SIP realtime use


I’m using Asterisk 10 on a Fedora machine. The SIP peers are supposed to be handled by realtime, but when SIP peers try to call each other audio typically becomes one-way for the one of the peers. However, if the authentication info is stored in sip.conf everything works as it should! Does anyone know whats wrong here.

The environment:

Fedora 17 server
Mysql 5.1 on another server
Asterisk 10.9.0

BR. Aggi