Hi Raul,
Try canreinvite=no in your sip.conf file. Then all calls will go via
asterisk.
Marcus
Raul Dias skrev:
Hi,
I have the following setup:
[ Voip Provider ] ------ (XX) XXXX-XXXX
x.x.x.x (real world phone number)
|
{ The Internet }
|
200.x.x.x (Internet IP)
[linux router]
10.0.51.1
|
------------------------- -> (The Lan)
| |
[sip peer 1/client] [asterisk server]
10.0.51.3 10.0.51.2
The linux router does Nat/firewall for The Lan.
sip clients inside the Lan can talk to each other (and asterisk) fine.
The router has port forwarding for IAX[2], SIP, RTP (10000-20000) and
MGCP to the asterisk box (10.0.51.2).
When I have a call between the outside world (VOIP provider) and a
internal sip peer, I can see that the data transfer (RTP) is between the
the VOIP provider and the client (10.0.51.3).
That said, the PROBLEM is:
After a few seconds (2 to 20) the call becomes mute (but still active).
It does not matter which side started the call.
For what I understood, shouldnt asterisk (10.0.51.2) work as a proxy for
(10.0.51.3), instead of letting it talk directly with the VOIP Provider?
I think that this is where the problem is. In sip.conf I have externip
set to the router Internet ip address. However as the peer is also
behind the nat (10.0.51.3), the voip provider will see the same IP
because of nat. But they are different boxes.
- Raul Dias
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
