[Asterisk-Users] NAT/Qualify/RTP bug

[Arthur B Olsen]

Got a really wierd problem her. Maby it's a bug.
But before i report it, i'll try my luck here.

I have one asterisk server on public ip.

I have two identical hardphones on two different LAN's. The firewall are 
different.

Both are configured in asterisk with nat=>yes and qualify=>yes.

For one phone everything works. SIP and audio is sent to the global address of 
the client.

But for the other it's a bit different. SIP messages are sent to the global 
address of the client. You can call in and out. But the audio (RTP) is sent 
to the local address found in the SIP packets.

The only thing that is different is the firewalls.

How can a firewall, or anything else, tell asterisk to use the ipaddress in 
the sip packets instead of the global address, when i have told asterisk 
nat=>yes????

Is this a bug? Or something i've missed.


PS: i'v tried nat=>route, same results

Thanks.