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It's critical that we get an ACK on a 200 OK to an INVITE. If we do not get the ACK, 

tear down the call. (Discovered at SIPit18)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@21061 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej 2006-04-18 07:03:36 +00:00
parent 8da38ca62e
commit 8169417975
1 changed files with 2 additions and 2 deletions
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4
channels/chan_sip.c
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@ -2790,7 +2790,7 @@ static int sip_answer(struct ast_channel *ast)
ast_setstate(ast, AST_STATE_UP);
if (option_debug)
ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_RELIABLE);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
}
ast_mutex_unlock(&p->lock);
return res;
@ -11097,7 +11097,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
transmit_response(p, "180 Ringing", req);
break;
case AST_STATE_UP:
transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE);
transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
break;
default:
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);