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

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.