[OpenSIPS-Users] Sending call to Gateway

[Bogdan-Andrei Iancu]

Hello,

this still does not answer to my question - does your SIP signaling work 
ok (for the established call) ?

Regards,

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


On 04/05/2013 05:36 PM, Jagadish Thoutam wrote:
> 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 <mailto:11%7D at 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 
> <mailto: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 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
>>>
>>>
>>>         _______________________________________________
>>>         Users mailing list
>>>         Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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