--- 1.6.0.9/chan_sip.c	2009/04/07 07:14:33	13
+++ 1.6.0.9/chan_sip.c	2009/04/07 08:49:16	15
@@ -995,11 +995,12 @@
 #define SIP_PAGE2_BUGGY_MWI		(1 << 26)	/*!< DP: Buggy CISCO MWI fix */
 #define SIP_PAGE2_REGISTERTRYING        (1 << 29)       /*!< DP: Send 100 Trying on REGISTER attempts */
 #define SIP_PAGE2_UDPTL_DESTINATION     (1 << 30)       /*!< DP: Use source IP of RTP as destination if NAT is enabled */
+#define SIP_PAGE2_Q850_REASON           (1 << 31)       /*!< DP: Get/send cause code via Reason header */
 
 #define SIP_PAGE2_FLAGS_TO_COPY \
 	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
 	SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
-	SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION)
+	SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_Q850_REASON)
 
 /*@}*/ 
 
@@ -1729,12 +1730,12 @@
 static int sipsock_read(int *id, int fd, short events, void *ignore);
 static int __sip_xmit(struct sip_pvt *p, char *data, int len);
 static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod);
-static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
+static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int hangupcause);
 static int retrans_pkt(const void *data);
 static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req);
 static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
 static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
-static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
+static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req, int hangupcause);
 static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
 static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp);
 static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
@@ -4986,9 +4987,9 @@
 			} else {	/* Incoming call, not up */
 				const char *res;
 				if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
-					transmit_response_reliable(p, res, &p->initreq);
+					transmit_response_reliable(p, res, &p->initreq, p->hangupcause);
 				else 
-					transmit_response_reliable(p, "603 Declined", &p->initreq);
+					transmit_response_reliable(p, "603 Declined", &p->initreq, p->hangupcause);
 				p->invitestate = INV_TERMINATED;
 			}
 		} else {	/* Call is in UP state, send BYE */
@@ -5324,7 +5325,7 @@
 		break;
 	case AST_CONTROL_BUSY:
 		if (ast->_state != AST_STATE_UP) {
-			transmit_response_reliable(p, "486 Busy Here", &p->initreq);
+			transmit_response_reliable(p, "486 Busy Here", &p->initreq, 0);
 			p->invitestate = INV_COMPLETED;
 			sip_alreadygone(p);
 			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
@@ -5334,7 +5335,7 @@
 		break;
 	case AST_CONTROL_CONGESTION:
 		if (ast->_state != AST_STATE_UP) {
-			transmit_response_reliable(p, "503 Service Unavailable", &p->initreq);
+			transmit_response_reliable(p, "503 Service Unavailable", &p->initreq, 0);
 			p->invitestate = INV_COMPLETED;
 			sip_alreadygone(p);
 			ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
@@ -5404,7 +5405,7 @@
 				if (p->t38.state == T38_PEER_REINVITE) {
 					AST_SCHED_DEL(sched, p->t38id);
 					change_t38_state(p, T38_DISABLED);
-					transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
+					transmit_response_reliable(p, "488 Not acceptable here", &p->initreq, 0);
 				} else if (p->t38.state == T38_ENABLED)
 					transmit_reinvite_with_sdp(p, FALSE, FALSE);
 				break;
@@ -7811,7 +7812,7 @@
 }
 
 /*! \brief Base transmit response function */
-static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
+static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int hangupcause)
 {
 	struct sip_request resp;
 	int seqno = 0;
@@ -7824,13 +7825,26 @@
 	add_header_contentLength(&resp, 0);
 	/* If we are cancelling an incoming invite for some reason, add information
 		about the reason why we are doing this in clear text */
-	if (p->method == SIP_INVITE && msg[0] != '1' && p->owner && p->owner->hangupcause) {
+	if (p->method == SIP_INVITE && msg[0] != '1') {
 		char buf[10];
+		char buf2[20];
 
-		add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
-		snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause);
+		if (!hangupcause) {
+			int respcode;
+			if (sscanf(msg, "%d ", &respcode))
+				hangupcause = hangup_sip2cause(respcode);
+		}
+
+		add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(hangupcause));
+		snprintf(buf, sizeof(buf), "%d", hangupcause);
 		add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
+
+		if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON)) {
+			sprintf(buf2, "Q.850;cause=%i", hangupcause);
+			add_header(&resp, "Reason", buf2);
+		}
 	}
+
 	return send_response(p, &resp, reliable, seqno);
 }
 
@@ -7895,7 +7909,7 @@
 	copy_socket_data(&p->socket, &req->socket);
 
 	/* Use this temporary pvt structure to send the message */
-	__transmit_response(p, msg, req, XMIT_UNRELIABLE);
+	__transmit_response(p, msg, req, XMIT_UNRELIABLE, 0);
 
 	/* Free the string fields, but not the pool space */
 	ast_string_field_init(p, 0);
@@ -7906,7 +7920,7 @@
 /*! \brief Transmit response, no retransmits */
 static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req) 
 {
-	return __transmit_response(p, msg, req, XMIT_UNRELIABLE);
+	return __transmit_response(p, msg, req, XMIT_UNRELIABLE, 0);
 }
 
 /*! \brief Transmit response, no retransmits */
@@ -7940,9 +7954,9 @@
 /*! \brief Transmit response, Make sure you get an ACK
 	This is only used for responses to INVITEs, where we need to make sure we get an ACK
 */
-static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req)
+static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req, int hangupcause)
 {
-	return __transmit_response(p, msg, req, req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL);
+	return __transmit_response(p, msg, req, req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL, hangupcause);
 }
 
 /*! \brief Append date to SIP message */
@@ -9911,6 +9925,11 @@
 		add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->hangupcause));
 		snprintf(buf, sizeof(buf), "%d", p->hangupcause);
 		add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
+
+		if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON)) {
+			sprintf(buf, "Q.850;cause=%i", p->hangupcause);
+			add_header(&resp, "Reason", buf);
+		}
 	}
 
 	add_header_contentLength(&resp, 0);
@@ -13143,6 +13162,7 @@
 		ast_cli(fd, "  Sess-Refresh : %s\n", strefresher2str(peer->stimer.st_ref));
 		ast_cli(fd, "  Sess-Expires : %d secs\n", peer->stimer.st_max_se);
 		ast_cli(fd, "  Min-Sess     : %d secs\n", peer->stimer.st_min_se);
+		ast_cli(fd, "  Use Reason   : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON) ? "Yes" : "No");
 		ast_cli(fd, "\n");
 		unref_peer(peer);
 	} else  if (peer && type == 1) { /* manager listing */
@@ -13219,6 +13239,7 @@
  		astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
  		astman_append(s, "Reg-Contact : %s\r\n", peer->fullcontact);
 		astman_append(s, "QualifyFreq : %d ms\n", peer->qualifyfreq);
+		astman_append(s, "SIP-Use-Reason-Header : %s\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)) ? "Y" : "N");
 		if (peer->chanvars) {
 			for (v = peer->chanvars ; v ; v = v->next) {
  				astman_append(s, "ChanVariable:\n");
@@ -15652,9 +15673,22 @@
 
 	sipmethod = find_sip_method(msg);
 
-	owner = p->owner;
-	if (owner) 
-		owner->hangupcause = hangup_sip2cause(resp);
+	owner = p->owner; 
+	if (owner) {
+		const char *rp = NULL, *rh = NULL;
+		if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON) && (rh = get_header(req, "Reason"))) {
+			rh = ast_skip_blanks(rh);
+			if (!strncasecmp(rh, "Q.850", 5)) {
+				rp = strstr(rh, "cause=");
+				rp += 6;
+				owner->hangupcause = atoi(rp);
+				if (req->debug)
+					ast_verbose("Using Reason header for cause code:%i\n", owner->hangupcause);
+			}
+		} else {
+			owner->hangupcause = hangup_sip2cause(resp);
+		}
+	}
 
 	/* Acknowledge whatever it is destined for */
 	if ((resp >= 100) && (resp <= 199))
@@ -16554,7 +16588,7 @@
 	if (!c) {
 		/* What to do if no channel ??? */
 		ast_log(LOG_ERROR, "Unable to create new channel.  Invite/replace failed.\n");
-		transmit_response_reliable(p, "503 Service Unavailable", req);
+		transmit_response_reliable(p, "503 Service Unavailable", req, 0);
 		append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		sip_pvt_unlock(p->refer->refer_call);
@@ -16951,7 +16985,7 @@
 	struct sip_pvt *p = (struct sip_pvt *) data;
 
 	change_t38_state(p, T38_DISABLED);
-	transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
+	transmit_response_reliable(p, "488 Not acceptable here", &p->initreq, 0);
 	p->t38id = -1;
 	return 0;
 }
@@ -17060,7 +17094,7 @@
 	
 	if (!req->ignore && p->pendinginvite) {
 		/* We already have a pending invite. Sorry. You are on hold. */
-		transmit_response_reliable(p, "491 Request Pending", req);
+		transmit_response_reliable(p, "491 Request Pending", req, 0);
 		ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
 		/* Don't destroy dialog here */
 		return 0;
@@ -17077,7 +17111,7 @@
 
 		if (p->owner) {
 			ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
-			transmit_response_reliable(p, "400 Bad request", req);	/* The best way to not not accept the transfer */
+			transmit_response_reliable(p, "400 Bad request", req, 0);	/* The best way to not not accept the transfer */
 			/* Do not destroy existing call */
 			return -1;
 		}
@@ -17089,7 +17123,7 @@
 		ast_uri_decode(replace_id);
 
 		if (!p->refer && !sip_refer_allocate(p)) {
-			transmit_response_reliable(p, "500 Server Internal Error", req);
+			transmit_response_reliable(p, "500 Server Internal Error", req, 0);
 			append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 			p->invitestate = INV_COMPLETED;
@@ -17127,7 +17161,7 @@
 		*/
 		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_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
+			transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req, 0);
 			error = 1;
 		}
 
@@ -17140,7 +17174,7 @@
 		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 = dialog_unref(p->refer->refer_call);
-			transmit_response_reliable(p, "400 Bad request", req);	/* The best way to not not accept the transfer */
+			transmit_response_reliable(p, "400 Bad request", req, 0);	/* The best way to not not accept the transfer */
 			error = 1;
 		}
 
@@ -17148,13 +17182,13 @@
 			/* 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_reliable(p, "481 Call Leg Does Not Exist (Replace)", req);
+			transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req, 0);
 			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_reliable(p, "603 Declined (Replaces)", req);
+			transmit_response_reliable(p, "603 Declined (Replaces)", req, 0);
 			error = 1;
 		}
 
@@ -17199,7 +17233,7 @@
 			/* Handle SDP here if we already have an owner */
 			if (find_sdp(req)) {
 				if (process_sdp(p, req, SDP_T38_INITIATE)) {
-					transmit_response_reliable(p, "488 Not acceptable here", req);
+					transmit_response_reliable(p, "488 Not acceptable here", req, 0);
 					if (!p->lastinvite)
 						sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 					return -1;
@@ -17229,7 +17263,7 @@
 				transmit_fake_auth_response(p, SIP_INVITE, req, XMIT_RELIABLE);
 			} else {
 				ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From"));
-				transmit_response_reliable(p, "403 Forbidden", req);
+				transmit_response_reliable(p, "403 Forbidden", req, 0);
 			}
 			p->invitestate = INV_COMPLETED;	
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -17241,7 +17275,7 @@
 		if (find_sdp(req)) {
 			if (process_sdp(p, req, SDP_T38_INITIATE)) {
 				/* Unacceptable codecs */
-				transmit_response_reliable(p, "488 Not acceptable here", req);
+				transmit_response_reliable(p, "488 Not acceptable here", req, 0);
 				p->invitestate = INV_COMPLETED;	
 				sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 				ast_debug(1, "No compatible codecs for this SIP call.\n");
@@ -17268,7 +17302,7 @@
 		if ((res = update_call_counter(p, INC_CALL_LIMIT))) {
 			if (res < 0) {
 				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);
+				transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req, 0);
 				sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 				p->invitestate = INV_COMPLETED;	
 			}
@@ -17286,11 +17320,11 @@
 
 		if (!replace_id && gotdest) {	/* No matching extension found */
 			if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP))
-				transmit_response_reliable(p, "484 Address Incomplete", req);
+				transmit_response_reliable(p, "484 Address Incomplete", req, 0);
 			else {
 				char *decoded_exten = ast_strdupa(p->exten);
 				
-				transmit_response_reliable(p, "404 Not Found", req);
+				transmit_response_reliable(p, "404 Not Found", req, 0);
 				ast_uri_decode(decoded_exten);
 				ast_log(LOG_NOTICE, "Call from '%s' to extension"
 					" '%s' rejected because extension not found.\n",
@@ -17348,7 +17382,7 @@
 		if (!ast_strlen_zero(p_uac_se_hdr)) {
 			rtn = parse_session_expires(p_uac_se_hdr, &uac_max_se, &st_ref);
 			if (rtn != 0) {
-				transmit_response_reliable(p, "400 Session-Expires Invalid Syntax", req);
+				transmit_response_reliable(p, "400 Session-Expires Invalid Syntax", req, 0);
 				p->invitestate = INV_COMPLETED;
 				if (!p->lastinvite) {
 					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -17362,7 +17396,7 @@
 		if (!ast_strlen_zero(p_uac_min_se)) {
 			rtn = parse_minse(p_uac_min_se, &uac_min_se); 
 			if (rtn != 0) {
-        			transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
+        			transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req, 0);
        	   			p->invitestate = INV_COMPLETED;
        	   			if (!p->lastinvite) {
 					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -17490,12 +17524,12 @@
 				case AST_PBX_FAILED:
 					ast_log(LOG_WARNING, "Failed to start PBX :(\n");
 					p->invitestate = INV_COMPLETED;
-					transmit_response_reliable(p, "503 Unavailable", req);
+					transmit_response_reliable(p, "503 Unavailable", req, 0);
 					break;
 				case AST_PBX_CALL_LIMIT:
 					ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
 					p->invitestate = INV_COMPLETED;
-					transmit_response_reliable(p, "480 Temporarily Unavailable", req);
+					transmit_response_reliable(p, "480 Temporarily Unavailable", req, 0);
 					break;
 				case AST_PBX_SUCCESS:
 					/* nothing to do */
@@ -17516,7 +17550,7 @@
 				*nounlock = 1;
 				if (ast_pickup_call(c)) {
 					ast_log(LOG_NOTICE, "Nothing to pick up for %s\n", p->callid);
-					transmit_response_reliable(p, "503 Unavailable", req);
+					transmit_response_reliable(p, "503 Unavailable", req, 0);
 					sip_alreadygone(p);
 					/* Unlock locks so ast_hangup can do its magic */
 					sip_pvt_unlock(p);
@@ -17571,7 +17605,7 @@
 				ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
 				msg = "503 Unavailable";
 			}
-			transmit_response_reliable(p, msg, req);
+			transmit_response_reliable(p, msg, req, 0);
 			p->invitestate = INV_COMPLETED;
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		}
@@ -18059,7 +18093,7 @@
 	else
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 	if (p->initreq.len > 0) {
-		transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
+		transmit_response_reliable(p, "487 Request Terminated", &p->initreq, 0);
 		transmit_response(p, "200 OK", req);
 		return 1;
 	} else {
@@ -18159,7 +18193,7 @@
 	
 	/* If we have an INCOMING invite that we haven't answered, terminate that transaction */
 	if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !req->ignore) {
-		transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
+		transmit_response_reliable(p, "487 Request Terminated", &p->initreq, 0);
 	}
 
 	__sip_pretend_ack(p);
@@ -18348,7 +18382,7 @@
 			transmit_fake_auth_response(p, SIP_SUBSCRIBE, req, XMIT_UNRELIABLE);
 		} else {
 			ast_log(LOG_NOTICE, "Failed to authenticate user %s for SUBSCRIBE\n", get_header(req, "From"));
-			transmit_response_reliable(p, "403 Forbidden", req);
+			transmit_response_reliable(p, "403 Forbidden", req, 0);
 		}
 		p->needdestroy = 1;
 		return 0;
@@ -18748,7 +18782,7 @@
 		if (!p->initreq.headers && req->has_to_tag) {
 			/* If this is a first request and it got a to-tag, it is not for us */
 			if (!req->ignore && req->method == SIP_INVITE) {
-				transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
+				transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req, 0);
 				/* Will cease to exist after ACK */
 			} else if (req->method != SIP_ACK) {
 				transmit_response(p, "481 Call/Transaction Does Not Exist", req);
@@ -21096,7 +21130,9 @@
 			} else {
 				peer->stimer.st_ref = i;
 			}
-		}
+		} else if (!strcasecmp(v->name, "use_q850_reason")) {
+			ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_Q850_REASON);
+ 		}
 	}
 
 	if (!peer->socket.type) {
@@ -22342,7 +22378,7 @@
 	}
 
 	ast_string_field_build(p, our_contact, "Transfer <sip:%s@%s%s%s>", extension, host, port ? ":" : "", port ? port : "");
-	transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq);
+	transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq, 0);
 
 	sip_scheddestroy(p, SIP_TRANS_TIMEOUT);	/* Make sure we stop send this reply. */
 	sip_alreadygone(p);