[Freeswitch-users] Does FS handle remote NATed phones?

[Fred]

Hello

	It's been over a week now, and I still can't get voice RX/TX to work 
between a Linksys VoIP gateway and a remote IP phone with Asterisk as the PBX:

http://codecomplete.free.fr/3102_nat/voip_no_sound_open_ports.jpg

The phone rings, but no sound either way.

I'm beginning to wonder if Asterisk can handle being behind a NAT, and 
bridge calls to remote phones that are themselves behind a NAT :-/ If not, 
is FreeSwitch the solution because of its support for STUN? Is it ready for 
production, or is it a bit early yet?

Thank you
Fred.