[Serusers] dialing PSTN using x-lite

[Jan Janak]

404 means that the phone you are trying to call is not registered on the
server. Make sure that you have proper domain in subscriber table
(sipserv.foo.com and not just foo.com).

 Jan.

On 06-01 15:21, Andy Singh wrote:
> 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
> 
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