[Serusers] Ser 0.9.0 and Cisco Voice Gateway

Ahmad Cheikh-Moussa acm at netuse.de
Wed Apr 20 15:15:34 CEST 2005


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


-- 
Ahmad Cheikh-Moussa 
NetUSE AG
Dr.-Hell-Straße, 24107 Kiel, Germany
Telefon: +49 431 2390 400 --  Telefax: +49 431 2390 499
Service: Service at NetUSE.DE --  http://NetUSE.DE/




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