This should have all the info you need.
Set up two forwarding entries the "Port Forwarding" (or similar) configuration form on the NAT configuration interface, each of which cause the NAT device to forward all traffic destined for the designated range of port numbers to the fixed IP address of the SIP phone:
* SIP signaling: Ports 5060 to 5070
* RTP audio: Ports 8766 to 35000
Or if you are running Asterisk and everything supports IAX, use IAX instead of SIP. IAX was designed to work through NAT without all the hassles..