diff options
| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-07-24 18:32:25 +0000 |
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| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-07-24 18:32:25 +0000 |
| commit | 3c71404a7e38442055391d5561d43bca487ccf4d (patch) | |
| tree | 47ee780bcade1f7a17d3bad860cb01c23d77d9ae /channels | |
| parent | 1cc91efc731654e3a71151fc53f2598bcda5f1e6 (diff) | |
Merged revisions 208588 via svnmerge from
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: http://svn.digium.com/svn/asterisk/branches/1.6.1@208590 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 4 |
1 files changed, 3 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index d5dabe19f..2ac7045cf 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) { |