[asterisk-commits] oej: branch 1.4 r126516 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: oej
Date: Mon Jun 30 07:50:55 2008
New Revision: 126516

URL: http://svn.digium.com/view/asterisk?view=rev&rev=126516
Log:
Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and
also fail if we don't get the very same precious ACK. Based on patch by tsearle, with
my own additions.

(closes issue #12951)

Reported by: tsearle
Patches: 
      busy_retransmit.patch uploaded by tsearle (license 373)

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=126516&r1=126515&r2=126516
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Mon Jun 30 07:50:55 2008
@@ -3924,7 +3924,7 @@
 		break;
 	case AST_CONTROL_BUSY:
 		if (ast->_state != AST_STATE_UP) {
-			transmit_response(p, "486 Busy Here", &p->initreq);
+			transmit_response_reliable(p, "486 Busy Here", &p->initreq);
 			p->invitestate = INV_COMPLETED;
 			sip_alreadygone(p);
 			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
@@ -3934,7 +3934,7 @@
 		break;
 	case AST_CONTROL_CONGESTION:
 		if (ast->_state != AST_STATE_UP) {
-			transmit_response(p, "503 Service Unavailable", &p->initreq);
+			transmit_response_reliable(p, "503 Service Unavailable", &p->initreq);
 			p->invitestate = INV_COMPLETED;
 			sip_alreadygone(p);
 			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
@@ -13815,7 +13815,7 @@
 	   	being able to call yourself */
 		/* If pedantic is on, we need to check the tags. If they're different, this is
 	   	in fact a forked call through a SIP proxy somewhere. */
-		transmit_response(p, "482 Loop Detected", req);
+		transmit_response_reliable(p, "482 Loop Detected", req);
 		p->invitestate = INV_COMPLETED;
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		return 0;
@@ -13823,7 +13823,7 @@
 	
 	if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->pendinginvite) {
 		/* We already have a pending invite. Sorry. You are on hold. */
-		transmit_response(p, "491 Request Pending", req);
+		transmit_response_reliable(p, "491 Request Pending", req);
 		if (option_debug)
 			ast_log(LOG_DEBUG, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
 		/* Don't destroy dialog here */
@@ -13842,7 +13842,7 @@
 		if (p->owner) {
 			if (option_debug > 2)
 				ast_log(LOG_DEBUG, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
-			transmit_response(p, "400 Bad request", req);	/* The best way to not not accept the transfer */
+			transmit_response_reliable(p, "400 Bad request", req);	/* The best way to not not accept the transfer */
 			/* Do not destroy existing call */
 			return -1;
 		}
@@ -13854,7 +13854,7 @@
 		ast_uri_decode(replace_id);
 
 		if (!p->refer && !sip_refer_allocate(p)) {
-			transmit_response(p, "500 Server Internal Error", req);
+			transmit_response_reliable(p, "500 Server Internal Error", req);
 			append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 			p->invitestate = INV_COMPLETED;
@@ -13892,7 +13892,7 @@
 		*/
 		if ((p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
 			ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
-			transmit_response(p, "481 Call Leg Does Not Exist (Replaces)", req);
+			transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
 			error = 1;
 		}
 
@@ -13905,7 +13905,7 @@
 		if (p->refer->refer_call == p) {
 			ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid);
 			p->refer->refer_call = NULL;
-			transmit_response(p, "400 Bad request", req);	/* The best way to not not accept the transfer */
+			transmit_response_reliable(p, "400 Bad request", req);	/* The best way to not not accept the transfer */
 			error = 1;
 		}
 
@@ -13913,13 +13913,13 @@
 			/* Oops, someting wrong anyway, no owner, no call */
 			ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id);
 			/* Check for better return code */
-			transmit_response(p, "481 Call Leg Does Not Exist (Replace)", req);
+			transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req);
 			error = 1;
 		}
 
 		if (!error && p->refer->refer_call->owner->_state != AST_STATE_RINGING && p->refer->refer_call->owner->_state != AST_STATE_RING && p->refer->refer_call->owner->_state != AST_STATE_UP ) {
 			ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
-			transmit_response(p, "603 Declined (Replaces)", req);
+			transmit_response_reliable(p, "603 Declined (Replaces)", req);
 			error = 1;
 		}
 
@@ -13961,7 +13961,7 @@
 			/* Handle SDP here if we already have an owner */
 			if (find_sdp(req)) {
 				if (process_sdp(p, req)) {
-					transmit_response(p, "488 Not acceptable here", req);
+					transmit_response_reliable(p, "488 Not acceptable here", req);
 					if (!p->lastinvite)
 						sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 					return -1;