[OpenSIPS-Users] Sending call to Gateway

[Jagadish Thoutam]

Hi Bogdan,


here is my setup

(X-lite)client---------->Asterisk-----> Opensips(NAT) ----->Gateway.


*################################## Opensips Config
File#####################  *

*route{

if (is_method("INVITE")) {
setflag(1); # do accouting
if (uri=~"sip:[0-9]{10,11}@192.168.7.80")
{
xlog("*********CALL WILL GO HERE VOIP INOVATIONS********");
xlog("*****************GOING TO ROUTE @6****************");
route(6);
}

}

route[6] {
rewritehost("67.37.xx.35:5060"); # Provider IP
xlog("************** new ruri=<$ru>, dst=<$du>***********\n");
xlog("***********$ru**************\n");
xlog("*********CALL WILL GO TO VOIP @@@@@@@@@OUT BOUND @@@@@@@********");
t_relay();
exit;
}


*
*Here is My Trace File see attachment


*
*Thanks
*
*Jagadish
*
*
  *


On 5 April 2013 09:34, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:

> **
> 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 Developerhttp://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> 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 Developerhttp://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
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
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