[Asterisk-Users] RTP dataflow directly from a SIP phone to a H323 phone

[pesb]

Hi there,
             Is there anyway to make the RTP data flow directly a SIP phone 
and a H323 phone through the oh323 or chan_h323 modules? Something like waht 
the canreinvite = yes option inside the sip.conf does for SIP to SIP calls.

thanks,
               Pablo Salinas