[asterisk-commits] oej: trunk r46383 - in /trunk: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Sat Oct 28 14:16:23 2006
New Revision: 46383

URL: http://svn.digium.com/view/asterisk?rev=46383&view=rev
Log:
Merge from 1.4 : Don't send 183 reliably...

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=46383&r1=46382&r2=46383&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sat Oct 28 14:16:23 2006
@@ -3546,7 +3546,7 @@
 				if ((ast->_state != AST_STATE_UP) &&
 					!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) && 
 				    !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
-					transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_RELIABLE);
+					transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
 					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
 				}
 				res = ast_udptl_write(p->udptl, frame);
@@ -12899,7 +12899,7 @@
 		/* We should answer something here. If we are here, the
 			call we are replacing exists, so an accepted 
 			can't harm */
-		transmit_response_with_sdp(p, "200 OK", req, 1);
+		transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE);
 		/* Do something more clever here */
 		ast_channel_unlock(c);
 		sip_pvt_unlock(p->refer->refer_call);
@@ -12908,7 +12908,7 @@
 	if (!c) {
 		/* What to do if no channel ??? */
 		ast_log(LOG_ERROR, "Unable to create new channel.  Invite/replace failed.\n");
-		transmit_response_with_sdp(p, "503 Service Unavailable", req, 1);
+		transmit_response_reliable(p, "503 Service Unavailable", req);
 		append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		sip_pvt_unlock(p->refer->refer_call);
@@ -12933,7 +12933,7 @@
 	   Targetcall is not touched by the masq */
 
 	/* Answer the incoming call and set channel to UP state */
-	transmit_response_with_sdp(p, "200 OK", req, 1);
+	transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE);
 	ast_setstate(c, AST_STATE_UP);
 	
 	/* Stop music on hold and other generators */