[OpenSIPS-Users] Sending call to Gateway

[Bogdan-Andrei Iancu]

Ok, then the IPs in SDP cannot ensure the RTP connectivity (some private 
IPs ?) - check the IPs on each SDP and be sure the end point can use 
them to route to the other party.

Regards,

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


On 04/08/2013 04:57 PM, Jagadish Thoutam wrote:
> HI Bogdan,
>
>  Yes it is.  But No Audio.
>
>
>
>
> Thanks
>
> Jagadish
>
>
> On 8 April 2013 08:20, Bogdan-Andrei Iancu <bogdan at opensips.org 
> <mailto:bogdan at opensips.org>> wrote:
>
>     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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