Hello Jagadish,
Using a network tracer (tcpdump, ngrep, wireshark), do you see the
INVITE going out (sent out by OpenSIPS) ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 04/04/2013 02:46 AM, Jagadish Thoutam wrote:
Hi All,
i having issue with URI routing , when i am trying with the Voip
Provider IP its Not Going Through, i have IP authentication with Provider
here is the my script
if (is_method("INVITE")) {
setflag(1);
if (uri=~"sip:[0-9]{10,11}@192.168.XX.XX") # Asterisk server
{
xlog("*********CALL WILL GO HERE VOIP PROVIDER********");
xlog("*****************GOING TO ROUTE @6****************");
route(6);
}
}
route[6] {
rewritehostport("64.XX.XX.XX:5060"); # VOIP Provider IP Address
xlog("*********CALL WILL GO TO VOIP GATEWAY @@@@@@OUT********");
t_relay();
exit;
}
Thanks
Jagan
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