H323 <-> SIP Translation

Hello everyone,

i am trying to set up Asterisk to work as an intermediate point between H323 client and a SIP provider
As a first step, i’m testing outboud calls from H323 to SIP:
I installed OpenH323 add-on to Asterisk, and set up the SIP trunk to the provider; the caller is a phone connected to a PABX, the PABX uses a software from Aastra called NexSpan that establishes the H323 call with Asterisk, and then gives the address of Asterisk to the PABX so it can forward RTP packets;
the call establishement is working fine, the only problem is that the caller’s voice isn’t heard at the other eand; the callee doesn’t hear anything, but the caller can hear the callee perfectly; i already checked and it’s not a NAT problem;
when capturing the packets, i see something strange that maybe could be the origin of the problem: there are two RTP flows opened from the same source to the same destination (PABX to Asterisk), with the same source and destination ports!

does anyone have an idea how to resolve this?

Thanks in advance

So i have been modifying settings in oh323.conf in Asterisk, and testing with a softphone that connects directly to the Asterisk;
when i activate the “faststart”, the sound passes in both directions, but the caller’s voice isn’t very clear; when faststart is deactivated, the caller’s voice doesn’t pass anymore…
any suggestions?