[asterisk-commits] may: branch may/ooh323_ipv6_direct_rtp r321499 - /team/may/ooh323_ipv6_direct...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sat May 28 02:28:13 CDT 2011


Author: may
Date: Sat May 28 02:28:05 2011
New Revision: 321499

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=321499
Log:
Merged revisions 315453,315504,315649,315670,315674-315675,315766,315811,315855-315856,315895,316054,316095,316213 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

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  r315453 | rmudgett | 2011-04-26 22:02:07 +0400 (Tue, 26 Apr 2011) | 9 lines
  
  Merged revisions 315452 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r315452 | rmudgett | 2011-04-26 13:00:34 -0500 (Tue, 26 Apr 2011) | 1 line
    
    Add missing set of name valid flag when dialing.
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  r315504 | tilghman | 2011-04-26 23:38:41 +0400 (Tue, 26 Apr 2011) | 35 lines
  
  Merged revisions 315503 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r315503 | tilghman | 2011-04-26 14:32:50 -0500 (Tue, 26 Apr 2011) | 28 lines
    
    Merged revisions 315502 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.6.2
    
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      r315502 | tilghman | 2011-04-26 14:22:52 -0500 (Tue, 26 Apr 2011) | 21 lines
      
      Merged revisions 315501 via svnmerge from 
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
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        r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines
        
        Fix the bounds-checking code.
        
        The code that set the bit within the select bitfield was correct, but the
        bounds-checking code was not.  The change to that line uses the new _bitsize
        macro for clarity.  Also, FD_ZERO macro did not zero-out anything but the
        first word of the bitfield, so this could have caused problems with modules
        using that macro with the expanded bitfield.
        
        (closes issue #18773)
         Reported by: jamicque
         Patches: 
               20110423__issue18773.diff.txt uploaded by tilghman (license 14)
         Tested by: chris-mac
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  r315649 | rmudgett | 2011-04-27 02:18:41 +0400 (Wed, 27 Apr 2011) | 28 lines
  
  Merged revisions 315645 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) | 21 lines
    
    The 'e' special extension fails to trigger in at least two cases.
    
    The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if
    any of them do not exist.  Many of the places the 'e' extension was
    supposed to be invoked fail because the priority was set wrong.  There
    were two places where the 'e' extension was not even checked for fall
    back.
    
    * Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T'
    extension check and added the 'e' extension as a fall back to the two
    missing locations.
    
    * Prioritized and optimized some hangup tests associated with the 'e'
    extension.
    
    (closes issue #19136)
    Reported by: kshumard
    Tested by: rmudgett
    
    Review: https://reviewboard.asterisk.org/r/1196/
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  r315670 | twilson | 2011-04-27 02:26:37 +0400 (Wed, 27 Apr 2011) | 39 lines
  
  Merged revisions 315644 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines
    
    Merged revisions 315643 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.6.2
    
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      r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines
      
      Merged revisions 315596 via svnmerge from 
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
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        r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines
        
        Allow transfer loops without allowing forwarding loops
        
        We try to avoid the situation where two phones may be forwarded to each other
        causing an infinite loop by storing each dialed interface in a channel
        datastore and checking the list before dialing out. This works, but currently
        breaks situations like A calls B, A transfers B to C, B transfers C to A, and A
        transfers C to B. Since human interaction is happening here and not an
        automated forwarding loop, it should be allowed.
        
        This patch removes the dialed_interfaces datastore when a call is bridged (a
        suggestion from the brilliant mmichelson). If a call is being bridged, it
        should be safe to assume that we aren't stuck in a loop.
        
        Since we are now handling this is the bridge code, the previous attempts at
        handling it in app_dial and app_queue are removed.
        
        Review: https://reviewboard.asterisk.org/r/1195/
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  r315674 | twilson | 2011-04-27 03:04:10 +0400 (Wed, 27 Apr 2011) | 4 lines
  
  Make sure to create the caps structure for autocreated peers
  
  Because crashing is bad.
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  r315675 | twilson | 2011-04-27 03:10:58 +0400 (Wed, 27 Apr 2011) | 32 lines
  
  Merged revisions 315673 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r315673 | twilson | 2011-04-26 15:56:19 -0700 (Tue, 26 Apr 2011) | 25 lines
    
    Merged revisions 315672 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.6.2
    
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      r315672 | twilson | 2011-04-26 15:52:25 -0700 (Tue, 26 Apr 2011) | 18 lines
      
      Merged revisions 315671 via svnmerge from 
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
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        r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) | 11 lines
        
        Make sure unregistering a peer unlinks it from the peer container
        
        Instead of mostly copying the code from expire_register, just use the function
        that "does the right thing".
        
        (closes issue #16033)
        Reported by: kkm
        Patches: 
              016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
        Tested by: kkm, tilghman, twilson
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  r315766 | lmadsen | 2011-04-27 16:37:13 +0400 (Wed, 27 Apr 2011) | 12 lines
  
  Merged revisions 315765 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r315765 | lmadsen | 2011-04-27 07:36:17 -0500 (Wed, 27 Apr 2011) | 4 lines
    
    Enable Russian core sound selection in menuselect.
    
    (closes issue #18724)
    Reported by: pbxware
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  r315811 | russell | 2011-04-27 19:56:44 +0400 (Wed, 27 Apr 2011) | 9 lines
  
  Merged revisions 315810 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r315810 | russell | 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011) | 2 lines
    
    Set the copyright year to 2011 in the startup message.
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  r315855 | dvossel | 2011-04-27 21:44:02 +0400 (Wed, 27 Apr 2011) | 2 lines
  
  Clears exception flag during ast_read when func_jitterbuffer is enabled
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  r315856 | dvossel | 2011-04-27 21:51:53 +0400 (Wed, 27 Apr 2011) | 2 lines
  
  Makes the new ConfBridge join and leave sounds be used by default rather than beep and beeperr.
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  r315895 | mnicholson | 2011-04-27 23:15:49 +0400 (Wed, 27 Apr 2011) | 35 lines
  
  Merged revisions 315894 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r315894 | mnicholson | 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) | 28 lines
    
    Merged revisions 315893 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.6.2
    
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      r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines
      
      Merged revisions 315891 via svnmerge from 
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
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        r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines
        
        Fix our compliance with RFC 3261 section 18.2.2.
        
        This change optimizes the free_via() function and removes some redundant null
        checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
        the port specified in the Via header for routing responses (even when maddr is
        not set). Also the htons() function is now used when setting the port.
        Additional documentation comments have been added in various places to make the
        logic in the code clearer.
        
        (closes issue #18951)
        Reported by: jmls
        Patches:
              issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)
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  r316054 | pabelanger | 2011-05-02 19:58:27 +0400 (Mon, 02 May 2011) | 2 lines
  
  Formatting change, remove red blobs
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  r316095 | tilghman | 2011-05-02 23:15:46 +0400 (Mon, 02 May 2011) | 22 lines
  
  Merged revisions 316094 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r316094 | tilghman | 2011-05-02 14:09:55 -0500 (Mon, 02 May 2011) | 15 lines
    
    Merged revisions 316093 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.6.2
    
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      r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011) | 8 lines
      
      More possible crashes based upon invalid inputs.
      
      (closes issue #18161)
       Reported by: wdoekes
       Patches: 
             20110301__issue18161.diff.txt uploaded by tilghman (license 14)
       Tested by: wdoekes
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  r316213 | seanbright | 2011-05-03 22:23:03 +0400 (Tue, 03 May 2011) | 15 lines
  
  Merged revisions 316206 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r316206 | seanbright | 2011-05-03 14:17:36 -0400 (Tue, 03 May 2011) | 8 lines
    
    If we aren't interested in events, don't generate the FullyBooted event on AMI login.
    
    (closes issue #19089)
    Reported by: bklang
    Patches:
          issue19089-1.8-r316204.patch uploaded by seanbright (license 71)
    Tested by: seanbright
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Modified:
    team/may/ooh323_ipv6_direct_rtp/   (props changed)

Propchange: team/may/ooh323_ipv6_direct_rtp/
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