[asterisk-users] SIP - ooh323 Bridging

Richard Scobie r.scobie at clear.net.nz
Mon Nov 19 20:59:06 CST 2007


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


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 

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

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?



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