[OpenSIPS-Users] Sending call to Gateway

[Bogdan-Andrei Iancu]

So you actually have a media problem. Is one way audio or no-audio at all ?

As OpenSIPS is nated and the GW public (I assume), is the signaling 
working properly (INVITE+200OK+ACK) ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 04/05/2013 04:08 PM, Jagadish Thoutam wrote:
>
> Yes i can see that, even call is instiating with opensips and provider 
> but no voice.
>
> My opensips is behind the NAT, so is there any issue with nat settings.
>
> Thanks
> jagadish.
>
> sent from samsung S3
>
> On 5 Apr 2013 20:04, "Bogdan-Andrei Iancu" <bogdan at opensips.org 
> <mailto:bogdan at opensips.org>> wrote:
>
>     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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>
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