[asterisk-dev] RTP streams between H323 and SIP

[Ken Chan]

Hello,
I am using Asterisk 1.2.9.1.

I have a h323 phone (OpenPhone) and SIP Phone.  I make a 
call between those 2 phones.

I set the "canreinvite=yes" for SIP phone.

The RTP stream for the h323 phone still go through Asterisk.
Anyone can give me a hand to fix this problem?
I want the RTP streams to go from endpoint to endpoint
without going through Asterisk.

I have tried different h323 channel driver such as 
"chan_h323" (comes with Asterisk), Asterisk-Addon
h323, asterisk-oh323 (from inAcess Network) and 
Objective Oh323 channel driver.

None of them can solve my problem.  Any idea?
ken

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