Merged revisions 283692 via svnmerge from

Merged revisions 283692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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

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    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

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      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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283693 65c4cc65-6c06-0410-ace0-fbb531ad65f3