I've got freeswitch installed with a digium te110p hooked up to a PRI. calls through the system work fine SIP->SIP, calling and receiving calls over the PRI works, (IE the signalling part, a call can be made and incoming calls come in over the PRI and I can route them based on DID), however, there is no audio to either party. I've checked, and there is an RTP stream running while the call is up, there is no reason it would be blocked (IE, SIP->SIP calls use the same RTP port range, and they work fine). I saw a few posts that talked about no audio for a few seconds after a call is connected, or that calls would connect and then drop after a few seconds. This is different, I can leave the call up for many minutes, it doesn't drop, and audio never starts coming through. I'm running openzap natively, no libpri. I am using the dahdi 2.2.1, tried the latest release of zaptel first, freeswitch wouldn't even load with that. I'm going to try libpri now I think as I've exhausted all the other options I can think of. Any input/ideas would be greatly appreciated. -Tom