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  r378288 | mjordan | 2013-01-02 09:39:42 -0600 (Wed, 02 Jan 2013) | 36 lines
  
  Resolve crashes due to large stack allocations when using TCP
  
  Asterisk had several places where messages received over various network
  transports may be copied in a single stack allocation. In the case of TCP,
  since multiple packets in a stream may be concatenated together, this can
  lead to large allocations that overflow the stack.
  
  This patch modifies those portions of Asterisk using TCP to either
  favor heap allocations or use an upper bound to ensure that the stack will not
  overflow:
   * For SIP, the allocation now has an upper limit
   * For HTTP, the allocation is now a heap allocation instead of a stack
     allocation
   * For XMPP (in res_jabber), the allocation has been eliminated since it was
     unnecesary.
  
  Note that the HTTP portion of this issue was independently found by Brandon
  Edwards of Exodus Intelligence.
  
  (issue ASTERISK-20658)
  Reported by: wdoekes, Brandon Edwards
  Tested by: mmichelson, wdoekes
  patches:
    ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
    issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
    issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
  ........
  
  Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378286 from http://svn.asterisk.org/svn/asterisk/branches/10
  ........
  
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  r378259 | lathama | 2013-01-01 13:02:52 -0600 (Tue, 01 Jan 2013) | 5 lines
  
  Add UUID packages now required to configure
  
  In ASTERISK-20726 UUID was added to Asterisk.  This commit is to add the dependancies to the install script
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  r378248 | seanbright | 2013-01-01 11:03:59 -0600 (Tue, 01 Jan 2013) | 2 lines
  
  Bail out early when building an ast_trans_pvt and the translator doesn't supply a 'newpvt'
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  r378249 | seanbright | 2013-01-01 11:10:42 -0600 (Tue, 01 Jan 2013) | 2 lines
  
  Revert 378248.  I changed the logic of this function unitentionally, pointed out by file.
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  r378220 | kmoore | 2012-12-31 08:46:06 -0600 (Mon, 31 Dec 2012) | 18 lines
  
  Ensure chan_sip rejects encrypted streams without crypto info
  
  This ensures that Asterisk rejects encrypted media streams (RTP/SAVP
  audio and video) that are missing cryptographic keys and ensures that
  the incoming SDP is consistent with RFC4568 as far as having a crypto
  attribute present for any SAVP streams.
  
  Review: https://reviewboard.asterisk.org/r/2204/
  ........
  
  Merged revisions 378217 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378218 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r378166 | rmudgett | 2012-12-20 15:51:03 -0600 (Thu, 20 Dec 2012) | 8 lines
  
  Give the causes[] a struct name.
  ........
  
  Merged revisions 378164 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378165 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378122 | kmoore | 2012-12-18 11:48:36 -0600 (Tue, 18 Dec 2012) | 17 lines
  
  Add test events for time limit-related hangups
  
  This patch adds hangup-related test events in order to support testing
  of time-limited bridges. This aids in testing the S() and L() bridge
  options.
  
  (issue SWP-4713)
  ........
  
  Merged revisions 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378120 from http://svn.asterisk.org/svn/asterisk/branches/10
  ........
  
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  r378091 | rmudgett | 2012-12-17 17:02:54 -0600 (Mon, 17 Dec 2012) | 22 lines
  
  Make chan_local module references tied to local_pvt lifetime.
  
  The chan_local module references were manually tied to the existence of
  the ;1 and ;2 channel links.
  
  * Made chan_local module references tied to the existence of the local_pvt
  structure as well as automatically take care of the module references.
  
  * Tweaked the wording of the local_fixup() failure warning message to make
  sense.
  
  Review: https://reviewboard.asterisk.org/r/2181/
  ........
  
  Merged revisions 378088 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378089 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 378090 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378095 | rmudgett | 2012-12-17 17:10:42 -0600 (Mon, 17 Dec 2012) | 11 lines
  
  Fix potential double free when unloading a module.
  ........
  
  Merged revisions 378092 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378093 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r378081 | rmudgett | 2012-12-17 15:22:21 -0600 (Mon, 17 Dec 2012) | 7 lines
  
  chan_local: Parse dial string consistently.
  
  * Fix local_alloc() unexpected limitation of exten and context length from
  a combined length of 80 characters to a normal 80 characters each.
  
  * Made local_alloc() and local_devicestate() parse the same way.
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  r378072 | rmudgett | 2012-12-17 14:34:25 -0600 (Mon, 17 Dec 2012) | 9 lines
  
  chan_local: Misc lock and ref tweaks.
  
  * awesome_locking() does not need to thrash the pvt lock as much.
  
  * local_setoption() does not need to check for NULL pvt on cleanup since
  it will never be NULL.
  
  * Made ref the pvt before locking for consistency.
................
  r378074 | qwell | 2012-12-17 14:59:51 -0600 (Mon, 17 Dec 2012) | 10 lines
  
  Make libasteriskssl.so symlink use a relative path.
  
  This was causing issues when using DESTDIR, since the path to which the link
  pointed is not likely to exist (and not useful to exist) on the target system.
  
  (issue ASTNOW-284)
  ........
  
  Merged revisions 378073 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378063 | jrose | 2012-12-14 16:34:18 -0600 (Fri, 14 Dec 2012) | 8 lines
  
  Features: BRIDGE_FEATURES variable automixmonitor support and use proper party
  
  BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it
  does. In addition, the BRIDGE_FEATURES variable would not apply features to
  the proper party based on whether the feature option letter was in caps or
  in lowercase (both ways would apply it to the caller). Now uppercase applies
  to the caller while lowercase applies to the callee (like with the dial option)
........
  r378064 | rmudgett | 2012-12-14 16:45:03 -0600 (Fri, 14 Dec 2012) | 4 lines
  
  chan_agent: Remove some duplicated code.
  
  No need to check for an agent twice.  Santa does that.
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  r378039 | rmudgett | 2012-12-14 15:35:44 -0600 (Fri, 14 Dec 2012) | 26 lines
  
  app_queue: Revert bad ringinuse=no patch.
  
  With the option ringinuse=no set, the patch committed for ASTERISK-16115
  causes non-SIP queue members to never be called because the device state
  is checked after a channel is created to determine if the member is busy.
  These queue members always get the "Member %s is busy, cannot dial"
  message.
  
  Most channel drivers other than chan_sip use the default device state
  handling.  The default device-state state is considered in use or unknown
  if the channel exists or not respectively.
  
  (closes issue ASTERISK-20801)
  Reported by: rmudgett
  Patches:
        jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett
  ........
  
  Merged revisions 378036 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r378029 | rmudgett | 2012-12-14 14:22:36 -0600 (Fri, 14 Dec 2012) | 1 line
  
  app_queue: Make update_status() not return anything.
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  r378011 | wedhorn | 2012-12-13 19:55:43 -0600 (Thu, 13 Dec 2012) | 15 lines
  
  Fix skinny to recognise vmexten in general section of conf
  
  Fixup the vmexten so if globally set in general section will be honored by
  chan_skinny. Also get rid of the 'global_' part of variable name to match
  regexten.
  
  (closes issue AST-20790)
  Reported by: snuffy
  Tested by: snuffy, myself
  Patches: 
      skinny-vm.diff uploaded by snuffy (license 5024)
  ........
  
  Merged revisions 378010 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378006 | wedhorn | 2012-12-13 19:02:15 -0600 (Thu, 13 Dec 2012) | 8 lines
  
  Add g722 codec support to skinny
  
  (closes issue AST-20788)
  Reported by: snuffy
  Tested by: snuffy, myself
  Patches: 
      skinny-g722.diff uploaded by snuffy (license 5024)
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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.
................
  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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  r377994 | dlee | 2012-12-13 15:15:44 -0600 (Thu, 13 Dec 2012) | 1 line
  
  Fixed svn merge property breakage from r377986
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  r377986 | wedhorn | 2012-12-13 12:28:41 -0600 (Thu, 13 Dec 2012) | 14 lines
  
  Fix skinny debug tab completion
  
  Review the syntax of the 'skinny debug' command to show more than
  just 'show' for options to 'skinny debug' command.
  
  (closes issue ASTERISK-20789)
  Reported by: snuffy
  Tested by: snuffy, myself
  Patches:
      skinny-debug.diff uploaded by snuffy (license 5024)
  ........
  
  Merged revisions 377985 from http://svn.asterisk.org/svn/asterisk/branches/11
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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.
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  r377966 | kmoore | 2012-12-13 08:28:57 -0600 (Thu, 13 Dec 2012) | 23 lines
  
  Ensure Min-SE is included in outbound INVITEs
  
  Asterisk now includes Min-SE in outbound INVITEs when the value is not
  90 (the default) and session timers are not disabled. This has the
  effect of Asterisk following RFC4028 more closely with regard to 422
  responses and preventing situations in which Asterisk would be forced
  to temporarily accept a call to tear it down based on a Session-Expires
  below the locally configured Min-SE.
  
  (issue SWP-5051)
  Review: https://reviewboard.asterisk.org/r/2222/
  Reported-by: Kinsey Moore
  Patch-by: Kinsey Moore
  ........
  
  Merged revisions 377946 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 377947 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377948 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r377925 | newtonr | 2012-12-12 16:43:40 -0600 (Wed, 12 Dec 2012) | 18 lines
  
  Incremented EXTRA_SOUNDS_VERSION in sounds/Makefile to 1.4.12 for new Extra Sounds releases
  
  See CHANGES-* files in English extra 1.4.12 tarballs for new sound prompts added.
  
  (closes ASTERISK-20328)
  Reported by: Matt Jordan
  (closes AST-755)
  Reported by: John Bigelow
  ........
  
  Merged revisions 377922 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 377923 from http://svn.asterisk.org/svn/asterisk/branches/10
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  r377915 | elguero | 2012-12-11 22:43:18 -0600 (Tue, 11 Dec 2012) | 17 lines
  
  Convert Dynamic Features Buffer To Use ast_str
  
  Currently, the buffer for the dynamic features list is set to a fixed size of
  128.  If the list is bigger than that, it results in the dynamic feature(s) not
  being recognized.
  
  This patch changes the buffer from a fixed size to a dynamic one.
  
  (closes issue ASTERISK-20680)
  Reported by: Clod Patry
  Tested by: Michael L. Young
  Patches: 
      asterisk-20680-dynamic-features-v2.diff 
                                       uploaded by Michael L. Young (license 5026)
  
  Review: https://reviewboard.asterisk.org/r/2221/
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  r377911 | mmichelson | 2012-12-11 18:02:31 -0600 (Tue, 11 Dec 2012) | 22 lines
  
  Fix a potential deadlock in chan_sip during transfers.
  
  The issue comes from the fact that transfers may perform
  a redirecting update on a channel. The issue is that lock
  inversion between the channel and its tech_pvt occurs since
  the channel lock is released during the transfer process.
  
  The fix is to move when the redirecting update occurs to a
  place where neither the tech_pvt or the channel is locked so
  that the two can be locked in the proper order.
  
  (closes issue ASTERISK-20708)
  reported by Mark Michelson
  patches:
  	ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049)
  
  Tested by:
  	Tim Ringenbach at Asteria Solutions Group
  ........
  
  Merged revisions 377910 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r377906 | mmichelson | 2012-12-11 16:42:11 -0600 (Tue, 11 Dec 2012) | 3 lines
  
  Add test events necessary for bridging tests to be able to properly run.
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  r377809 | rmudgett | 2012-12-11 14:05:32 -0600 (Tue, 11 Dec 2012) | 23 lines
  
  Cleanup pbx on exit.
  
  * Cleanup CLI commands on exit.
  
  * Unreference hints and statecbs containers on exit.
  
  (issue ASTERISK-20649)
  Reported by: Corey Farrell
  Patches:
        pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell
        pbx-cleanup-10.patch (license #5909) patch uploaded by Corey Farrell
        pbx-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
        Modified
  ........
  
  Merged revisions 377806 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 377807 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377808 from http://svn.asterisk.org/svn/asterisk/branches/11
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This is now ready for review board, imo!



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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

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

file:///srv/subversion/repos/asterisk/trunk

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  r377774 | rmudgett | 2012-12-10 20:44:53 -0600 (Mon, 10 Dec 2012) | 19 lines
  
  Cleanup logger on exit.
  
  * Cleanup CLI commands, destroy verbosers and logchannels lists on exit.
  
  (issue ASTERISK-20649)
  Reported by: Corey Farrell
  Patches:
        logger-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
        Modified
  ........
  
  Merged revisions 377771 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 377772 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377773 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@377781 65c4cc65-6c06-0410-ace0-fbb531ad65f3

file:///srv/subversion/repos/asterisk/trunk

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  r377743 | rmudgett | 2012-12-10 20:13:37 -0600 (Mon, 10 Dec 2012) | 25 lines
  
  Cleanup indications on exit.
  
  * Made ast_unregister_indication_country() unlink the found tone zone
  before selecting a new default_tone_zone to make it impossible to select
  the tone zone being unregistered again.
  
  * Ringcadence is no longer parsed twice in store_config_tone_zone().
  
  * Cleanup CLI commands and destroy default_tone_zone on exit.
  
  (issue ASTERISK-20649)
  Reported by: Corey Farrell
  Patches:
        indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
        Modified
  ........
  
  Merged revisions 377740 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 377741 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377742 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@377750 65c4cc65-6c06-0410-ace0-fbb531ad65f3

file:///srv/subversion/repos/asterisk/trunk

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  r377707 | rmudgett | 2012-12-10 18:37:01 -0600 (Mon, 10 Dec 2012) | 20 lines
  
  Cleanup dnsmgr on exit.
  
  * Cleanup dnsmgr thread and CLI commands on exit.
  
  (issue ASTERISK-20649)
  Reported by: Corey Farrell
  Patches:
        dnsmgr-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell
        dnsmgr-cleanup-10-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
        Modified
  ........
  
  Merged revisions 377704 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 377705 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377706 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r377711 | rmudgett | 2012-12-10 19:04:50 -0600 (Mon, 10 Dec 2012) | 19 lines
  
  Cleanup event on exit.
  
  * Cleanup CLI commands on exit.
  
  (issue ASTERISK-20649)
  Reported by: Corey Farrell
  Patches:
        event_shutdown-10-only.patch (license #5909) patch uploaded by Corey Farrell
        event_shutdown-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
  ........
  
  Merged revisions 377708 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 377709 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377710 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@377719 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377686 65c4cc65-6c06-0410-ace0-fbb531ad65f3

file:///srv/subversion/repos/asterisk/trunk

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  r377658 | kmoore | 2012-12-10 10:56:37 -0600 (Mon, 10 Dec 2012) | 20 lines
  
  Ensure ReceiveFax provides a CED tone via T.38
  
  When using res_fax_digium, the T.38 CED tone was not being provided
  properly which would cause some incoming faxes to fail. This was not an
  issue with res_fax_spandsp since it does not strictly honor the
  send_ced flag and sends the CED tone whenever receiving a T.38 fax.
  
  (closes issue FAX-343)
  Reported-by: Benjamin Tietz
  Patch-by: Kinsey Moore
  ........
  
  Merged revisions 377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 377656 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377657 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@377665 65c4cc65-6c06-0410-ace0-fbb531ad65f3