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Merged revisions 208588 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/trunk

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  r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16 lines
  
  Merged revisions 208587 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines
    
    Only send a BYE when hanging up a channel that is up.
    
    For cases where Asterisk sends an INVITE and receives a non 2XX final
    response, Asterisk would follow the INVITE transaction by immediately
    sending a BYE, which was unnecessary.
    
    (closes issue #14575)
    Reported by: chris-mac
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2009-07-24 18:32:50 +00:00
parent b7e75a4a56
commit 17fcb39e1a
1 changed files with 3 additions and 1 deletions
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channels/chan_sip.c
+3 -1
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@@ -5787,7 +5787,9 @@ static int sip_hangup(struct ast_channel *ast)
if (p->trtp)
textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY);
/* Send a hangup */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
if (oldowner->_state == AST_STATE_UP) {
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
}
/* Get RTCP quality before end of call */
if (p->do_history) {