It's critical that we get an ACK on a 200 OK to an INVITE. If we do not get the ACK,
authorOlle Johansson <oej@edvina.net>
Tue, 18 Apr 2006 07:03:36 +0000 (07:03 +0000)
committerOlle Johansson <oej@edvina.net>
Tue, 18 Apr 2006 07:03:36 +0000 (07:03 +0000)
tear down the call. (Discovered at SIPit18)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21061 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 525fd54..8414178 100644 (file)
@@ -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);