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  r284666 | tilghman | 2010-09-02 11:11:15 -0500 (Thu, 02 Sep 2010) | 9 lines
  
  Merged revisions 284665 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02 Sep 2010) | 2 lines
    
    Fixing build.
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  r284632 | tilghman | 2010-09-02 00:31:02 -0500 (Thu, 02 Sep 2010) | 14 lines
  
  Merged revisions 284631 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) | 7 lines
    
    Don't reset queue stats on a module reload.
    
    (closes issue #17535)
     Reported by: raarts
     Patches: 
           20100819__issue17535.diff.txt uploaded by tilghman (license 14)
  ........
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  r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines
  
  When optional_api is non-optional, force dependent modules to be loaded.
  
  (closes issue #17707)
   Reported by: ira
   Patches: 
         20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14)
   Tested by: tilghman
   
  Review: https://reviewboard.asterisk.org/r/876/
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  r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines
  
  Merged revisions 284593,284595 via svnmerge from 
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  ................
    r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines
    
    Merged revisions 284478 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines
      
      Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
      
      This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing
      a potential crash bug in all supported releases.
      
      (closes issue #17678)
       Reported by: russell
      Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select 
      
      Review: https://reviewboard.asterisk.org/r/824/
    ........
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    r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines
    
    Failed to rerun bootstrap.sh after last commit
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  r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) | 9 lines
  
  During request to dialog matching, verify init_ruri is present before comparing.
  
  During request to dialog matching, we attempt a best effort routine for fork
  detection which requires several elements to be in place.  The dialog's
  initial request uri is one of those elements.  Since it is best effort,
  if the init_ruri is not present for some reason we can not proceed with that
  routine.
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  r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines
  
  Fix SRTP for changing SSRC and multiple a=crypto SDP lines
  
  Adding code to Asterisk that changed the SSRC during bridges and masquerades
  broke SRTP functionality. Also broken was handling the situation where an
  incoming INVITE had more than one crypto offer. This patch caches the SRTP
  policies the we use so that we can change the ssrc and inform libsrtp of the
  new streams. It also uses the first acceptable a=crypto line from the incoming
  INVITE.
  
  (closes issue #17563)
  Reported by: Alexcr
  Patches: 
        srtp.diff uploaded by twilson (license 396)
  Tested by: twilson
  
  Review: https://reviewboard.asterisk.org/r/878/
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  r284473 | tilghman | 2010-09-01 13:16:37 -0500 (Wed, 01 Sep 2010) | 12 lines
  
  Merged revisions 284472 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01 Sep 2010) | 5 lines
    
    Don't warn on floats and timestamps
    
    (closes issue #17082)
    Reported by: coolmig
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  r284415 | tilghman | 2010-08-31 15:22:10 -0500 (Tue, 31 Aug 2010) | 21 lines
  
  Merged revisions 284399 via svnmerge from 
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  ................
    r284399 | tilghman | 2010-08-31 15:18:32 -0500 (Tue, 31 Aug 2010) | 14 lines
    
    Merged revisions 284393 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines
      
      Don't send a devstate change on poke_noanswer if the state did not change.
      
      (closes issue #17741)
       Reported by: schmidts
       Patches: 
             chan_sip.c.patch uploaded by schmidts (license 1077)
    ........
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  r284318 | lmadsen | 2010-08-31 14:00:15 -0500 (Tue, 31 Aug 2010) | 22 lines
  
  Merged revisions 284317 via svnmerge from 
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  ................
    r284317 | lmadsen | 2010-08-31 13:59:31 -0500 (Tue, 31 Aug 2010) | 15 lines
    
    Merged revisions 284316 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines
      
      Update say.conf.sample to match the rules in say.c
      
      (closes issue #17835)
      Reported by: RoadKill
      Patches:
            say.conf.sample.patch.rules uploaded by RoadKill (license 933)
      Tested by: RoadKill
    ........
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(closes issue #17860)
Reported by: jtodd
Patches:
      __20100816-chan_sip-sip-show-settings.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, russell

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  r284281 | tilghman | 2010-08-30 17:28:47 -0500 (Mon, 30 Aug 2010) | 18 lines
  
  Merged revisions 284280 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) | 11 lines
    
    Fix 3 coding errors:
      1) After we close FD, we should not be trying to write to it.
      2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child.
      3) Use endian, not processor, detection to ensure bytes are written in the correct order.
    
    (closes issue #15706)
     Reported by: modelnine
     Patches: 
           asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865)
     Tested by: gmartinez
  ........
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You can't write "same as above" in hypertext documentation. Above doesn't make sense in
hyperspace.


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  r284158 | tilghman | 2010-08-29 02:05:27 -0500 (Sun, 29 Aug 2010) | 2 lines
  
  Missed adding this file
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  r284127 | tilghman | 2010-08-29 00:17:37 -0500 (Sun, 29 Aug 2010) | 2 lines
  
  Also ignore the checksums
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  r284096 | tilghman | 2010-08-28 21:51:14 -0500 (Sat, 28 Aug 2010) | 3 lines
  
  Rename CEL adaptive driver to plain driver, since there isn't another ODBC driver
  (and the other CEL drivers have adaptive capabilities, anyway).
........


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  r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010) | 13 lines
  
  Be more flexible with whitespace on AMI action headers.
  
  Previously, this code required exactly one space to be after the ':' in headers
  for an AMI action.  This now makes whitespace optional, and allows whitespace that
  is there to vary in amount.
  
  (closes issue #17862)
  Reported by: cmoye
  Patches:
        manager.c.patch_trunk uploaded by cmoye (license 858)
        manager.c.patch_1.8 uploaded by cmoye (license 858)
  Tested by: cmoye
........


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  r284032 | dvossel | 2010-08-27 17:37:11 -0500 (Fri, 27 Aug 2010) | 21 lines
  
  Merged revisions 284002 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r284002 | dvossel | 2010-08-27 17:27:50 -0500 (Fri, 27 Aug 2010) | 14 lines
    
    Merged revisions 283960 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines
      
      Parse all "Accept" headers for SIP SUBSCRIBE requests.
      
      (closes issue #17758)
      Reported by: ibc
      Patches:
            multiple_accept_headers_1.4.diff uploaded by dvossel (license 671)
    ........
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  r283951 | russell | 2010-08-27 16:33:55 -0500 (Fri, 27 Aug 2010) | 2 lines
  
  Print exten@context:priority in verbose messages from pbx_realtime.
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  r283882 | qwell | 2010-08-27 15:31:55 -0500 (Fri, 27 Aug 2010) | 22 lines
  
  Merged revisions 283881 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r283881 | qwell | 2010-08-27 15:30:27 -0500 (Fri, 27 Aug 2010) | 15 lines
    
    Merged revisions 283880 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | 8 lines
      
      Fix issue with decoding ^-escaped characters in realtime.
      
      (closes issue #17790)
      Reported by: denzs
      Patches: 
            17790-chunky.diff uploaded by qwell (license 4)
      Tested by: qwell, denzs
    ........
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  r283770 | tilghman | 2010-08-26 18:47:02 -0500 (Thu, 26 Aug 2010) | 8 lines
  
  Convert MOH to use generic timers.
  
  (closes issue #17726)
   Reported by: lmadsen
   Patches: 
         20100825__issue17726__2.diff.txt uploaded by tilghman (license 14)
   Tested by: tilghman
........


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  r283692 | dvossel | 2010-08-26 10:26:37 -0500 (Thu, 26 Aug 2010) | 32 lines
  
  Merged revisions 283691 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r283691 | dvossel | 2010-08-26 10:24:40 -0500 (Thu, 26 Aug 2010) | 25 lines
    
    Merged revisions 283690 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines
      
      Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response.
      
      If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response
      to its outgoing INVITE, Asterisk used to pretend_ack the INVITE.  This is not rfc
      compliant and results in confusion at the other endpoint.  sip_pretend_ack will ack
      and remove all the packets in the retransmit queue.  This means that the INVITE will
      stop retransmitting, and that any response to that INVITE that comes after the pretend_ack
      occurs will be ignored.
      
      Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal
      hangup, we should let the protocol stack process the INVITE transaction and terminate
      the dialog properly.  This is achieved by setting the PENDING_BYE flag.  When this flag
      is used, once the dialog proceeds to an escapable state the transaction will either be
      canceled with a SIP_CANCEL or completed followed immediately by a BYE.  Attempting to do
      this any other way is incorrect.  If the endpoint is not responding to the INVITE request,
      the INVITE must continue to be retransmitted until it times out which will result in the
      dialog being destroyed.
    ........
  ................
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  r283659 | russell | 2010-08-26 08:26:14 -0500 (Thu, 26 Aug 2010) | 2 lines
  
  Slight improvement to a debug message.
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  r283629 | russell | 2010-08-26 07:48:45 -0500 (Thu, 26 Aug 2010) | 2 lines
  
  Remove public keys that are no longer useful.
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  r283627 | russell | 2010-08-26 07:26:22 -0500 (Thu, 26 Aug 2010) | 2 lines
  
  Move httptimeout out from in between port and bindaddr.
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  r283595 | dvossel | 2010-08-25 17:57:56 -0500 (Wed, 25 Aug 2010) | 14 lines
  
  Merged revisions 283594 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) | 7 lines
    
    Add to and from tags to NOTIFY dialog-info xml body so pickup can occur.
    
    When pedantic mode is used, the dialog-info xml generated during a
    ringing event must contain the to and from tag values.  Otherwise if
    a pickup occurs using INVITE with replaces, Astrisk will not be able
    to locate the subscription.
  ........
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  r283561 | tilghman | 2010-08-25 11:12:43 -0500 (Wed, 25 Aug 2010) | 5 lines
  
  Initialize connect timeout on each time through the loop.
  
  (closes issue #17911)
   Reported by: wurstsalat
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  r283559 | dvossel | 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) | 16 lines
  
  Merged revisions 283558 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines
    
    Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used.
    
    Asterisk now dynamically builds the "Supported" header depending
    on what is enabled/disabled in sip.conf.  Session timers used
    to always be advertised as being supported even when they were disabled
    in the configuration.  This caused problems with some end points.
    
    (issue #17005)
  ........
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  r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) | 2 lines
  
  Convert ast_log(LOG_DEBUG, ...) to ast_debug(...)
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(closes issue #17887)
Reported by: salecha
Patches:
      skinny.redial.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, salecha


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  r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) | 2 lines
  
  Changes the default behavior for sip.conf's pedantic option from "no" to "yes".
........


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  r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) | 9 lines
  
  Fix issue where TOS is no longer set on RTP packets.
  Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk.
  
  (closes issue #17890)
  Reported by: elguero
  Patches:
        qos_18.diff uploaded by elguero (license 37)
  
  Review: https://reviewboard.asterisk.org/r/868
........


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  r283382 | dvossel | 2010-08-24 11:11:18 -0500 (Tue, 24 Aug 2010) | 25 lines
  
  Merged revisions 283381 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines
    
    Merged revisions 283380 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines
      
      This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set.
      
      When the pending bye flag is used, it is possible that the dialog will terminate
      and leave the sip_pvt->owner channel up.  This is because we never hangup the
      ast_channel after sending the SIP_BYE request.  When we receive the response for
      the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the
      next do_monitor loop, but this is not the case.  The dialog will only be destroyed
      once the owner is hungup even with the need_destroy flag set.  This patch sets the
      softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the
      pending bye flag.
    ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283383 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r283350 | russell | 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010) | 2 lines
  
  Don't attempt to release a NULL ODBC handle.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283351 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r283319 | tilghman | 2010-08-23 16:33:47 -0500 (Mon, 23 Aug 2010) | 9 lines
  
  Merged revisions 283318 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010) | 2 lines
    
    CDR drivers depend upon res_odbc, not directly on the ODBC libraries
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283320 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283289 65c4cc65-6c06-0410-ace0-fbb531ad65f3