Commit Graph

23271 Commits

Author SHA1 Message Date
Automerge script ad73fe2c9f Merged revisions 378634 via svnmerge from
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  r378634 | wedhorn | 2013-01-06 15:37:59 -0600 (Sun, 06 Jan 2013) | 6 lines
  
  Skinny blob cleanup
  
  Cleanup of red blobs in chan_skinny and possible other small formatting issues.
  
  Review: https://reviewboard.asterisk.org/r/2262/
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Automerge script 00b40f6e2a Merged revisions 378623-378624 via svnmerge from
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  r378623 | wedhorn | 2013-01-06 14:45:12 -0600 (Sun, 06 Jan 2013) | 12 lines
  
  Rewrite skinny dialing to remove threaded simpleswitch
  
  This rewrite changes skinny dialing from the threaded simpleswitch
  to a scheduled timeout approach. There were some underlying issues
  with the threaded simple switch with occasional corruption and
  possible segfaults.
  
  Review: https://reviewboard.asterisk.org/r/2240/
  ........
  
  Merged revisions 378622 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378624 | wedhorn | 2013-01-06 15:09:43 -0600 (Sun, 06 Jan 2013) | 6 lines
  
  Add group and namedgroup pickup to skinny
  
  Above says it all. Code by snuff, cleaned up by me. 
  
  Review: https://reviewboard.asterisk.org/r/2246/
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2013-01-06 21:20:13 +00:00
Automerge script 2765de67b6 Merged revisions 378593 via svnmerge from
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  r378593 | jrose | 2013-01-04 17:14:54 -0600 (Fri, 04 Jan 2013) | 23 lines
  
  res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create
  
  Under some circumstances, libsrtp's srtp_create function deallocates memory that
  it wasn't initially responsible for allocating. Because we weren't initially
  aware of this behavior, this memory was still used in spite of being unallocated
  during the course of the srtp_unprotect function. A while back I made a patch
  which would set this value to NULL, but that exposed a possible condition where
  we would then try to check a member of the struct which would cause a segfault.
  In order to address these problems, ast_srtp_unprotect will now set an error value
  when it ends without a valid SRTP session which will result in the caller of
  srtp_unprotect observing this error and hanging up the relevant channel instead of
  trying to keep using the invalid session address.
  
  (closes issue ASTERISK-20499)
  Reported by: Tootai
  Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header
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  Merged revisions 378591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378592 from http://svn.asterisk.org/svn/asterisk/branches/11
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2013-01-04 23:19:39 +00:00
Automerge script 77150eecd9 Merged revisions 378565,378585 via svnmerge from
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  r378565 | elguero | 2013-01-04 15:20:12 -0600 (Fri, 04 Jan 2013) | 27 lines
  
  Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field
  
  On a multihomed server when sending a NOTIFY message, we were not figuring out
  which network should be used to contact the peer.
  
  This patch fixes the problem by calling ast_sip_ouraddrfor() and then
  build_via() so that our NOTIFY message contains the correct IP address.
  
  Also, a debug message is being added to help follow the call-id changes that
  occur.  This was helpful for confirming that the IP address was set properly
  since the call-id contains the IP address.  It also will be helpful for
  troubleshooting purposes when following a call in the debug logs.
  
  (closes issue ASTERISK-20805)
  Reported by: Bryan Hunt
  Tested by: Bryan Hunt, Michael L. Young
  Patches:
      asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026)
  
  Review: https://reviewboard.asterisk.org/r/2255/
  ........
  
  Merged revisions 378554 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378559 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378585 | kmoore | 2013-01-04 16:19:16 -0600 (Fri, 04 Jan 2013) | 13 lines
  
  Fix pjproject compilation in certain circumstances
  
  On a fresh checkout of Asterisk 11, running make before ./configure
  could cause the pjproject subdirectory to get in an odd state that
  would prevent compilation. This patch by Tilghman prevents that from
  occurring.
  
  (closes issue ASTERISK-20681)
  Patch-by: Tilghman Lesher
  ........
  
  Merged revisions 378582 from http://svn.asterisk.org/svn/asterisk/branches/11
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2013-01-04 22:19:35 +00:00
Automerge script 98fe918625 Merged revisions 378557 via svnmerge from
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  r378557 | file | 2013-01-04 15:18:07 -0600 (Fri, 04 Jan 2013) | 11 lines
  
  Don't pass STUN packets through the SRTP unprotect function.
  
  (closes issue AST-1036)
  Reported by: jbigelow
  ........
  
  Merged revisions 378553 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378555 from http://svn.asterisk.org/svn/asterisk/branches/11
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2013-01-04 21:18:39 +00:00
Automerge script cd0aef75ca Merged revisions 378543 via svnmerge from
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  r378543 | lathama | 2013-01-04 10:44:33 -0600 (Fri, 04 Jan 2013) | 6 lines
  
  Doxygen Cleanups
  
  Baseline clean up of formating to make room for extended documentation
  
  (issue ASTERISK-20259)
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Automerge script 7980f15ada Merged revisions 378516 via svnmerge from
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  r378516 | elguero | 2013-01-03 16:14:20 -0600 (Thu, 03 Jan 2013) | 25 lines
  
  Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present
  
  When the "h" extension is present within the context of the queue, all calls
  are being reported COMPLETECALLER even when the agent is hanging up the call.
  
  This patch checks to see if the agent hung-up or not instead of only relying on
  checking if the queue (caller) channel hung-up or not.  It would appear that
  having the h extension in the mix, the pbx goes to the h extension,
  "hanging-up" the queue channel and triggering the reporting of COMPLETECALLER.
  
  (closes issue ASTERISK-20743)
  Reported by: call
  Tested by: call, Michael L. Young
  Patches: 
      asterisk-20743-q-cmplt-caller.diff 
                                       uploaded by Michael L. Young (license 5026)
  
  Review: https://reviewboard.asterisk.org/r/2256/
  ........
  
  Merged revisions 378514 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378515 from http://svn.asterisk.org/svn/asterisk/branches/11
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2013-01-03 22:19:46 +00:00
Automerge script d359e1225e Merged revisions 378488 via svnmerge from
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  r378488 | rmudgett | 2013-01-03 13:42:54 -0600 (Thu, 03 Jan 2013) | 15 lines
  
  chan_agent: Fix wrapup time wait response.
  
  * Made agent_cont_sleep() and agent_ack_sleep() stop waiting if the wrapup 
  time expires.  agent_cont_sleep() had tried but returned the wrong value 
  to stop waiting.  
  
  * Made agent_ack_sleep() take a struct agent_pvt pointer instead of a void
  pointer for better type safety.
  ........
  
  Merged revisions 378486 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378487 from http://svn.asterisk.org/svn/asterisk/branches/11
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2013-01-03 20:19:37 +00:00
Automerge script f824b845ce Merged revisions 378458,378460 via svnmerge from
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  r378458 | rmudgett | 2013-01-03 12:47:29 -0600 (Thu, 03 Jan 2013) | 18 lines
  
  chan_agent: Misc code cleanup.
  
  * Fix off-nominal path resource cleanup in agent_request().
  
  * Create agent_pvt_destroy() to eliminate inlined versions in many places.
  
  * Pull invariant code out of loop in add_agent().
  
  * Remove redundant module user references in login_exec().
  
  * Remove unused struct agent_pvt logincallerid[] member.
  ........
  
  Merged revisions 378456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378457 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378460 | kmoore | 2013-01-03 12:51:43 -0600 (Thu, 03 Jan 2013) | 13 lines
  
  Add missing test event
  
  This test event was missing from channel.c causing the dial_LS_options
  test to fail intermittently because of a race condition where most code
  paths emitted the test event but this one did not. The dial_LS_options
  test should stop bouncing now.
  ........
  
  Merged revisions 378455 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378459 from http://svn.asterisk.org/svn/asterisk/branches/11
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2013-01-03 19:19:40 +00:00
Automerge script 06b4ef8317 Merged revisions 378429 via svnmerge from
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  r378429 | rmudgett | 2013-01-03 11:48:14 -0600 (Thu, 03 Jan 2013) | 10 lines
  
  chan_agent: Fix agent_indicate() locking.
  
  Avoid deadlock potential with local channels and simplify the locking.
  ........
  
  Merged revisions 378427 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378428 from http://svn.asterisk.org/svn/asterisk/branches/11
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Automerge script 01250e2585 Merged revisions 378410,378412,378414 via svnmerge from
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  r378410 | mjordan | 2013-01-03 09:37:31 -0600 (Thu, 03 Jan 2013) | 13 lines
  
  Prevent crashes in res_xmpp when receiving large messages
  
  Similar to r378287, res_xmpp was marshaling data read from an external source
  onto the stack. For a sufficiently large message, this could cause a stack
  overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by
  removing the stack allocation, as it was unnecessary.
  
  (issue ASTERISK-20658)
  Reported by: wdoekes
  ........
  
  Merged revisions 378409 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378412 | file | 2013-01-03 09:40:21 -0600 (Thu, 03 Jan 2013) | 11 lines
  
  Prevent exhaustion of system resources through exploitation of event cache
  
  This patch changes res_xmpp to no longer cache events under certain circumstances.
  
  (issue ASTERISK-20175)
  Reported by: Russell Bryant, Leif Madsen, Joshua Colp
  Tested by: kmoore
  ........
  
  Merged revisions 378411 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378414 | tilghman | 2013-01-03 10:04:11 -0600 (Thu, 03 Jan 2013) | 11 lines
  
  Add aliases to the Directory.
  
  This is an interesting feature that allows additional strings to be used to
  search the Directory, primarily intended to be used with nicknames, but could
  be used with affiliations and the like.  Because the name field is used in
  more than one place (such as email notifications), it is important that these
  additional strings not be placed in the name field, but be specified
  separately.
  
  Review: https://reviewboard.asterisk.org/r/2244/
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Automerge script b375abae50 Merged revisions 378374,378377,378384 via svnmerge from
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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
  ........
  
  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)
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Automerge script 675914bb17 Merged revisions 378322 via svnmerge from
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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
  ........
  
  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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Automerge script 5966364588 Merged revisions 378288 via svnmerge from
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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
  ........
  
  Merged revisions 378287 from http://svn.asterisk.org/svn/asterisk/branches/11
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Automerge script 95cb4fa886 Merged revisions 378259 via svnmerge from
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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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Automerge script fe1822bf92 Merged revisions 378248-378249 via svnmerge from
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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'
........
  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
  ........
  
  Merged revisions 378219 from http://svn.asterisk.org/svn/asterisk/branches/11
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Automerge script 550581db83 Merged revisions 378166 via svnmerge from
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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
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  Merged revisions 378165 from http://svn.asterisk.org/svn/asterisk/branches/11
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Automerge script e0fa5cf52d Merged revisions 378122 via svnmerge from
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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
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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/
  ........
  
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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.
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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.
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  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)
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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)
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  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
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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)
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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/)
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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/
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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)
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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
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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
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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
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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
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Mark Michelson 7995bc63c7 Some general cleanup, plus we now send state changes when threads activate.
This is now ready for review board, imo!



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Mark Michelson 2abda64003 Add threadpool option version check into threadpool creation routine.
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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.



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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.



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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
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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
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  Merged revisions 377740 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-11 02:19:39 +00:00
Automerge script 56a9f5b6cb Merged revisions 377707,377711 via svnmerge from
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
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-11 01:19:39 +00:00