[asterisk-commits] trunk r21061 - /trunk/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Tue Apr 18 02:03:36 2006
New Revision: 21061

URL: http://svn.digium.com/view/asterisk?rev=21061&view=rev
Log:
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)

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=21061&r1=21060&r2=21061&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Apr 18 02:03:36 2006
@@ -2790,7 +2790,7 @@
 		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 @@
 			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);