[OpenSIPS-Users] Sending call to Gateway

[Bogdan-Andrei Iancu]

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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