Here is my problem: The Asterisk server is NOT behind NAT. On an incomming call neither side has audio. Internal and outgoing calls work fine. An outside line can also leave a message on the asterisk server and the inside line can listen to the message. One would think that the problem lies between the asterisk box and the inside phone, but internal calls work fine. It is only when an outside line dials to an inside line.
canreinvite=no in sip.conf
This worked. Now I gotta figure out why.