[svn-commits] qwell: branch qwell/fun_with_transports r385106 - in /team/qwell/fun_with_tra...

SVN commits to the Digium repositories svn-commits at lists.digium.com
Tue Apr 9 09:45:54 CDT 2013


Author: qwell
Date: Tue Apr  9 09:45:34 2013
New Revision: 385106

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=385106
Log:
Multiple revisions 383730,383757,383801,383844,383882,383914,383924,383927,383951,383986,384032,384052,384124,384167,384203,384221,384242,384268,384304,384330,384392,384418,384454,384476,384490,384516,384533,384535,384550,384619,384673,384713,384736,384749,384762,384831,384859,384891,384918,384920,384967,384975

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  r383730 | root | 2013-03-25 12:17:41 -0500 (Mon, 25 Mar 2013) | 40 lines
  
  Multiple revisions 383726,383728
  
  ........
    r383726 | dlee | 2013-03-25 11:19:55 -0500 (Mon, 25 Mar 2013) | 28 lines
    
    Move NewCallerid, HangupRequest and SoftHangupRequest to Stasis
    
    HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis
    messages, with the cause code as an optional field in the blob.
    
    NewCallerid now simply watches for changes in the callerid information
    in channel snapshots, and creates the AMI event appropriately.
    
    Since the original NewCallerid event honored the channelvars setting
    in manager.conf, the channel variables configured there had to become
    a part of the channel snapshot. These are now a part of every snapshot
    based event, making the configuration description "every time a
    channel-oriented event is emitted" less of a lie.
    
    There a a few other changes wrapped up in here as well.
    
     * When ast_channel_topic() is given NULL for a channel, it returns
       the ast_channel_topic_all() topic instead of NULL. This can clean
       up a lot of NULL checking we're doing currently.
     * The fields Cause and Cause-txt were removed from the base channel
       information and put only on the Hangup events, since those fields
       are meaningless outside of a Hangup event.
     * Removed the pipe-delimiter processing of the channelvars field,
       since that's been deprecated forever.
    
    (closes issue ASTERISK-21096)
    Review: https://reviewboard.asterisk.org/r/2405/
  ........
    r383728 | dlee | 2013-03-25 12:12:03 -0500 (Mon, 25 Mar 2013) | 1 line
    
    install_prereq: Adding jansson-devel to RH packages
  ........
  
  Merged revisions 383726,383728 from file:///srv/subversion/repos/asterisk/trunk
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  r383757 | root | 2013-03-25 15:17:39 -0500 (Mon, 25 Mar 2013) | 18 lines
  
  Multiple revisions 383747,383753-383754
  
  ........
    r383747 | dlee | 2013-03-25 14:28:04 -0500 (Mon, 25 Mar 2013) | 1 line
    
    install_prereq: removed some out-of-date comments
  ........
    r383753 | kmoore | 2013-03-25 15:07:00 -0500 (Mon, 25 Mar 2013) | 2 lines
    
    Fix missing ' ' around '='
  ........
    r383754 | kmoore | 2013-03-25 15:15:09 -0500 (Mon, 25 Mar 2013) | 2 lines
    
    Fix typo
  ........
  
  Merged revisions 383747,383753-383754 from file:///srv/subversion/repos/asterisk/trunk
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  r383801 | root | 2013-03-25 19:17:38 -0500 (Mon, 25 Mar 2013) | 18 lines
  
  Set the CALLERID(dnid-num-plan) for incoming ISDN calls.
  
  The CALLEDTON channel variable is set for incoming ISDN calls to the lower
  7 bits of the Q.931 type-of-number/numbering-plan octet.  The
  CALLERID(dnid-num-plan) should have the same value.
  
  (closes issue ASTERISK-21248)
  Reported by: rmudgett
  ........
  
  Merged revisions 383796 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 383798 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 383799 from file:///srv/subversion/repos/asterisk/trunk
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  r383844 | root | 2013-03-25 21:17:40 -0500 (Mon, 25 Mar 2013) | 60 lines
  
  Multiple revisions 383837-383838,383841
  
  ........
    r383837 | russell | 2013-03-25 20:38:56 -0500 (Mon, 25 Mar 2013) | 19 lines
    
    Fix multi-station answer race condition.
    
    When an SLA trunk is ringing (inbound call on the trunk) Asterisk will
    make outbound calls to the stations that have that trunk.  If more than
    one station answers the call at the same time, all channels other than
    the first one to answer are left in a bad state.  The channel gets
    leaked, is not connected to anything, and there's no way to get rid of
    it.
    
    We now properly clean up these losing channels by hanging up on them.
    Since they lost the race, as we process their answer, there is no
    ringing trunk for them to answer.
    ........
    
    Merged revisions 383835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
    ........
    
    Merged revisions 383836 from http://svn.asterisk.org/svn/asterisk/branches/11
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    r383838 | russell | 2013-03-25 20:46:39 -0500 (Mon, 25 Mar 2013) | 7 lines
    
    Suppress compiler warning.
    
    This code caused a compiler warning when --enable-dev-mode was not used.
    The warning was that this variable was set but not used.  That was indeed
    the case as the only place this is used is as an argument to SKINNY_DEBUG
    which is compiled out when not in dev mode.
  ........
    r383841 | mjordan | 2013-03-25 20:58:45 -0500 (Mon, 25 Mar 2013) | 22 lines
    
    Resolve deadlock between pending CDR and batch CDR locks
    
    r375757 attempted to resolve a race condition between multiple submissions of
    CDRs while in batch mode from attempting to destroy the scheduled batch
    submission by extending the batch CDR lock. Unfortunately, this causes a
    deadlock between the pending CDR lock and the batch CDR lock. This patch
    resolves the intent of r375757 by simply providing a new lock that protects
    the scheduling of the batches. The original batch CDR lock is kept to protect
    manipulation of the batch CDR settings, but has been placed such that it
    is not held when the pending lock is held.
    
    Thanks to Chase Venters for providing lock analysis on the issue.
    
    (issue ASTERISK-21162)
    Reported by: Chase Venters
    ........
    
    Merged revisions 383839 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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    Merged revisions 383840 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 383837-383838,383841 from file:///srv/subversion/repos/asterisk/trunk
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  r383882 | root | 2013-03-25 22:17:37 -0500 (Mon, 25 Mar 2013) | 44 lines
  
  Resolve deadlock between SIP registration and channel based functions
  
  In r373424, several reentrancy problems in chan_sip were addressed. As a
  result, the SIP channel driver is now properly locking the channel driver
  private information in certain operations that it wasn't previously. This
  exposed two latent problems either in register_verify or by functions called
  by register_verify. This includes:
   * Holding the private lock while calling sip_send_mwi_to_peer. This can create
     a new sip_pvt via sip_alloc, which will obtain the channel container lock.
     This is a locking inversion, as any channel related lock must be obtained
     prior to obtaining the SIP channel technology private lock.
  
     Note that this issue was already fixed in Asterisk 11.
  
   * Holding the private lock while calling sip_poke_peer. In the same vein as
     sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing
     the same locking inversion.
  
  Note that this locking inversion typically occured when CLI commands were run
  while a SIP REGISTER request was being processed, as many CLI commands (such
  as 'sip show channels', 'core show channels', etc.) have to obtain the channel
  container lock.
  
  (issue ASTERISK-21068)
  Reported by: Nicolas Bouliane
  
  (issue ASTERISK-20550)
  Reported by: David Brillert
  
  (issue ASTERISK-21314)
  Reported by: Badalian Vyacheslav
  
  (issue ASTERISK-21296)
  Reported by: Gabriel Birke
  ........
  
  Merged revisions 383863 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 383878 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 383879 from file:///srv/subversion/repos/asterisk/trunk
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  r383914 | file | 2013-03-26 13:24:53 -0500 (Tue, 26 Mar 2013) | 2 lines
  
  Don't attempt to authenticate ACKs.
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  r383924 | file | 2013-03-26 18:33:24 -0500 (Tue, 26 Mar 2013) | 2 lines
  
  Just want to see what this does to Bamboo...
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  r383927 | root | 2013-03-26 19:17:41 -0500 (Tue, 26 Mar 2013) | 5 lines
  
  Remove the noop handler from sorcery so it does not produce an empty value.
  ........
  
  Merged revisions 383925 from file:///srv/subversion/repos/asterisk/trunk
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  r383951 | root | 2013-03-27 03:17:38 -0500 (Wed, 27 Mar 2013) | 15 lines
  
  Fix skinny encall button to not blind xfer.
  
  The softbutton endcall should not turn a transfer into a blind transfer but
  hangup the exten being called and leave the original call on hold. This does
  that.
  
  (closes issue ASTERISK-21321)
  Reported by: wedhorn
  Tested by: snuffy, myself
  Patches: 
      skinny-xferendcall01.diff uploaded by wedhorn (license 5019)
  ........
  
  Merged revisions 383948 from file:///srv/subversion/repos/asterisk/trunk
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  r383986 | root | 2013-03-27 10:18:21 -0500 (Wed, 27 Mar 2013) | 50 lines
  
  Multiple revisions 383975,383980
  
  ........
    r383975 | mjordan | 2013-03-27 09:28:36 -0500 (Wed, 27 Mar 2013) | 16 lines
    
    AST-2013-001: Prevent buffer overflow through H.264 format negotiation
    
    The format attribute resource for H.264 video performs an unsafe read against a
    media attribute when parsing the SDP. The value passed in with the format
    attribute is not checked for its length when parsed into a fixed length buffer.
    This patch resolves the vulnerability by only reading as many characters from
    the SDP value as will fit into the buffer.
    
    (closes issue ASTERISK-20901)
    Reported by: Ulf Harnhammar
    patches:
      h264_overflow_security_patch.diff uploaded by jrose (License 6182)
    ........
    
    Merged revisions 383973 from http://svn.asterisk.org/svn/asterisk/branches/11
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    r383980 | mjordan | 2013-03-27 09:39:11 -0500 (Wed, 27 Mar 2013) | 24 lines
    
    AST-2013-002: Prevent denial of service in HTTP server
    
    AST-2012-014, fixed in January of this year, contained a fix for Asterisk's
    HTTP server for a remotely-triggered crash. While the fix put in place fixed
    the possibility for the crash to be triggered, a denial of service vector still
    exists with that solution if an attacker sends one or more HTTP POST requests
    with very large Content-Length values. This patch resolves this by capping
    the Content-Length at 1024 bytes. Any attempt to send an HTTP POST with
    Content-Length greater than this cap will not result in any memory allocation.
    The POST will be responded to with an HTTP 413 "Request Entity Too Large"
    response.
    
    This issue was reported by Christoph Hebeisen of TELUS Security Labs
    
    (closes issue ASTERISK-20967)
    Reported by: Christoph Hebeisen
    patches:
      AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
      AST-2013-002-10.diff uploaded by mmichelson (License 5049)
      AST-2013-002-11.diff uploaded by mmichelson (License 5049)
    ........
    
    Merged revisions 383978 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 383975,383980 from file:///srv/subversion/repos/asterisk/trunk
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  r384032 | root | 2013-03-27 11:17:40 -0500 (Wed, 27 Mar 2013) | 38 lines
  
  AST-2013-003: Prevent username disclosure in SIP channel driver
  
  When authenticating a SIP request with alwaysauthreject enabled, allowguest
  disabled, and autocreatepeer disabled, Asterisk discloses whether a user
  exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The
  information is disclosed when:
   * A "407 Proxy Authentication Required" response is sent instead of a
     "401 Unauthorized" response
   * The presence or absence of additional tags occurs at the end of "403
     Forbidden" (such as "(Bad Auth)")
   * A "401 Unauthorized" response is sent instead of "403 Forbidden" response
     after a retransmission
   * Retransmission are sent when a matching peer did not exist, but not when a
     matching peer did exist.
  
  This patch resolves these various vectors by ensuring that the responses sent
  in all scenarios is the same, regardless of the presence of a matching peer.
  
  This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
  the testing and the solution to this problem was done by Walter as well - a
  huge thanks to his tireless efforts in finding all the ways in which this
  setting didn't work, providing automated tests, and working with Kinsey on
  getting this fixed.
  
  (closes issue ASTERISK-21013)
  Reported by: wdoekes
  Tested by: wdoekes, kmoore
  patches:
    AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674)
    AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674)
    AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674)
  ........
  
  Merged revisions 384003 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 384019 from file:///srv/subversion/repos/asterisk/trunk
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  r384052 | root | 2013-03-27 12:17:44 -0500 (Wed, 27 Mar 2013) | 25 lines
  
  Fix white noise on SRTP decryption
  
  When res_rtp_asterisk.c was altered to avoid attempting to apply
  unprotect algorithms to non-audio RTP packets, the test used was
  incorrect. This caused the audio packets to not be decrypted and
  resulted in loud white noise on the other endpoint (or both endpoints
  depending on the call legs involved). The test now properly checks the
  version field in the RTP header to ensure that RTP and RTCP are
  decrypted while other types of packets are not.
  
  (closes issue ASTERISK-21323)
  Reported by: andrea
  Tested by: Kinsey Moore, andrea, John Bigelow
  Patches:
      whitenoise_fix.diff uploaded by Kinsey Moore
  ........
  
  Merged revisions 384048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 384049 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 384050 from file:///srv/subversion/repos/asterisk/trunk
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  r384124 | root | 2013-03-27 14:17:52 -0500 (Wed, 27 Mar 2013) | 23 lines
  
  Fix a file descriptor leak in off nominal path
  
  While looking at the security vulnerability in ASTERISK-20967, Walter noticed
  a file descriptor leak and some other issues in off nominal code paths. This
  patch corrects them.
  
  Note that this patch is not related to the vulnerability in ASTERISK-20967,
  but the patch was placed on that issue.
  
  (closes issue ASTERISK-20967)
  Reported by: wdoekes
  patches:
    issueA20967_file_leak_and_unused_wkspace.patch uploaded by wdoekes (License 5674)
  ........
  
  Merged revisions 384118 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 384119 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 384120 from file:///srv/subversion/repos/asterisk/trunk
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  r384167 | root | 2013-03-27 15:17:38 -0500 (Wed, 27 Mar 2013) | 11 lines
  
  Address uninitialized conditional that valgrind found
  ........
  
  Merged revisions 384162 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 384163 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 384164 from file:///srv/subversion/repos/asterisk/trunk
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  r384203 | root | 2013-03-27 17:17:36 -0500 (Wed, 27 Mar 2013) | 5 lines
  
  Added a doxygen group for Stasis messages and topics
  ........
  
  Merged revisions 384201 from file:///srv/subversion/repos/asterisk/trunk
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  r384221 | root | 2013-03-27 18:17:46 -0500 (Wed, 27 Mar 2013) | 5 lines
  
  Convert MWI state message type to the new stasis naming convention
  ........
  
  Merged revisions 384219 from file:///srv/subversion/repos/asterisk/trunk
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  r384242 | file | 2013-03-28 07:52:58 -0500 (Thu, 28 Mar 2013) | 2 lines
  
  Change a verbose message to a debug message as on low expiration times it can be logged quite frequently.
........
  r384268 | root | 2013-03-28 11:18:02 -0500 (Thu, 28 Mar 2013) | 5 lines
  
  Break the world. Stasis message type accessors should now all be named correctly.
  ........
  
  Merged revisions 384261 from file:///srv/subversion/repos/asterisk/trunk
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  r384304 | root | 2013-03-28 19:17:38 -0500 (Thu, 28 Mar 2013) | 19 lines
  
  Add uuid wrapper API call ast_uuid_generate_str().
  
  * Updated test_uuid.c to test the new API call.
  
  * Made system use the new API call to eliminate "10's of lines" where
  used.
  
  * Fixed untested ast_strdup() return in stasis_subscribe() by eliminating
  the need for it.  struct stasis_subscription now contains the uniqueid[]
  string.
  
  * Fixed some issues in exchangecal_write_event():
    Create uid with enough space for a UUID string to avoid a realloc.
    Fix off by one error if the calendar event provided a UUID string.
    There is no need to check for NULL before calling ast_free().
  ........
  
  Merged revisions 384302 from file:///srv/subversion/repos/asterisk/trunk
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  r384330 | root | 2013-03-29 12:17:42 -0500 (Fri, 29 Mar 2013) | 22 lines
  
  app_voicemail: Add blank argument to externnotify if no context argument
  
  At least one call to run_externnotify provides a NULL context parameter and
  because the snprintf statement doesn't account for a NULL context parameter,
  it simply writes '(null)' to the arguments string instead. This patch makes
  it write two quotes back to back for that argument instead in the event of
  a NULL context.
  
  (closes issue ASTERISK-18207)
  Reported by: Barry L. Kline
  Patches:
  	modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930)
  ........
  
  Merged revisions 384325 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 384326 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 384327 from file:///srv/subversion/repos/asterisk/trunk
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  r384392 | root | 2013-03-30 00:17:38 -0500 (Sat, 30 Mar 2013) | 19 lines
  
  Multiple revisions 384389-384390
  
  ........
    r384389 | mjordan | 2013-03-30 00:06:54 -0500 (Sat, 30 Mar 2013) | 8 lines
    
    Convert TestEvent AMI events over to Stasis Core
    
    This patch migrates the TestEvent AMI events to first be dispatched over the
    Stasis-Core message bus. This helps to preserve the ordering of the events
    with other events in the AMI system, such as the various channel related
    events.
  ........
    r384390 | mjordan | 2013-03-30 00:15:42 -0500 (Sat, 30 Mar 2013) | 2 lines
    
    Properly format an intmax_t value
  ........
  
  Merged revisions 384389-384390 from file:///srv/subversion/repos/asterisk/trunk
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  r384418 | root | 2013-04-01 09:18:08 -0500 (Mon, 01 Apr 2013) | 58 lines
  
  Multiple revisions 384412-384413,384416
  
  ........
    r384412 | dlee | 2013-04-01 08:34:51 -0500 (Mon, 01 Apr 2013) | 19 lines
    
    Fix parallel make problems.
    
    Occasionally, make -j would fail due to missing includes, or other
    unusual errors.
    
    This was due to the 'cleantest' target, which was designed to force a
    make clean when some change in the code would cause the typical
    depedency checking to fail. Several targets in the main Makefile did
    not depend upon cleantest, hence would run in parallel to it. By
    adding the dependency, make -j runs happily now.
    
    Review: https://reviewboard.asterisk.org/r/2418/
    ........
    
    Merged revisions 384410 from http://svn.asterisk.org/svn/asterisk/branches/1.8
    ........
    
    Merged revisions 384411 from http://svn.asterisk.org/svn/asterisk/branches/11
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    r384413 | dlee | 2013-04-01 08:37:51 -0500 (Mon, 01 Apr 2013) | 22 lines
    
    stasis: Fixed message ordering issues when forwarding
    
    This patch fixes an issue of message ordering that occurs when
    multiple topics are forwarded to an aggregator topic (such as
    ast_channel_topic_all()).
    
    It is (very reasonably) expected that the rules governing message
    dispatch order still apply, so long as the messages start from the
    same thread, and are received by the same subscription. Because the
    existing code had an additional layer of dispatching via the Stasis
    thread pool for forwards, those promises couldn't be kept.
    
    Forwarding subscriptions no longer have their own mailbox, and now
    dispatch directly from the forwarding topic's stasis_publish()
    call. This means that the topic's lock is held for the duration of not
    only a message's dispatch, but the dispatch of all the forwards. This
    shouldn't be a problem right now, but if an aggregator topic had many
    subscribers, it could become a problem. But I figure we can write more
    clever code when the time comes, if necessary.
    
    Review: https://reviewboard.asterisk.org/r/2419/
  ........
    r384416 | file | 2013-04-01 09:10:46 -0500 (Mon, 01 Apr 2013) | 5 lines
    
    Remove silly use of strncmp.
    ........
    
    Merged revisions 384414 from http://svn.asterisk.org/svn/asterisk/branches/11
  ........
  
  Merged revisions 384412-384413,384416 from file:///srv/subversion/repos/asterisk/trunk
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  r384454 | root | 2013-04-01 10:17:36 -0500 (Mon, 01 Apr 2013) | 9 lines
  
  Make appropriate items parse using '|' instead of ','
  
  This patch fixes a bug introduced in r76703, wherein Asterisk could only parse
  arguments in the so-called 'recommended' way, e.g., NoOp(foo,bar). The proper
  syntax of NoOp,foo|bar is now parsed correctly.
  ........
  
  Merged revisions 384452 from file:///srv/subversion/repos/asterisk/trunk
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  r384476 | mmichelson | 2013-04-01 12:07:52 -0500 (Mon, 01 Apr 2013) | 3 lines
  
  Be sure to properly free authentication response if we don't actually send it.
........
  r384490 | root | 2013-04-01 15:17:36 -0500 (Mon, 01 Apr 2013) | 31 lines
  
  install_prereq: Build jansson from source, when necessary
  
  When r383579 was committed, it made Jansson a required dependency.
  
  While libjansson-dev and jansson-devel are available on recent
  distros, some older (but still supported) distros don't have
  it. There's a pull request[1] to get it into repoforge, but that still
  doesn't help everyone. (And helps no one until the pull request is
  merged and packages are built).
  
  This patch adds Jansson install from source to the install_unpackaged()
  function. There are a few gotcha's, which makes this change not
  completely trivial.
  
   * Since Jansson may be installed by a package, don't install from
     source if a package installation can be found
     * libresample may also be installed via package, so I added a
       similar check to that.
   * Since Jansson installs into /usr/local, this patch also adds
     /usr/local/lib to /etc/ld.so.conf.d so that the library can be
     found.
     * The alternative was to install into /usr, but then it gets
       complicated having to deal with EL's /usr/lib{32,64} shenanigans.
  
   [1]: https://github.com/repoforge/rpms/pull/250
  
  Review: https://reviewboard.asterisk.org/r/2414/
  ........
  
  Merged revisions 384488 from file:///srv/subversion/repos/asterisk/trunk
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  r384516 | root | 2013-04-02 07:18:18 -0500 (Tue, 02 Apr 2013) | 9 lines
  
  Make things work again
  
  Sorry folks. ',' are still greater than '|'.
  
  Thanks for playing along :-)
  ........
  
  Merged revisions 384514 from file:///srv/subversion/repos/asterisk/trunk
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  r384533 | file | 2013-04-02 07:25:12 -0500 (Tue, 02 Apr 2013) | 2 lines
  
  Fix description of the res_sip_logger module.
........
  r384535 | root | 2013-04-02 08:17:36 -0500 (Tue, 02 Apr 2013) | 5 lines
  
  Pass the object type name to the configuration framework.
  ........
  
  Merged revisions 384518 from file:///srv/subversion/repos/asterisk/trunk
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  r384550 | root | 2013-04-02 13:17:37 -0500 (Tue, 02 Apr 2013) | 18 lines
  
  Fixed spurious rebuilds of func_version.
  
  func_version.so was being rebuilt every time, because build.h was
  changing every build, because of the cleantest dependency that was
  added in r384410 to fix parallel make bugs.
  
  Now build.h will only be created if it does not exist, which was the
  original behavior of the Makefile.
  ........
  
  Merged revisions 384544 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 384545 from http://svn.asterisk.org/svn/asterisk/branches/11
  ........
  
  Merged revisions 384546 from file:///srv/subversion/repos/asterisk/trunk
........
  r384619 | root | 2013-04-03 11:17:43 -0500 (Wed, 03 Apr 2013) | 11 lines
  
  astobj2: Fix rbtree duplicate handling.
  
  OBJ_PARTIAL_KEY searching a rbtree did not find all possible matches if
  the container did not accept duplicates.
  
  Added matching node bias to indicate which matching node is being searched
  for: first, last, any.
  ........
  
  Merged revisions 384616 from file:///srv/subversion/repos/asterisk/trunk
........
  r384673 | root | 2013-04-03 13:17:36 -0500 (Wed, 03 Apr 2013) | 13 lines
  
  Update documentation for CHANNEL function
  
  Document that you can read/write the 'accountcode' and 'amaflags' on a channel.
  ........
  
  Merged revisions 384640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 384641 from http://svn.asterisk.org/svn/asterisk/branches/11
  ........
  
  Merged revisions 384642 from file:///srv/subversion/repos/asterisk/trunk
........
  r384713 | root | 2013-04-03 16:17:38 -0500 (Wed, 03 Apr 2013) | 40 lines
  
  Multiple revisions 384696,384711
  
  ........
    r384696 | rmudgett | 2013-04-03 15:20:09 -0500 (Wed, 03 Apr 2013) | 26 lines
    
    chan_dahdi: Add inband_on_proceeding compatibility option.
    
    The new inband_on_proceeding option causes Asterisk to assume inband audio
    may be present when a PROCEEDING message is received.
    
    Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
    attached to the B channel at this time without explicitly sending the
    progress indicator ie informing the CPE side to attach to the B channel
    for audio.  However, some non-compliant ISDN switches send a PROCEEDING
    without the progress indicator ie indicating inband audio is available and
    assume that the CPE device has connected the media path for listening to
    ringback and other messages.
    
    ASTERISK-17834 which causes this issue was dealing with a non-compliant
    network switch.
    
    (closes issue ASTERISK-21151)
    Reported by: Gianluca Merlo
    Tested by: rmudgett
    ........
    
    Merged revisions 384685 from http://svn.asterisk.org/svn/asterisk/branches/1.8
    ........
    
    Merged revisions 384689 from http://svn.asterisk.org/svn/asterisk/branches/11
  ........
    r384711 | rmudgett | 2013-04-03 15:27:11 -0500 (Wed, 03 Apr 2013) | 4 lines
    
    chan_dahdi: Change inband_on_proceeding option default to no/disabled.
    
    (issue ASTERISK-21151)
  ........
  
  Merged revisions 384696,384711 from file:///srv/subversion/repos/asterisk/trunk
........
  r384736 | kharwell | 2013-04-04 10:54:27 -0500 (Thu, 04 Apr 2013) | 12 lines
  
  SIP video support.
  
  Adding in video support and capabilities.  This addition handles negotiation
  of video streams, including attribute negotiation, and also the video source
  update control frame (initiates a fast video update request - RFC 5168).
  
  Since there was similar functionality between the audio and video these
  modifications also merged the handling of the two media types into a single
  file, res_sip_sdp_rtp.c
  
  (issue ASTERISK-21077)
........
  r384749 | kharwell | 2013-04-04 11:57:09 -0500 (Thu, 04 Apr 2013) | 1 line
  
  fixed mixed declarations in code
........
  r384762 | root | 2013-04-04 13:17:37 -0500 (Thu, 04 Apr 2013) | 5 lines
  
  Separate some event struct definitions from instantiation.
  ........
  
  Merged revisions 384760 from file:///srv/subversion/repos/asterisk/trunk
........
  r384831 | root | 2013-04-05 16:17:57 -0500 (Fri, 05 Apr 2013) | 32 lines
  
  Fix For Not Overriding The Default Settings In chan_sip
  
  The initial report was that the "nat" setting in the [general] section was not
  having any effect in overriding the default setting.  Upon confirming that this
  was happening and looking into what was causing this, it was discovered that
  other default settings would not be overriden as well.
  
  This patch works similar to what occurs in build_peer().  We create a temporary
  ast_flags structure and using a mask, we override the default settings with
  whatever is set in the [general] section.
  
  In the bug report, the reporter who helped to test this patch noted that the
  directmedia settings were being overriden properly as well as the nat settings.
  
  This issue is also present in Asterisk 1.8 and a separate patch will be applied
  to it.
  
  (issue ASTERISK-21225)
  Reported by: Alexandre Vezina
  Tested by: Alexandre Vezina, Michael L. Young
  Patches:
    asterisk-21225-handle-options-default-prob_v4.diff
  						Michael L. Young (license 5026)
  
  Review: https://reviewboard.asterisk.org/r/2385/
  ........
  
  Merged revisions 384827 from http://svn.asterisk.org/svn/asterisk/branches/11
  ........
  
  Merged revisions 384828 from file:///srv/subversion/repos/asterisk/trunk
........
  r384859 | root | 2013-04-06 11:18:23 -0500 (Sat, 06 Apr 2013) | 7 lines
  
  Add a res_sorcery_astdb module which uses the astdb to persist objects.
  
  Review: https://reviewboard.asterisk.org/r/2420/
  ........
  
  Merged revisions 384857 from file:///srv/subversion/repos/asterisk/trunk
........
  r384891 | root | 2013-04-08 09:17:57 -0500 (Mon, 08 Apr 2013) | 29 lines
  
  Stasis application WebSocket support
  
  This is the API that binds the Stasis dialplan application to external
  Stasis applications. It also adds the beginnings of WebSocket
  application support.
  
  This module registers a dialplan function named Stasis, which is used
  to put a channel into the named Stasis app. As a channel enters and
  leaves the Stasis diaplan applcation, the Stasis app receives a
  'stasis-start' and 'stasis-end' events.
  
  Stasis apps register themselves using the stasis_app_register and
  stasis_app_unregister functions. Messages are sent to an appliction
  using stasis_app_send.
  
  Finally, Stasis apps control channels through the use of the
  stasis_app_control object, and the family of stasis_app_control_*
  functions.
  
  Other changes along for the ride are:
   * An ast_frame_dtor function that's RAII_VAR safe
   * Some common JSON encoders for name/number, timeval, and
     context/extension/priority
  
  Review: https://reviewboard.asterisk.org/r/2361/
  ........
  
  Merged revisions 384879 from file:///srv/subversion/repos/asterisk/trunk
........
  r384918 | root | 2013-04-08 10:17:46 -0500 (Mon, 08 Apr 2013) | 19 lines
  
  Add multi-channel Stasis messages; refactor Dial AMI events to Stasis
  
  This patch does the following:
   * A new Stasis payload has been defined for multi-channel messages. This
     payload can store multiple ast_channel_snapshot objects along with a single
     JSON blob. The payload object itself is opaque; the snapshots are stored
     in a container keyed by roles. APIs have been provided to query for and
     retrieve the snapshots from the payload object.
   * The Dial AMI events have been refactored onto Stasis. This includes dial
     messages in app_dial, as well as the core dialing framework. The AMI events
     have been modified to send out a DialBegin/DialEnd events, as opposed to
     the subevent type that was previously used.
   * Stasis messages, types, and other objects related to channels have been
     placed in their own file, stasis_channels. Unit tests for some of these
     objects/messages have also been written.
  ........
  
  Merged revisions 384910 from file:///srv/subversion/repos/asterisk/trunk
........
  r384920 | mmichelson | 2013-04-08 10:18:18 -0500 (Mon, 08 Apr 2013) | 6 lines
  
  Add outbound authentication support.
  
  This adds module outbound authentication and a digest module that uses
  PJSIP's auth_client API.
........
  r384967 | mmichelson | 2013-04-08 11:25:10 -0500 (Mon, 08 Apr 2013) | 3 lines
  
  Remove automerge props that were merged during outbound_auth merge.
........
  r384975 | root | 2013-04-08 12:17:22 -0500 (Mon, 08 Apr 2013) | 11 lines
  
  Don't attempt a websocket protocol removal if res_http_websocket isn't there
  
  This patch sets the protocols container provided by res_http_websocket to NULL
  when the module gets unloaded and adds the necessary checks when adding/
  removing a websocket protocol. This prevents some FRACKing on an invalid
  pointer to the disposed container if a module that uses res_http_websocket is
  unloaded after it.
  ........
  
  Merged revisions 384942 from file:///srv/subversion/repos/asterisk/trunk
........

Merged revisions 383730,383757,383801,383844,383882,383914,383924,383927,383951,383986,384032,384052,384124,384167,384203,384221,384242,384268,384304,384330,384392,384418,384454,384476,384490,384516,384533,384535,384550,384619,384673,384713,384736,384749,384762,384831,384859,384891,384918,384920,384967,384975 from http://svn.asterisk.org/svn/asterisk/team/group/pimp_my_sip

Added:
    team/qwell/fun_with_transports/apps/app_stasis.c
      - copied unchanged from r384975, team/group/pimp_my_sip/apps/app_stasis.c
    team/qwell/fun_with_transports/apps/app_stasis.exports.in
      - copied unchanged from r384975, team/group/pimp_my_sip/apps/app_stasis.exports.in
    team/qwell/fun_with_transports/apps/stasis_json.c
      - copied unchanged from r384975, team/group/pimp_my_sip/apps/stasis_json.c
    team/qwell/fun_with_transports/include/asterisk/app_stasis.h
      - copied unchanged from r384975, team/group/pimp_my_sip/include/asterisk/app_stasis.h
    team/qwell/fun_with_transports/include/asterisk/stasis_channels.h
      - copied unchanged from r384975, team/group/pimp_my_sip/include/asterisk/stasis_channels.h
    team/qwell/fun_with_transports/main/stasis_channels.c
      - copied unchanged from r384975, team/group/pimp_my_sip/main/stasis_channels.c
    team/qwell/fun_with_transports/res/res_sip/sip_outbound_auth.c
      - copied unchanged from r384975, team/group/pimp_my_sip/res/res_sip/sip_outbound_auth.c
    team/qwell/fun_with_transports/res/res_sip_outbound_authenticator_digest.c
      - copied unchanged from r384975, team/group/pimp_my_sip/res/res_sip_outbound_authenticator_digest.c
    team/qwell/fun_with_transports/res/res_sip_sdp_rtp.c
      - copied unchanged from r384975, team/group/pimp_my_sip/res/res_sip_sdp_rtp.c
    team/qwell/fun_with_transports/res/res_sorcery_astdb.c
      - copied unchanged from r384975, team/group/pimp_my_sip/res/res_sorcery_astdb.c
    team/qwell/fun_with_transports/res/res_stasis_websocket.c
      - copied unchanged from r384975, team/group/pimp_my_sip/res/res_stasis_websocket.c
    team/qwell/fun_with_transports/tests/test_app_stasis.c
      - copied unchanged from r384975, team/group/pimp_my_sip/tests/test_app_stasis.c
    team/qwell/fun_with_transports/tests/test_sorcery_astdb.c
      - copied unchanged from r384975, team/group/pimp_my_sip/tests/test_sorcery_astdb.c
    team/qwell/fun_with_transports/tests/test_stasis_channels.c
      - copied unchanged from r384975, team/group/pimp_my_sip/tests/test_stasis_channels.c
Removed:
    team/qwell/fun_with_transports/res/res_sip_sdp_audio.c
Modified:
    team/qwell/fun_with_transports/   (props changed)
    team/qwell/fun_with_transports/CHANGES
    team/qwell/fun_with_transports/Makefile
    team/qwell/fun_with_transports/UPGRADE-11.txt
    team/qwell/fun_with_transports/UPGRADE.txt
    team/qwell/fun_with_transports/apps/Makefile
    team/qwell/fun_with_transports/apps/app_dial.c
    team/qwell/fun_with_transports/apps/app_meetme.c
    team/qwell/fun_with_transports/apps/app_userevent.c
    team/qwell/fun_with_transports/apps/app_voicemail.c
    team/qwell/fun_with_transports/channels/chan_dahdi.c
    team/qwell/fun_with_transports/channels/chan_gulp.c
    team/qwell/fun_with_transports/channels/chan_iax2.c
    team/qwell/fun_with_transports/channels/chan_mgcp.c
    team/qwell/fun_with_transports/channels/chan_sip.c
    team/qwell/fun_with_transports/channels/chan_skinny.c
    team/qwell/fun_with_transports/channels/chan_unistim.c
    team/qwell/fun_with_transports/channels/sig_pri.c
    team/qwell/fun_with_transports/channels/sig_pri.h
    team/qwell/fun_with_transports/channels/sip/include/sip.h
    team/qwell/fun_with_transports/channels/sip/security_events.c
    team/qwell/fun_with_transports/configs/chan_dahdi.conf.sample
    team/qwell/fun_with_transports/contrib/scripts/install_prereq
    team/qwell/fun_with_transports/funcs/func_channel.c
    team/qwell/fun_with_transports/include/asterisk/app.h
    team/qwell/fun_with_transports/include/asterisk/channel.h
    team/qwell/fun_with_transports/include/asterisk/frame.h
    team/qwell/fun_with_transports/include/asterisk/json.h
    team/qwell/fun_with_transports/include/asterisk/localtime.h
    team/qwell/fun_with_transports/include/asterisk/res_sip.h
    team/qwell/fun_with_transports/include/asterisk/res_sip_session.h
    team/qwell/fun_with_transports/include/asterisk/stasis.h
    team/qwell/fun_with_transports/include/asterisk/test.h
    team/qwell/fun_with_transports/include/asterisk/uuid.h
    team/qwell/fun_with_transports/main/app.c
    team/qwell/fun_with_transports/main/astobj2.c
    team/qwell/fun_with_transports/main/cdr.c
    team/qwell/fun_with_transports/main/channel.c
    team/qwell/fun_with_transports/main/channel_internal_api.c
    team/qwell/fun_with_transports/main/dial.c
    team/qwell/fun_with_transports/main/event.c
    team/qwell/fun_with_transports/main/features.c
    team/qwell/fun_with_transports/main/format_pref.c
    team/qwell/fun_with_transports/main/frame.c
    team/qwell/fun_with_transports/main/http.c
    team/qwell/fun_with_transports/main/json.c
    team/qwell/fun_with_transports/main/manager.c
    team/qwell/fun_with_transports/main/manager_channels.c
    team/qwell/fun_with_transports/main/pbx.c
    team/qwell/fun_with_transports/main/sorcery.c
    team/qwell/fun_with_transports/main/stasis.c
    team/qwell/fun_with_transports/main/stasis_cache.c
    team/qwell/fun_with_transports/main/test.c
    team/qwell/fun_with_transports/main/uuid.c
    team/qwell/fun_with_transports/pbx/pbx_realtime.c
    team/qwell/fun_with_transports/res/res_calendar_exchange.c
    team/qwell/fun_with_transports/res/res_format_attr_h264.c
    team/qwell/fun_with_transports/res/res_http_websocket.c
    team/qwell/fun_with_transports/res/res_jabber.c
    team/qwell/fun_with_transports/res/res_rtp_asterisk.c
    team/qwell/fun_with_transports/res/res_sip.c
    team/qwell/fun_with_transports/res/res_sip.exports.in
    team/qwell/fun_with_transports/res/res_sip/include/res_sip_private.h
    team/qwell/fun_with_transports/res/res_sip/location.c
    team/qwell/fun_with_transports/res/res_sip/sip_configuration.c
    team/qwell/fun_with_transports/res/res_sip/sip_distributor.c
    team/qwell/fun_with_transports/res/res_sip/sip_options.c
    team/qwell/fun_with_transports/res/res_sip_authenticator_digest.c
    team/qwell/fun_with_transports/res/res_sip_logger.c
    team/qwell/fun_with_transports/res/res_sip_registrar.c
    team/qwell/fun_with_transports/res/res_sip_session.c
    team/qwell/fun_with_transports/res/res_sorcery_config.c

[... 7704 lines stripped ...]



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