be INT_MAX instead of 0. (closes issue
+ ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/
+ Reported by: Schmooze Com ........ Merged revisions 402645 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-11-08 22:48 +0000 [r402605] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ keep same local (from) tag for outgoing register requests For
+ outbound register requests the tag on the From line was updated
+ every 20 seconds prior to a successful registration and also once
+ for each registration renewal. That behavior can possibly cause
+ the registration to be denied because of the different tag, and
+ is not aligned with the intention of RFC 3261 8.1.3.5 "...
+ request constitutes a new transaction and SHOULD have the same
+ value of the Call-ID, To, and From of the previous request...".
+ This updates chan_sip to have a field to keep the local tag in
+ the registration structure and use that tag for registration
+ requests where the callid is also unchanged. (closes issue
+ ASTERISK-12117) Reported by: Pawel Pierscionek Review:
+ https://reviewboard.asterisk.org/r/2988/ ........ Merged
+ revisions 402604 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-11-05 15:11 +0000 [r402450-402469] Kevin Harwell <kharwell at digium.com>
+
+ * /: Recorded merge of revisions 402468 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+ chan_sip: notify dialog info ignores presentation indicator in
+ callerid The presentation indicator in a callerid (e.g. set by
+ dialplan function Set(CALLERID(name-pres)= ...)) is not checked
+ when SIP Dialog Info Notifies are generated during extension
+ monitoring. Added a check to make sure the name and/or number
+ presentations on the callee (remote identity) are set to allow.
+ If they are restricted then "anonymous" is used instead. (closes
+ issue AST-1175) Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2976/
+
+ * channels/chan_sip.c: chan_sip: notify dialog info ignores
+ presentation indicator in callerid The presentation indicator in
+ a callerid (e.g. set by dialplan function
+ Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
+ Info Notifies are generated during extension monitoring. Added a
+ check to make sure the name and/or number presentations on the
+ callee (remote identity) are set to allow. If they are restricted
+ then "anonymous" is used instead. (closes issue AST-1175)
+ Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2976/
+
+2013-11-02 02:11 +0000 [r402407-402425] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/confbridge/conf_state_inactive.c,
+ apps/confbridge/conf_state_single_marked.c,
+ apps/confbridge/include/confbridge.h,
+ apps/confbridge/conf_state_multi.c, apps/app_confbridge.c,
+ apps/confbridge/conf_state_multi_marked.c,
+ apps/confbridge/conf_state.c,
+ apps/confbridge/conf_state_single.c: confbridge: Separate user
+ muting from system muting overrides. The system overrides the
+ user muting requests when MOH is playing or a waitmarked user is
+ waiting for a marked user to join. System muting overrides
+ interfere with what the user may wish the muting to be when the
+ system override ends. * User muting requests are now independent
+ of the system muting overrides. The effective muting is now the
+ logical or of the user request and system override. * Added a
+ Muted column to the CLI "confbridge list <conference>" command. *
+ Added a Muted header to the AMI ConfbridgeList action
+ ConfbridgeList event. (closes issue AST-1102) Reported by: John
+ Bigelow Review: https://reviewboard.asterisk.org/r/2960/
+
+ * main/config.c, configs/confbridge.conf.sample: config: Allow
+ ConfBridge DTMF menus to have '#' as the first digit. ConfBridge
+ allows custom DTMF menus to be created in the confbridge.conf
+ file by assigning a DTMF key sequence to a sequence of actions as
+ follows: DTMF-sequence = action,action... Unfortunately, the
+ normal config file processing code interprets an initial '#'
+ character as starting a directive such as #include. * Add the
+ ability to escape the first non-blank character in a config line
+ so the '#' character can be used without triggering the directive
+ processing code. (closes issue AFS-2) (closes issue
+ ASTERISK-22478) Reported by: Nicolas Tanski Patches:
+ jira_asterisk_22478_v11.patch (license #5621) patch uploaded by
+ rmudgett (modified) Review:
+ https://reviewboard.asterisk.org/r/2969/
+
+2013-11-01 12:31 +0000 [r402345] Kinsey Moore <kmoore at digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ channels/chan_sip.c: chan_sip: Fix RTCP port for SRFLX ICE
+ candidates This corrects one-way audio between Asterisk and
+ Chrome/jssip as a result of Asterisk inserting the incorrect RTCP
+ port into RTCP SRFLX ICE candidates. This also exposes an ICE
+ component enumeration to extract further details from candidates.
+ (closes issue ASTERISK-21383) Reported by: Shaun Clark Review:
+ https://reviewboard.asterisk.org/r/2967/
+
+2013-10-31 15:59 +0000 [r402288] Matthew Jordan <mjordan at digium.com>
+
+ * main/loader.c, /: core/loader: Don't call dlclose in a while loop
+ For awhile now, we've noticed continuous integration builds
+ hanging on CentOS 6 64-bit build agents. After resolving a number
+ of problems with symbols, strange locks, and other shenanigans,
+ the problem has persisted. In all cases, gdb shows the Asterisk
+ process stuck in loader.c on one of the infinite while loops that
+ calls dlclose repeatedly until success. The documentation of
+ dlclose states that it returns 0 on success; any other value on
+ error. It does not state that repeatedly calling it will
+ eventually clear those errors. Most likely, the repeated calls to
+ dlclose was to force a close by exhausting the references on the
+ library; however, that will never succeed if: (a) There is some
+ fundamental error at work in the loaded library that precludes
+ unloading it (b) Some other loaded module is referencing a symbol
+ in the currently loaded module This results in Asterisk sitting
+ forever. Since we have matching pairs of dlopen/dlclose, this
+ patch opts to only call dlclose once, and log out as an ERROR if
+ dlclose fails to return success. If nothing else, this might help
+ to determine why on the CentOS 6 64-bit build agent things are
+ not closing successfully. Review:
+ https://reviewboard.asterisk.org/r/2970 ........ Merged revisions
+ 402287 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-29 23:42 +0000 [r402225] Rusty Newton <rnewton at digium.com>
+
+ * sounds/Makefile, /: Updates for 1.4.25 core sounds and 1.4.14
+ extra sounds, plus new en_GB language set The new sound packages
+ relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413,
+ ASTERISK-20782 Modified sounds/Makefile for the new sound
+ versions and to account for the new en_GB language set. (issue
+ ASTERISK-22659) (closes issue ASTERISK-22659) (closes issue
+ ASTERISK-22411) (closes issue ASTERISK-22544) ........ Merged
+ revisions 402224 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-29 12:49 +0000 [r402151] Matthew Jordan <mjordan at digium.com>
+
+ * main/xmldoc.c, main/channel.c, main/pbx.c, /, main/translate.c:
+ Remove some spammy debug messages; improve clarity of others
+ Debug messages aren't free. Even when the debug level is
+ sufficiently low such that the messages are never evaluated,
+ there is a cost to having to parse Asterisk logs that contain
+ debug messages that (a) fail to convey sufficient information or
+ (b) occur so frequently as to be next to meaningless. Based on
+ having to stare at lots of DEBUG messages, this patch makes the
+ following changes: * channel.c: When copying variables from a
+ parent channel to a child channel, specify the channels involved.
+ Do not log anything for a variable that is not inherited; the
+ fact that it doesn't have an _ or __ already signifies that it
+ won't be inherited. * pbx.c: Specify what function evaluation has
+ occurred that created the result. * translate.c: Bump up the
+ translator path messages to 10. I've never once had to use these
+ debug messages, and for each format that is registered (on
+ startup) and unregistered (on shutdown) the entire f^2 matrix is
+ logged out. For short tests in the Asterisk Test Suite, this
+ should make finding the actual test much easier. * xmldoc.c: The
+ debug message that 'blah' is not found in the tree is expected.
+ Often, description elements - which are not required - are not
+ provided. This debug message adds no additional value, as it is
+ not indicative of an error or helpful in debugging which element
+ did not contain a 'blah' element as a child. If an element is
+ supposed to contain a child element, then that XML tree should
+ have failed validation in the first place. Review:
+ https://reviewboard.asterisk.org/r/2966/ ........ Merged
+ revisions 402150 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-28 14:50 +0000 [r402111] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_sip.c, UPGRADE.txt: chan_sip: Clarify 'Forcerport'
+ Setting Displayed When Running "sip show peers" While looking at
+ ASTERISK-22236, Walter Doekes pointed out that when running "sip
+ show peers", the setting being displayed can be confusing. The
+ display of "N" used to mean NAT (i.e. yes). The NAT setting has
+ gone through many different changes resulting in the display of
+ different characters to try and convey what the current setting
+ is for 'Forcerport' (A for Auto and Forcerport is currently on, a
+ for Auto but Forcerport is off, Y for yes, and N for no). During
+ the initial code review to try and clarify these settings
+ (especially since "N" no longer meant what it used to mean in
+ prior versions of Asterisk), Mark Michelson suggested using the
+ full space available to display the settings which helped to make
+ the settings very clear. That was a great suggestion. Therefore,
+ this patch does the following: * The column for 'Forcerport' now
+ will show: Auto (Yes), Auto (No), Yes, or No. * A column for the
+ 'Comedia' setting has been added. It too will display the setting
+ in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No. *
+ UPGRADE.txt has been updated to document this change. (closes
+ issue ASTERISK-22728) Reported by: Walter Doekes Tested by:
+ Michael L. Young Patches:
+ asterisk-forcerport-display-clarification_v3.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2941
+
+2013-12-17 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.7.0 Released.
+
+2013-12-16 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.7.0-rc2 Released.
+
+ * AST-2013-006 - app_sms: BufferOverflow when receiving odd length 16
+ bit message
+
+ This patch prevents an infinite loop overwriting memory when a
+ message is received into the unpacksms16() function, where the length
+ of the message is an odd number of bytes.
+ (closes issue ASTERISK-22590)
+
+ * AST-2013-007 - security: Inhibit execution of privilege escalating
+ functions
+
+ This patch allows individual dialplan functions to be marked as
+ 'dangerous', to inhibit their execution from external sources.
+
+ A 'dangerous' function is one which results in a privilege
+ escalation. For example, if one were to read the channel variable
+ SHELL(rm -rf /) Bad Things(TM) could happen; even if the external
+ source has only read permissions.
+
+ Execution from external sources may be enabled by setting
+ 'live_dangerously' to 'yes' in the [options] section of
+ asterisk.conf. Although doing so is not recommended.
+
[... 27628 lines stripped ...]
More information about the svn-commits
mailing list