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 <[email protected]
<mailto:[email protected]>> 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%[email protected]>")
{
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 <[email protected]
<mailto:[email protected]>> 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"
<[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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