Force RTP Keep Alive upon connecting?

My asterisk installation is doing some strange activity when it tries to bridge two SIP legs (one from the provider, and the other another PBX server).

It does not initiate an RTP steam until the remote PBX sends a noise. And this happens when we try to dial outbound from the Remote PBX --> Asterisk --> SIP Provider. Dialing inbound works correctly.

I have a quick fix – that is to set the RTPKeepAlive value to 1… so that after one second it sends an RTP packet, and that’s enough to get things going. However, 1 second is not going to cut it because that’s when the callee says “Hello”. This causes much confusion and frustration because the callee simply hangs up after a while because we didn’t know they answered.

Is there a way to send an RTP Keep Alive packet immediately upon answering? THat would probably be the best fix. I spent many days pinpointing this problem, trying to implement premature media, directRTPsetup, directmedia, etc… you name it.