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" <[email protected]
<mailto:[email protected]>> 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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