From: Mark Michelson <mmichelson@digium.com>
Date: Fri, 24 Jul 2009 18:32:25 +0000 (+0000)
Subject: Merged revisions 208588 via svnmerge from
X-Git-Tag: 1.6.1.3-rc1~19
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=623e055a283e95d76a2da6c0bbe338243c45e217;p=thirdparty%2Fasterisk.git

Merged revisions 208588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  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

  ........
    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
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d5dabe19f6..2ac7045cfc 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5360,7 +5360,9 @@ 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) {