SIP to H323 Calls with Asterisk and Avaya

I have an Avaya PBX that uses H323. The Avaya PBX has a H323 Trunk to my Asterisk box. The Asterisk box has SIP clients that work with it. The issue I am having is that when a sip client calls an h323 client or vice versa the only audio that is heard is the Avaya phone. I know why this is occuring I am just not sure how to correct the problem.

The SIP clients are set up to canreinvite=no so that all rtp traffic will go through the Asterisk box. When I sniff the traffic on the Asterisk box I see that Asterisk first talks to the Avaya CLAN IP address and then the CLAN tells me to start talking with the Avaya Media Resource IP. Once the call is established the Asterisk PBX starts receiving RTP traffic from the actual H323 Avaya phone which is why I am able to hear the Avaya audio on the sip phone. The problem is that the Asterisk PBX is sending the audio coming from the sip phone to the Avaya Media Resource IP still. It's like in the process between Asterisk and the Avaya CLAN no one ever told the Asterisk PBX to send the RTP traffic to the Avaya phone.

I know that one solution is to have all H323 calls go through the Avaya PBX like im doing for SIP on Asterisk, but the Avaya PBX won't be able to handle that.

Here is what the sniffer is showing me


IP Addresses of Devices
SIPphone ------ Asterisk ------ AvayaPBXMedRes ---- H323phone
x.x.250.100 --- x.x.250.10 ----- x.9.2.208 ------- x.9.2.107



Sniffer data
Source ---------- Destination -------------- Protocol
x.x.250.100 ----- x.x.250.10 --------------- RTP

x.x.250.10 ------ x.9.2.208 ---------------- RTP

x.9.2.107 ------- x.x.250.10 --------------- RTP

x.x.250.10 ------ x.x.250.100 -------------- RTP

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