Index: channels/chan_sip.c
===================================================================
--- channels/chan_sip.c	(revision 13388)
+++ channels/chan_sip.c	(working copy)
@@ -459,7 +459,9 @@
 #define DEC_CALL_LIMIT	0
 #define INC_CALL_LIMIT	1
 
+char reg_dialstring[64] = "";
 
+
 /*! \brief sip_request: The data grabbed from the UDP socket */
 struct sip_request {
 	char *rlPart1; 		/*!< SIP Method Name or "SIP/2.0" protocol version */
@@ -1661,9 +1663,11 @@
 	snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port));
 	
 	if (fullcontact)
-		ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, "port", port, "regseconds", regseconds, "username", username, "fullcontact", fullcontact, NULL);
+		ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, "port", port, "regseconds", regseconds, "username", username,
+			"reg_dialstring", reg_dialstring, "fullcontact", fullcontact, NULL);
 	else
-		ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, "port", port, "regseconds", regseconds, "username", username, NULL);
+		ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr, "port", port, "regseconds", regseconds, "username", username,
+			"reg_dialstring", reg_dialstring, NULL);
 }
 
 /*! \brief Automatically add peer extension to dial plan */
@@ -5781,7 +5785,8 @@
 {
 	if (!ast_test_flag((&global_flags_page2), SIP_PAGE2_IGNOREREGEXPIRE)) {
 		if (ast_test_flag(&(peer->flags_page2), SIP_PAGE2_RT_FROMCONTACT)) {
-			ast_update_realtime("sippeers", "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", "username", "", NULL);
+			ast_update_realtime("sippeers", "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", "username", "",
+				"reg_dialstring", "", NULL);
 		} else {
 			ast_db_del("SIP/Registry", peer->name);
 		}
@@ -12663,6 +12668,9 @@
 			}
 		} else if (!strcasecmp(v->name, "callevents")) {
 			global_callevents = ast_true(v->value);
+		} else if (!strcasecmp(v->name, "reg-dialstring")) {
+			ast_copy_string(reg_dialstring, v->value, sizeof reg_dialstring);
+			ast_verbose(VERBOSE_PREFIX_2 "Got SIP registrar dialstring \"%s\"\n", reg_dialstring);
 		}
 	}
 
Index: configs/sip.conf.sample
===================================================================
--- configs/sip.conf.sample	(revision 13388)
+++ configs/sip.conf.sample	(working copy)
@@ -112,6 +112,14 @@
 ;notifyringing = yes		; Notify subscriptions on RINGING state
 ;callevents=no			; generate manager events when sip ua performs events (e.g. hold)
 
+;reg-dialstring=SIP/myname	; If using a two-tire model with n sip/sip gateways in front with DNS
+				; roundrobin and a bunch of sip/pstn gateways in the back, the sip/pstn
+				; gateways should dial this server on incoming. This field is updated on SIP
+				; REGISTER and ensures that calls from PSTN are directed through the correct sip
+				; gateway in front to avoid NAT problems (if using the wrong sip gateway,
+				; client's nat router will not accept the packets since that sip gateway's IP is
+				; not in NAT cache).
+
 ;
 ; If regcontext is specified, Asterisk will dynamically create and destroy a
 ; NoOp priority 1 extension for a given peer who registers or unregisters with