The first thing that caught my attention is the lack of a port in your pots dial-peer. If this peer is matched where should the call go? Do you have a VWIC interface in your router?

Ahmad Cheikh-Moussa wrote:

Hi!

Phone to Phone call functions now properly. But I still got problems to make an externall call.

Is this configuration right for stateless forwarding ? The ip of the gateway is 192.168.254.30.

Here a part of my ser.cfg: # 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;
   };

   if (src_ip==193.175.135.0/24){
           #force_send_socket(smaug:5080);
           forward(193.175.135.179);
           break;
   }

   #if (uri=~"^sip:0[0-9]*@netuse.de") {
   #    forward(192.168.254.203);
   #    break;
   #}
   # Default route zu Cisco Gateway
   if (method == "INVITE" && uri=~"^sip:0") {
     rewritehostport("192.168.254.203:5060");
     t_relay_to_udp("192.168.254.203", "5060");
     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
   if (!method=="REGISTER") record_route();        

   # subsequent messages withing a dialog should take the
   # path determined by record-routing
   if (loose_route()) {
           # mark routing logic in request
           append_hf("P-hint: rr-enforced\r\n"); 
           route(1);
           break;
   };

   if (!uri==myself) {
           # mark routing logic in request
           append_hf("P-hint: outbound\r\n"); 
           route(1);
           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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # };

                   save("location");
                   break;
           };

           lookup("aliases");
           if (!uri==myself) {
                   append_hf("P-hint: outbound alias\r\n"); 
                   route(1);
                   break;
           };

           # native SIP destinations are handled using our USRLOC DB
           if (!lookup("location")) {
                   sl_send_reply("404", "Not Found");
                   break;
           };
   };
   append_hf("P-hint: usrloc applied\r\n"); 
   route(1);

}

route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

router configuration: voice service voip sip ! ! voice class codec 2 codec preference 1 g711alaw ! dial-peer voice 1 pots description Default-Dial-peer fuer ausgehende Anrufe preference 3 service session max-conn 25 destination-pattern 0T progress_ind alert enable 8 direct-inward-dial ! dial-peer voice 10 voip preference 2 destination-pattern 4.. session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711alaw ! sip-ua set sip-status 401 pstn-cause 127 set sip-status 407 pstn-cause 127 set sip-status 410 pstn-cause 22 set sip-status 415 pstn-cause 127 set sip-status 480 pstn-cause 19 set sip-status 503 pstn-cause 127 set sip-status 580 pstn-cause 127 retry invite 3 retry register 3 timers register 150 registrar ipv4:192.168.254.30 expires 3600 sip-server ipv4:192.168.254.30 !

Thanks, Ahmad