[Serusers] dialing PSTN using x-lite - sr-users

6 Jan 2004


      Hello all,
My sip domain is sipserv.foo.com, i have a user1@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@sipserv.foo.com i immediatly get 404 not found. but i can
dial 1212@foo.com from windows messenger, since i don't have  the option to
specify just @foo.com in x-lite it always dials 
1212@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