Sending voice and sip signalling to different servers


I’ve been using astersik for quite a while now. Mostly I use asterisk as auto-dialout send message. Right now, I’m experimenting to connect to a nokia sip and media server. The nokia servers is connected to a TDM network.Right now I’m experiencing problem with our counterpart has two server. I have configured my asterisk to connect to their sip server.I did not added in my configuration the media server since I thought that registering to their sip server will all be fine. The problem began when I tried to call to a mobile number. There were no ringing on the handset. I’ve also got a “487 Request terminated”. As per our counterpart, I should have added their media server. He also said that the voice traffic is being handled by the media server and my system should directly passed the voice traffic to their media server in which it diffrent vlan/network. My problem is where am I going to send the the call request?
Here is in my extensions.conf

exten => X,1,Dial(SIP/{EXTEN}@x.x.x.x)" - where x.x.x.x is their sip server. Where am going to configure the the rtp/voice packets directly to their media server.

Am I going to use canreninvite=yes? Or do I need a rtp proxy? Can asterisk send sip signalling to a different server and send the voice packets to different server.

I hope my I’m not confusing everyone. Thanks