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Don't reinvite if we got a refer 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer 2003-11-12 00:13:55 +00:00
parent 5865f43a79
commit a2303e3d0f
1 changed files with 7 additions and 2 deletions
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9
channels/chan_sip.c
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@ -240,6 +240,7 @@ static struct sip_pvt {
int amaflags; /* AMA Flags */
int pendinginvite; /* Any pending invite */
int pendingbye; /* Need to send bye after we ack? */
int gotrefer; /* Got a refer? */
struct sip_request initreq; /* Initial request */
int maxtime; /* Max time for first response */
@ -5123,6 +5124,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
attempt_transfer(p, p->refer_call);
ast_mutex_unlock(&p->refer_call->lock);
p->refer_call = NULL;
p->gotrefer = 1;
} else {
ast_log(LOG_DEBUG,"202 Accepted (blind)\n");
c = p->owner;
@ -5133,6 +5135,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
ast_async_goto(transfer_to,p->context, p->refer_to,1, 1);
}
}
p->gotrefer = 1;
}
/* Always increment on a BYE */
transmit_request_with_auth(p, "BYE", 0, 1);
@ -6231,8 +6234,10 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
ast_rtp_get_peer(vrtp, &p->vredirip);
else
memset(&p->vredirip, 0, sizeof(p->vredirip));
transmit_reinvite_with_sdp(p, rtp, vrtp);
p->outgoing = 1;
if (!p->gotrefer) {
transmit_reinvite_with_sdp(p, rtp, vrtp);
p->outgoing = 1;
}
return 0;
}
return -1;