[asterisk-users] SIP - ooh323 Bridging

[Richard Scobie]

Hi,

I have the following setup, with asterisk on a dual homed box:


PolyIP500(SIP)--<192.168.4.0>--Asterisk--<192.168.0.0>--Panasonic(H323)

It is running a recent SVN version of Asterisk 1.2 and ooh323.

The problem I have, is that despite having "canreinvite=no" in the 
sip.conf, asterisk still insists in attempting to native bridge the RTP 
streams:

     -- Executing Dial("OOH323/192.168.0.2-540d", "SIP/polywn1") in new 
stack
     -- Called polywn1
     -- SIP/polywn1-0817e828 is ringing
     -- SIP/polywn1-0817e828 answered OOH323/192.168.0.2-540d
     -- Attempting native bridge of OOH323/192.168.0.2-540d and 
SIP/polywn1-0817e828

This results in audio only running in the H323 to SIP direction - 
nothing the other way.

There is NO NAT involved in the asterisk box.

Investigation with Wireshark shows that as the call is setup, a couple 
of packets worth of RTP audio does flow in the SIP-H323 direction, until 
the native bridge occurs, at which point it fails.

I realise that this may be something that is fixed in 1.4, but that is 
not an option at this stage.

Can anyone offer a way of forcing asterisk to stay in the path and not 
be bridged out?

Regards,

Richard