[Serusers] dialing PSTN using x-lite

[Andy Singh]

Hello all,

My sip domain is sipserv.foo.com, i have a user1 at sipserv.foo.com. i can log
in fine via messenger and via x-lite 2.0, but when i dial a phone number
let's say 1212 at sipserv.foo.com i immediatly get 404 not found. but i can
dial 1212 at foo.com from windows messenger, since i don't have  the option to
specify just @foo.com in x-lite it always dials 
1212 at sipserv.foo.com. How can i make x-lite dial differently or how can i
make sipserv.foo.com work. Here's my ser.cfg file


#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#

# ----------- global configuration parameters ------------------------

#debug=3         # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no	# (cmd line: -E)

/* Uncomment these lines to enter debugging mode 
debug=7
fork=no
log_stderror=yes
*/

check_via=no	# (cmd. line: -v)
dns=yes           # (cmd. line: -r)
rev_dns=no      # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"

# ------------------ module loading ----------------------------------

# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"

loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"

# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"

# ----------------- setting module-specific parameters ---------------

# -- usrloc params --

#modparam("usrloc", "db_mode",   0)

# Uncomment this if you want to use SQL database 
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config), 
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -------------------------  request routing logic -------------------
# main routing logic

route{

	# initial sanity checks -- messages with
	# max_forwards==0, or excessively long requests
	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		break;
	};
	if ( msg:len > max_len ) {
		sl_send_reply("513", "Message too big");
		break;
	};

	# we record-route all messages -- to make sure that
	# subsequent messages will go through our proxy; that's
	# particularly good if upstream and downstream entities
	# use different transport protocol
	record_route();	
	# loose-route processing
	if (loose_route()) {
		t_relay();
		break;
	};

	# if the request is for other domain use UsrLoc
	# (in case, it does not work, use the following command
	# with proper names and addresses in it)
	if (uri==myself) {

		if (method=="REGISTER") {

# Uncomment this if you want to use digest authentication
		if (!www_authorize("sipserv.foo.com", "subscriber")) {
			www_challenge("sipserv.foo.com", "0");
				break;
			};

			save("location");
			break;
		};

		# native SIP destinations are handled using our USRLOC DB
		if (!lookup("location")) {
			sl_send_reply("404", "Not Found");
			break;
		};
	};
# forward to current uri now; use stateful forwarding; that
	# works reliably even if we forward from TCP to UDP
	if (!t_relay()) {
		sl_reply_error();
	};
 attempt handoff to PSTN
if (uri=~"^sip:3[0-9]*") { ## This assumes that the caller is
    log("Forwarding to PSTN\n");      ##  registered in our realm
  forward( 156.151.96.253, 5060 );  ##  Our Cisco router
   break;
	};

}

	
Please help

Thanks