diff options
| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-07-24 18:26:50 +0000 |
|---|---|---|
| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-07-24 18:26:50 +0000 |
| commit | f7111173898045b29c6a290d724bd9545f91f8ff (patch) | |
| tree | 11b87d7ffc187d73cf0ec7cb9f77476b0787ea86 | |
| parent | a39611a6f1fd3395c29d0bec30b50424d5bdf5f7 (diff) | |
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: http://svn.digium.com/svn/asterisk/branches/1.4@208587 f38db490-d61c-443f-a65b-d21fe96a405b
| -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 890904908..52f4708fd 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -3726,7 +3726,9 @@ static int sip_hangup(struct ast_channel *ast) if (p->vrtp) videoqos = ast_rtp_get_quality(p->vrtp, NULL); /* 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 (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) { |