Merged revisions 189771 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
  r189771 | dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines
  
  Fixes segfault when switching UDP to TCP in sip.conf after reload.
  
  If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload.  The problem is the socket type is changed to TCP but the fd may still be present for UDP.  Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present.  Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found.
  
  (closes issue #14727)
  Reported by: pj
  Tested by: dvossel
  
  Review: http://reviewboard.digium.com/r/229/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@189775 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -3261,14 +3261,18 @@ static int __sip_xmit(struct sip_pvt *p, struct ast_str *data, int len)
 	if (p->socket.tcptls_session)
 		ast_mutex_lock(&p->socket.tcptls_session->lock);
 
-	if (p->socket.type & SIP_TRANSPORT_UDP) 
+	if (p->socket.type & SIP_TRANSPORT_UDP) {
 		res = sendto(p->socket.fd, data->str, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
-	else {
-		if (p->socket.tcptls_session->f) 
+	} else if (p->socket.tcptls_session) {
+		if (p->socket.tcptls_session->f) {
 			res = ast_tcptls_server_write(p->socket.tcptls_session, data->str, len);
-		else
+		} else {
 			ast_debug(2, "No p->socket.tcptls_session->f len=%d\n", len);
-	} 
+		}
+	} else {
+		ast_debug(2, "Socket type is TCP but no tcptls_session is present to write to\n");
+		return XMIT_ERROR;
+	}
 
 	if (p->socket.tcptls_session)
 		ast_mutex_unlock(&p->socket.tcptls_session->lock);
@@ -22583,8 +22587,10 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
 				else
 					ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
 
-				if (!peer->socket.type) /*!< The first transport listed should be used for outgoing */
+				if (!peer->socket.type) { /*!< The first transport listed should be used for outgoing */
 					peer->socket.type = peer->transports;
+					peer->socket.fd = -1;
+				}
 			}
 		} else if (realtime && !strcasecmp(v->name, "regseconds")) {
 			ast_get_time_t(v->value, &regseconds, 0, NULL);
@@ -22878,9 +22884,10 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
 
 	if (!peer->socket.type) {
 		/* Set default set of transports */
-		peer->transports  = default_transports;
+		peer->transports = default_transports;
 		/* Set default primary transport */
 		peer->socket.type = default_primary_transport;
+		peer->socket.fd = -1;
 	}
 
 	if (fullcontact->used > 0) {