[OpenSIPS-Users] Integration with Asterisk/Trixbox

[Iñaki Baz Castillo]

El Miércoles, 20 de Mayo de 2009, James Lamanna escribió:
> I'm using the basic NAT example with a little rewriting (as shown below).
> The problem I have is that the SDP address is not being rewritten, so
> on the asterisk box I see audio traces like this:
> (x.x.x.x is the phone NAT IP address, y.y.y.y is the asterisk box address).
> You'll notice that the first 2 lines are correct, but the second 2 are
> not (asterisk sends to the private IP of the phone)

An easy options is forcing the media to always pass though Asterisk. For this, 
just add "nat=yes" in opensips peer in sip.conf.
"nat=yes" enables Comedia Mode which means that Asterisk doesn't care about 
the public or private address of the received SDP. Instead if waits the peer 
to send RTP to Asterisk, and then Asterisk sends its RTP to the source address 
of the RTP.

-- 
Iñaki Baz Castillo <ibc at aliax.net>