asterisk users - Feb 2005 - incoming h323 calls, routed to SIP/H323 drop after connection

Hello, 

I am attempting to use Asterisk as a protocol converter.

I have set up asterisk to route incoming h323 calls to a SIP termination 
carrier. 

I make a test, call is coming correctly, is rerouted to termination carrier. 
Call connects and phone rings. Then, I pick up the phone and it hangs up after 
2 seconds. 

I initially thought it was a codec issue. I made sure codec is g729 in all 
sip.conf & h323.conf parts (general context + specific contexts). 

Still, call drops after connects and gives error "cannot bridge between X
call
and Y call". 

Is this familiar to anyone? Do you have idea what to search next?