[asterisk-commits] oej: branch 1.4 r50124 - /branches/1.4/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Tue Jan  9 05:25:20 2007
New Revision: 50124

URL: http://svn.digium.com/view/asterisk?view=rev&rev=50124
Log:
- handle re-invites properly in sip_hangup()
- Add some invitestate status changes just to be sure

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=50124&r1=50123&r2=50124
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Jan  9 05:25:20 2007
@@ -3285,7 +3285,7 @@
 		return 0;
 	}
 	/* If the call is not UP, we need to send CANCEL instead of BYE */
-	if (ast->_state == AST_STATE_RING || ast->_state == AST_STATE_RINGING || p->invitestate < INV_COMPLETED) {
+	if (ast->_state == AST_STATE_RING || ast->_state == AST_STATE_RINGING || (p->invitestate < INV_COMPLETED && ast->_state != AST_STATE_UP)) {
 		needcancel = TRUE;
 		if (option_debug > 3)
 			ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
@@ -3643,7 +3643,7 @@
 	case AST_CONTROL_BUSY:
 		if (ast->_state != AST_STATE_UP) {
 			transmit_response(p, "486 Busy Here", &p->initreq);
-			p->invitestate = INV_TERMINATED;
+			p->invitestate = INV_COMPLETED;
 			sip_alreadygone(p);
 			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
 			break;
@@ -3653,7 +3653,7 @@
 	case AST_CONTROL_CONGESTION:
 		if (ast->_state != AST_STATE_UP) {
 			transmit_response(p, "503 Service Unavailable", &p->initreq);
-			p->invitestate = INV_TERMINATED;
+			p->invitestate = INV_COMPLETED;
 			sip_alreadygone(p);
 			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
 			break;
@@ -13059,6 +13059,7 @@
 			/* At this point we only support REPLACES */
 			transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, required);
 			ast_log(LOG_WARNING,"Received SIP INVITE with unsupported required extension: %s\n", required);
+			p->invitestate = INV_COMPLETED;
 			if (!p->lastinvite)
 				sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 			return -1;
@@ -13073,6 +13074,7 @@
 		/* 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);
+		p->invitestate = INV_COMPLETED;
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		return 0;
 	}
@@ -13113,6 +13115,7 @@
 			transmit_response(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;
 			return -1;
 		}
 
@@ -13186,6 +13189,7 @@
 				ast_mutex_unlock(&p->refer->refer_call->lock);
 				ast_channel_unlock(p->refer->refer_call->owner);
 			}
+			p->invitestate = INV_COMPLETED;
 			return -1;
 		}
 	}
@@ -13236,6 +13240,7 @@
 		/* Handle authentication if this is our first invite */
 		res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin);
 		if (res == AUTH_CHALLENGE_SENT)
+			p->invitestate = INV_COMPLETED;		/* Needs to restart in another INVITE transaction */
 			return 0; 
 		if (res < 0) { /* Something failed in authentication */
 			if (res == AUTH_FAKE_AUTH) {
@@ -13245,6 +13250,7 @@
   				ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From"));
 				transmit_response_reliable(p, "403 Forbidden", req);
   			}
+			p->invitestate = INV_COMPLETED;	
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 			ast_string_field_free(p, theirtag);
 			return 0;
@@ -13255,6 +13261,7 @@
 			if (process_sdp(p, req)) {
 				/* Unacceptable codecs */
 				transmit_response_reliable(p, "488 Not acceptable here", req);
+				p->invitestate = INV_COMPLETED;	
 				sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 				if (option_debug)
 					ast_log(LOG_DEBUG, "No compatible codecs for this SIP call.\n");
@@ -13285,6 +13292,7 @@
 				ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username);
 				transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
 				sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+				p->invitestate = INV_COMPLETED;	
 			}
 			return 0;
 		}
@@ -13303,6 +13311,7 @@
 				transmit_response_reliable(p, "484 Address Incomplete", req);
 			else
 				transmit_response_reliable(p, "404 Not Found", req);
+			p->invitestate = INV_COMPLETED;	
 			update_call_counter(p, DEC_CALL_LIMIT);
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 			return 0;
@@ -13503,6 +13512,7 @@
 						/* No bridged peer with T38 enabled*/
 					}
 				} 
+				/* Respond to normal re-invite */
 				if (sendok)
 					transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
 
@@ -14590,7 +14600,7 @@
 	case SIP_ACK:
 		/* Make sure we don't ignore this */
 		if (seqno == p->pendinginvite) {
-			p->invitestate = INV_CONFIRMED;
+			p->invitestate = INV_TERMINATED;
 			p->pendinginvite = 0;
 			__sip_ack(p, seqno, FLAG_RESPONSE, 0);
 			if (find_sdp(req)) {