Commit Graph

2553 Commits

Author SHA1 Message Date
Automerge script b375abae50 Merged revisions 378374,378377,378384 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r378374 | rmudgett | 2013-01-02 15:23:16 -0600 (Wed, 02 Jan 2013) | 33 lines
  
  Fix AMI redirect action with two channels failing to redirect both channels.
  
  The AMI redirect action can fail to redirect two channels that are bridged
  together.  There is a race between the AMI thread redirecting the two
  channels and the bridge thread noticing that a channel is hungup from the
  redirects.
  
  * Made the bridge wait for both channels to be redirected before exiting.
  
  * Made the AMI redirect check that all required headers are present before
  proceeding with the redirection.
  
  * Made the AMI redirect require that any supplied ExtraChannel exist
  before proceeding.  Previously the code fell back to a single channel
  redirect operation.
  
  (closes issue ASTERISK-18975)
  Reported by: Ben Klang
  
  (closes issue ASTERISK-19948)
  Reported by: Brent Dalgleish
  Patches:
        jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
  Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode
  
  Review: https://reviewboard.asterisk.org/r/2243/
  ........
  
  Merged revisions 378356 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378358 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378377 | mjordan | 2013-01-02 16:10:32 -0600 (Wed, 02 Jan 2013) | 24 lines
  
  Prevent crashes from occurring when reading from data sources with large values
  
  When reading configuration data from an Asterisk .conf file or when pulling
  data from an Asterisk RealTime backend, Asterisk was copying the data on the
  stack for manipulation. Unfortunately, it is possible to read configuration
  data or realtime data from some data source that provides a large blob of
  characters. This could potentially cause a crash via a stack overflow.
  
  This patch prevents large sets of data from being read from an ARA backend or
  from an Asterisk conf file.
  
  (issue ASTERISK-20658)
  Reported by: wdoekes
  Tested by: wdoekes, mmichelson
  patches:
   * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
   * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
  ........
  
  Merged revisions 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378376 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378384 | mjordan | 2013-01-02 16:19:32 -0600 (Wed, 02 Jan 2013) | 11 lines
  
  Clean up app_mysql's application entry points to properly parse arguments
  
  When parsing arguments, application entry points should not attempt to
  directly modify the parameters to the function. This patch properly duplicates
  the passed in parameters before attempting to parse them.
  
  (issue ASTERISK-20658)
  Reported by: wdoekes
  patches:
    issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674)
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02 22:19:49 +00:00
Automerge script 675914bb17 Merged revisions 378322 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r378322 | mjordan | 2013-01-02 12:11:59 -0600 (Wed, 02 Jan 2013) | 33 lines
  
  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
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  Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 378321 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02 18:20:03 +00:00
Automerge script 5af578c022 Merged revisions 378000-378002 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r378000 | seanbright | 2012-12-13 15:20:32 -0600 (Thu, 13 Dec 2012) | 8 lines
  
  Make generate_exchange_uuid() always return the passed ast_str pointer.
  
  I changed this code earlier to return NULL if it wasn't able to generate a UUID,
  whereas the earlier code would always return the ast_str that was passed in.
  Switch back to returning the ast_str, only set it to the empty string instead if
  UUID generation fails.  We still do a validity check later which will catch this
  and blow up if necessary.
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  r378001 | wedhorn | 2012-12-13 15:25:31 -0600 (Thu, 13 Dec 2012) | 9 lines
  
  Minor fixes for chan_skinny
  
  Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and 
  correct len of 2 strcmp in skinny_setdebug(). (see opticron's review
  on https://reviewboard.asterisk.org/r/2240/)
  ........
  
  Merged revisions 377991 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378002 | rmudgett | 2012-12-13 15:28:15 -0600 (Thu, 13 Dec 2012) | 35 lines
  
  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/
  ........
  
  Merged revisions 377992 from http://svn.asterisk.org/svn/asterisk/branches/10
  ........
  
  Merged revisions 377993 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-13 22:17:36 +00:00
Automerge script 3f3ab25f91 Merged revisions 377981 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r377981 | dlee | 2012-12-13 10:43:40 -0600 (Thu, 13 Dec 2012) | 1 line
  
  Bail configure if it can't find libuuid.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-13 17:17:34 +00:00
Mark Michelson ece4c95798 Resolve conflict and reset automerge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-13 16:39:40 +00:00
Mark Michelson f5e9cf5975 Add automerge property back after conflict.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-11 21:52:09 +00:00
Mark Michelson 7995bc63c7 Some general cleanup, plus we now send state changes when threads activate.
This is now ready for review board, imo!



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-11 16:53:16 +00:00
Mark Michelson 8760e32ae3 Add auto-increment option and accompanying test.
This allows for the threadpool to automatically grow if tasks
are pushed to it and no idle threads are currently available.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-11 16:34:00 +00:00
Mark Michelson 29fc122783 Some documentation fixes and function call name fixes.
The documentation for taskprocessors was incorrect with
regards to when a listener's alloc callback was called.

I also made the names of queued function calls in the
threadpool more uniform.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-11 04:23:31 +00:00
Mark Michelson 1310e659bd Solve the issue about the "CHANGE THIS" and "XXX CHANGE THIS XXX" taskprocessor names.
Unfortunately, this required a taskprocessor listener change that makes listener allocation
utterly silly. I'm going to change the scheme so that allocation of taskprocessor listeners
is done internally within taskprocessor code. This will make it parallel with threadpool
code, which is a good thing.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-10 20:14:23 +00:00
Mark Michelson 64deed062a Add threadpool options and accompanying test.
The only test added so far is an idle thread timeout
option. This will greatly aid threadpool users who wish
to maintain a threadpool by allowing for idle threads to
die out as necessary.

Test passes.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-10 06:13:09 +00:00
Mark Michelson 4590bfd93d Add new threadpool test and fix some taskprocessor bugs.
The new thread creation test fails because Asterisk locks up
while trying to lock a taskprocessor.

While trying to debug that, I found a race condition during taskprocessor
creation where a default taskprocessor listener could try to operate on
a partially started taskprocessor. This was fixed by adding a new callback
to taskprocessor listeners.

Then while testing that change, I found some bugs in the taskprocessor
tests where I was not properly unlocking when done with a lock. Scoped
locks have spoiled me a bit.

I still have not figured out why the threadpool thread creation test
is locking up.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-07 00:30:35 +00:00
Automerge script 521f9e8dfe Merged revisions 377245-377246 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r377245 | rmudgett | 2012-12-04 20:20:57 -0600 (Tue, 04 Dec 2012) | 8 lines
  
  Fix registering core show codecs/codec CLI commands twice.
  ........
  
  Merged revisions 377241 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377244 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r377246 | rmudgett | 2012-12-04 20:23:10 -0600 (Tue, 04 Dec 2012) | 1 line
  
  Remove init_framer(). It no longer does anything.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-05 03:19:08 +00:00
Mark Michelson cc63d2c380 Add better listener support.
Add some parameters to listener callbacks.
Add alloc and destroy callbacks for listeners.
Add public function for allocating a listener.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-04 23:45:39 +00:00
Mark Michelson 2158005bdb Remove zombie state from threadpool altogether.
After giving it some consideration, there's no real
use for zombie threads. Listeners can't really use the
current number of zombie threads as a way of gauging activity,
zombifying threads is just an extra step before they die that
really serves no purpose, and since there's no way to re-animate
zombies, the operation does not need to be around.

I also fixed up some miscellaneous compilation errors that
were lingering from some past revisions.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-04 22:11:31 +00:00
Mark Michelson a37fb2e8c8 Add some doxygen and rearrange code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-04 21:11:34 +00:00
Automerge script d53adbe449 Merged revisions 377138 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r377138 | rmudgett | 2012-12-03 14:46:11 -0600 (Mon, 03 Dec 2012) | 23 lines
  
  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
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  Merged revisions 377136 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377137 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-03 21:19:40 +00:00
Mark Michelson e7ce12839d This now compiles.
That's a milestone, of sorts. Things really need
arranging/documenting, and there's no function to
be able to push tasks to a threadpool.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-03 16:59:26 +00:00
Mark Michelson ddde765c59 Commit some progress towards threadpools.
Does this compile? Not even close.
But I figure I don't want to lose this all in the case
of some catastrophe.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-29 18:54:51 +00:00
Automerge script 8c84eb128f Merged revisions 376630 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r376630 | rmudgett | 2012-11-27 11:54:25 -0600 (Tue, 27 Nov 2012) | 13 lines
  
  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
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2012-11-27 18:20:18 +00:00
Automerge script 37ae4ad43f Merged revisions 376589 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r376589 | mjordan | 2012-11-22 18:02:23 -0600 (Thu, 22 Nov 2012) | 29 lines
  
  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
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2012-11-23 00:20:55 +00:00
Automerge script d16d0200d2 Merged revisions 376575 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r376575 | rmudgett | 2012-11-21 12:33:16 -0600 (Wed, 21 Nov 2012) | 20 lines
  
  Add red-black tree container type to astobj2.
  
  * Add red-black tree container type.
  
  * Add CLI command "astobj2 container dump <name>"
  
  * Added ao2_container_dump() so the container could be dumped by other
  modules for debugging purposes.
  
  * Changed ao2_container_stats() so it can be used by other modules like
  ao2_container_check() for debugging purposes.
  
  * Updated the unit tests to check red-black tree containers.
  
  (closes issue ASTERISK-19970)
  Reported by: rmudgett
  Tested by: rmudgett
  
  Review: https://reviewboard.asterisk.org/r/2110/
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2012-11-21 19:20:22 +00:00
Mark Michelson e2196d7981 Get rid of trailing whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-19 22:34:27 +00:00
Mark Michelson f4328e109d Reorganize code and change behavior of ast_taskprocessor_execute() when taskprocessor is shutting down.
Moved code around to be easier to follow.

ast_taskprocessor_execute() will now return 0 if the taskprocessor is being shut down.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-19 21:31:32 +00:00
Mark Michelson 2b36cbe2d5 Change the write-up on taskprocessors to reflect the new design.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 04:44:12 +00:00
Mark Michelson 12de4198b8 Add a shutdown callback to taskprocessor listeners.
This helps account for the fact that it is unknown just
how many references may exist for a given taskprocessor
listener, so simply unreffing it from the taskprocessor
shutdown function is not enough to convey the gravity
of the situation.

By putting in a shutdown callback, it now becomes clear
to the listener not to try to do any further operations
on the taskprocessor.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-16 04:33:53 +00:00
Automerge script e8898ec8ba Merged revisions 376341,376344-376345 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r376341 | dlee | 2012-11-15 18:08:00 -0600 (Thu, 15 Nov 2012) | 34 lines
  
  Migrate hashtest/hashtest2 to be unit tests.
  
  Both hashtest and hashtest2 are manual testing apps that thrash hash
  tables (hashtab and ao2 containers, respectively), by spinning up
  several threads that randomly insert, delete, lookup and iterate over
  the hash table. If the app doesn't crash, the hash table probably passes
  the test. Those utils are not a part of the typical Asterisk build, so
  they do not usually get compiled. This all makes them less that useful.
  
  This patch removes those manual test programs and replaces them with
  Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also
  attempts to make the tests more deterministic.
  
  * Rather than spinning up some number of threads that operate on the
    hash table randomly, spin up four threads that concurrenly add,
    remove, lookup and iterate over the hash table.
  * Each thread checks the state of the hash table both during and after
    execution, and indicates a test failure if things are not as expected.
  * Each thread times out after 60 seconds to prevent deadlocking the unit
    test run.
  
  (closes issue ASTERISK-20505)
  Reported by: Matt Jordan
  Review: https://reviewboard.asterisk.org/r/2189/
  ........
  
  Merged revisions 376306 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r376344 | dlee | 2012-11-15 18:14:00 -0600 (Thu, 15 Nov 2012) | 1 line
  
  Somehow I put in svn-1.6 merge information. Oops.
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  r376345 | dlee | 2012-11-15 18:15:30 -0600 (Thu, 15 Nov 2012) | 15 lines
  
  Fixed extconf.c breakage introduced in r376306.
  
  To quote wdoekes:
  > Note that I'm not confirming legitimacy of having that file in tree at
  > all. Is anyone using aelparse/conf2ael?
  ........
  
  Merged revisions 376340 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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2012-11-16 00:19:48 +00:00
Mark Michelson a4a48d9274 Add doxygen and constify some things.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-09 22:49:25 +00:00
Mark Michelson d5716ecae2 Genericize the allocation and destruction of taskprocessor listeners.
The goal of this is to take the responsibility away from individual
listeners to be sure to properly unref the taskprocessor.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-09 22:28:10 +00:00
Mark Michelson 77725bf293 Move taskprocessors to use a listener model.
Taskprocessors are now divided into two units: the task queue
and their listeners.

When a task is added to the queue, the listener is notified and
can take whatever action is desired. This means that taskprocessors
are no longer confined to having their tasks executed within a 
single thread.

A default taskprocessor listener has been added that mirrors the
old taskprocessor behavior.

I've tested it by running Asterisk and placing calls. It appears
to work as expected. I'm going to do some cleaning up first and
then write some unit tests to be sure everything works as expected.



git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-08 23:27:16 +00:00
Automerge script f69513b85b Merged revisions 376049 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r376049 | rmudgett | 2012-11-08 11:38:31 -0600 (Thu, 08 Nov 2012) | 41 lines
  
  Add MALLOC_DEBUG enhancements.
  
  * Makes malloc() behave like calloc().  It will return a memory block
  filled with 0x55.  A nonzero value.
  
  * Makes free() fill the released memory block and boundary fence's with
  0xdeaddead.  Any pointer use after free is going to have a pointer
  pointing to 0xdeaddead.  The 0xdeaddead pointer is usually an invalid
  memory address so a crash is expected.
  
  * Puts the freed memory block into a circular array so it is not reused
  immediately.
  
  * When the circular array rotates out a memory block to the heap it checks
  that the memory has not been altered from 0xdeaddead.
  
  * Made the astmm_log message wording better.
  
  * Made crash if the DO_CRASH menuselect option is enabled and something is
  found.
  
  * Fixed a potential alignment issue on 64 bit systems.
  struct ast_region.data[] should now be aligned correctly for all
  platforms.
  
  * Extracted region_check_fences() from __ast_free_region() and
  handle_memory_show().
  
  * Updated handle_memory_show() CLI usage help.
  
  Review: https://reviewboard.asterisk.org/r/2182/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-08 18:19:49 +00:00
Mark Michelson f2bb9afe17 Multiple revisions 375993-375994
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  r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
  
  Fix misuses of timeouts throughout the code.
  
  Prior to this change, a common method for determining if a timeout
  was reached was to call a function such as ast_waitfor_n() and inspect
  the out parameter that told how many milliseconds were left, then use
  that as the input to ast_waitfor_n() on the next go-around.
  
  The problem with this is that in some cases, submillisecond timeouts
  can occur, resulting in the out parameter not decreasing any. When this
  happens thousands of times, the result is that the timeout takes much
  longer than intended to be reached. As an example, I had a situation where
  a 3 second timeout took multiple days to finally end since most wakeups
  from ast_waitfor_n() were under a millisecond.
  
  This patch seeks to fix this pattern throughout the code. Now we log the
  time when an operation began and find the difference in wall clock time
  between now and when the event started. This means that sub-millisecond timeouts
  now cannot play havoc when trying to determine if something has timed out.
  
  Part of this fix also includes changing the function ast_waitfor() so that it
  is possible for it to return less than zero when a negative timeout is given
  to it. This makes it actually possible to detect errors in ast_waitfor() when
  there is no timeout.
  
  (closes issue ASTERISK-20414)
  reported by David M. Lee
  
  Review: https://reviewboard.asterisk.org/r/2135/
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  r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
  
  Remove some debugging that accidentally made it in the last commit.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-07 19:15:26 +00:00
Richard Mudgett 6ad0126425 Fix stuck DTMF when bridge is broken.
When a bridge is broken by an AMI Redirect action or the ChannelRedirect
application, an in progress DTMF digit could be stuck sending forever.

* Made simulate a DTMF end event when a bridge is broken and a DTMF digit
was in progress.

(closes issue ASTERISK-20492)
Reported by: Jeremiah Gowdy
Patches:
      bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by Jeremiah Gowdy
      Modified to jira_asterisk_20492_v1.8.patch
      jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/2169/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-06 19:05:11 +00:00
Matthew Jordan a0c363e227 Refactor ast_timer_ack to return an error and handle the error in timer users
Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor.  This can lead to situations where errors stream to the
CLI/log file.  This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.

This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures.  It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.

Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.

Review: https://reviewboard.asterisk.org/r/2178/

(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
  jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-05 23:10:14 +00:00
Richard Mudgett b0c3d288f2 build_tools: Allow Asterisk to report git SHAs in version string.
Make git more attractive for managing work-in-progress.  Especially
convenient when a potential patch set needs to be tested on multiple
platforms since one can use git to keep all the test environments in sync
independent of a subversion server.

Now the Asterisk version will show the exact git SHA5 that was used when
building (still appended by "M" if there are local modifications) from a
git clone of the Asterisk repository so the developer can more easily know
what is actually under test.

You will now get this:

  $ asterisk -V
  Asterisk GIT-1698298

Instead of this:

  $ asterisk -V
  Asterisk UNKNOWN__and_probably_unsupported

This has zero impact for those not using git with the exception of an
extra test in the configure script to gather git's path.  This is
necessary to prevent "sudo make install" from failing since git may not be
in the path in make's shell environment.

(closes issue ASTERISK-20483)
Reported by: Shaun Ruffell
Patches:
      0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch (license #5417) patch uploaded by Shaun Ruffell
      Modified
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-18 20:13:17 +00:00
Andrew Latham 6c20cf2d8a Doxygen Updates - Title update
Update and extend the configuration_file group and enable linking. Commit other cleanups from multi-version Doxygen testing.  Update title that was left behind many years ago.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-18 14:17:40 +00:00
Mark Michelson e9ab568f88 Fix some potential misuses of ast_str in the code.
Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.

This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.

I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.

Review: https://reviewboard.asterisk.org/r/2161
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-15 21:25:29 +00:00
Mark Michelson c7b23cbb0a Do not use a FILE handle when doing SIP TCP reads.
This is used to solve an issue where a poll on a file
descriptor does not necessarily correspond to the readiness
of a FILE handle to be read.

This change makes it so that for TCP connections, we do a
recv() on the file descriptor instead.

Because TCP does not guarantee that an entire message or even
just one single message will arrive during a read, a loop has
been introduced to ensure that we only attempt to handle a
single message at a time. The tcptls_session_instance structure
has also had an overflow buffer added to it so that if more
than one TCP message arrives in one go, there is a place to
throw the excess.

Huge thanks goes out to Walter Doekes for doing extensive review
on this change and finding edge cases where code could fail.

(closes issue ASTERISK-20212)
reported by Phil Ciccone

Review: https://reviewboard.asterisk.org/r/2123
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-12 16:31:01 +00:00
Andrew Latham 7226606f77 Continue to group config files
(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 22:39:02 +00:00
Mark Michelson 825607e09b Don't make chan_sip export global symbols.
During testing, it was discovered that having chan_sip
export global symbols was problematic.

The biggest problem was that load order was affected.
Trying to use realtime could be problematic since in
all likelihood the necessary realtime driver(s) would
not be loaded before chan_sip.

In addition, it was found that it was impossible to
use the Digium Phone Module for Asterisk since it
must be loaded before chan_sip since it must hook
into chan_sip's configuration parsing.

The solution is to use a virtual table in the same
manner that other modules in Asterisk do, like
app_voicemail.

(closes issue ASTERISK-20545)
Reported by: kmoore

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 15:49:02 +00:00
Joshua Colp d78f7f92b2 Add support for applying direct media ACLs between differing channel technologies.
Review: https://reviewboard.asterisk.org/r/2122/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-04 13:49:45 +00:00
Matthew Jordan a094707d51 Fix a variety of ref counting issues
This patch resolves a number of ref leaks that occur primarily on Asterisk
shutdown.  It adds a variety of shutdown routines to core portions of
Asterisk such that they can reclaim resources allocate duringd initialization.

Review: https://reviewboard.asterisk.org/r/2137
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2012-10-02 01:47:16 +00:00
Andrew Latham 4e228fce03 Doxygen Cleanup
Start adding configuration file linking and pages.  Add module loading doxygen block.

Breaking up commits to keep it easy to track

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-01 23:39:45 +00:00
Sean Bright b9eeff1521 app_queue: Support persisting and loading of long member lists.
Greenlight in #asterisk brought up that he was receiving an error message "Could
not create persistent member string, out of space" when running app_queue in
Asterisk 10.  dump_queue_members() made an assumption that 8K would be enough to
store the generated string, but with queues that have large member lists this is
not always the case.  This patch removes the limitation and uses ast_str instead
of a fixed sized buffer.

The complicating factor comes from the fact that ast_db_get requires a buffer
and buffer size argument, which doesn't let us pull back more than what we pass
in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d
copy of the value from astdb.

As an aside, I did some testing on the maximum size of data that we can store in
the BDB library we distribute and was able to store a 10MB string and retrieve
it with no problems, so I feel this is a safe patch.

Review: https://reviewboard.asterisk.org/r/2136/
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2012-10-01 20:36:25 +00:00
Joshua Colp 0fc114dc65 Add support for retrieving engine specific settings using the speech API and from dialplan.
(closes issue ASTERISK-17136)
Reported by: kenner


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-01 12:29:04 +00:00
Joshua Colp 9f55e5e928 Make res_http_websocket an optional dependency on supported platforms for chan_sip.
(closes issue ASTERISK-20439)
Reported by: sruffell
Patches:
     0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded by sruffell (license 5417)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-27 17:12:08 +00:00
Mark Michelson fdfb3ae5fa Allow for redirecting reasons to be set to arbitrary strings.
This allows for the REDIRECTING dialplan function to be
used to set the reason to any string.

The SIP channel driver has been modified to set the redirecting
reason string to the value received in a Diversion header. In
addition, SIP 480 response reason text will set the redirecting
reason as well.

(closes issue AST-942)
reported by Malcolm Davenport

(closes issue AST-943)
reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/2101



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-25 19:29:14 +00:00
Andrew Latham fd98835f1f Doxygen Updates Janitor Work
* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-22 20:43:30 +00:00
Andrew Latham 6f61cb50c5 Doxygen Updates - janitor work
Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style.  Some missing txt file links are removed but their content or essense will be included in some later updates.  A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen.

Further updates coming.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-21 17:14:59 +00:00
Joshua Colp e8380afc8a Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
As mentioned on the review for this, WebRTC has moved towards choosing
DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds
support for this but makes it available for normal SIP clients as well.

Testing has been done to ensure that this introduces no regressions with
existing behavior and also that it functions as expected.

Review: https://reviewboard.asterisk.org/r/2113/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-20 18:27:28 +00:00