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(a)NetUSE.DE --
http://NetUSE.DE/