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${ noResults }
23780 Commits (a100307eaa3e9348d9743dc63cfbbf4962fccfc1)
Author | SHA1 | Message | Date |
---|---|---|---|
|
4314a33ae2 |
chan_mgcp: Enforce locking for oseq
This restricts direct usage of global oseq so that all accesses are locked and threads are not racing to get oseq values that they did not claim. This also fixes a build error in res_pktccops under dev mode. (closes issue ASTERISK-23100) Reported by: adomjan Patch by: adomjan ........ Merged revisions 406037 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@406038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
916d97c5a0 |
chan_dahdi/PRI: Suppress CONNECTED_LINE updates when nothing in the udpate is valid.
* Also simplified some subddress handling code. (closes issue ASTERISK-23008) Reported by: Michael Cargile ........ Merged revisions 405926 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405927 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
9e6407596b |
Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405792 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
6a7984281d |
manager: Originate doesn't abort on failed format_cap allocation
action_originate responds to the remote system with an error when cap==NULL, but doesn't return (abort the originate). Patched to return. (closes issue ASTERISK-23034) Reported by: Corey Farrell Patches: ASTERISK-23034.patch uploaded by coreyfarrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405745 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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b83fb704c9 |
res_fax: check_modem_rate() returned incorrect rate for V.27
According to the new standard for V.27 and V.32 they are able to transmit at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be updated to reflect this. Also, because of this change the default 'minrate' value was updated to be 4800. (closes issue ASTERISK-22790) Reported by: Paolo Compagnini Patches: res_fax.txt uploaded by looserouting (license 6548) ........ Merged revisions 405656 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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3d1ce59cf2 |
cel_manager: Don't crash if configuration file is invalid.
The cel_manager module did not properly handle the case where the configuration file was invalid. The module will now output a warning message and disable itself if this occurs. Reported by: Bryan Walters ........ Merged revisions 405581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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7f12e96389 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405487 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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f8a0a58f0d |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405434 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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f90a045a36 |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405431 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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b12f6bc358 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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6d73ecd6e9 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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9556fb2516 |
Logging callid: Fix some sizeof() references per coding guidelines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405281 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
dee9ce4a0b |
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 Valentinavičius Patches: works_on_my_machine.patch uploaded by xytis (license 6558) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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f9465fd40d |
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/ Note that the patch committed here is essentially the same as uploaded by Simon Moxon on ASTERISK-22740, with the addition of the double kick prevention. (closes issue AST-1258) Reported by: Steve Pitts (closes issue ASTERISK-22740) Reported by: Simon Moxon patches: ASTERISK-22740.diff uploaded by Simon Moxon (license 6546) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405215 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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06e1bdd480 |
"Minimun" typo.
........ Merged revisions 405160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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10cf34d36a |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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05f0ad2c56 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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9433c584d8 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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fec771b7c8 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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4b7d8e4df4 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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dd413377dc |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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a8393e64bc |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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0abb5b9e18 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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27eb7cc704 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404604 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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864d62349b |
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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f7a03cc968 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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1f1ac57bbe |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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e87fa7d731 |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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c9b2d2ee68 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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38c1c1de83 |
Add AMI event for presence state.
Review: https://reviewboard.asterisk.org/r/3039/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404275 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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2b39b36de8 |
ooh323c: Fix gcc 4.6.3 compiler warnings.
........ Merged revisions 404212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404219 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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d9b7ce0599 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404136 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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acaa978d71 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404087 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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05117c87b7 |
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404045 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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a9ee948e90 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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faf4f6187b |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403863 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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8d5186ef53 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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788dccff42 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403635 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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e7d81f02e5 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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6861233e52 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403288 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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bef4ec92ad |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403015 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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c5c9050b89 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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6d85bb82e1 |
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! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402686 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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3b3c9f38f4 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402646 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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fe7c661e12 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402605 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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5b51b31321 |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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1664326ba9 |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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864163492c |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402425 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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1f051c7e2d |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402407 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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2d5debf45c |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402345 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |