Merged revisions 183117 via svnmerge from

Merged revisions 183117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines

  Merged revisions 183115 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.4

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    r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines

    Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use."

    A user was having an issue where if an outgoing SIP call was canceled, the SIP device
    would remain in use if we had not received any response to the initial INVITE we sent out.
    The SIP device would remain in use until the autocongestion timer was exhausted.

    I tracked down the cause of this to be the section of code I am removing here. I asked several
    people what the purpose of this code was meant to be, but no one could give me any sort of
    answer as to why this was here. The person who was having this issue has been using this patch
    for several months and it has stopped the problems they have had.

    AST-196
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@183121 65c4cc65-6c06-0410-ace0-fbb531ad65f3