[lemminge]

Hello,
i have a problem setting up an asterisk as sip server in routed nets.

My 3 networks:
i) 192.168.1.0/24 with one ip phone
ii) 192.168.2.0/24 with the asterisk
iii) 192.168.3.0/24 with another ip phone

Facts:
- net1 has a route to net2
- net3 has a route to net2
- net1 and net3 does not know each other
- everything via vpn without nat!
- both phones are registered and can call each other, but there is no audio because they try to send the audio data directly

Is there a way the asterisk acts as a client and redirects the audio data to each other?

Thanks, Lemminge