Re: [Serusers] Ser 0.9.0 and Cisco Voice Gateway

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