We are essentially in BETA right now for the first release so we would welcome your input as a tester. FreeSWITCH is more than ready to move SIP traffic, all I would warn is that some of the configuration and behaviors therein have not been finalized and are hence subject to change between now and the final release but there probably aren't many. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Fred <codecomplete@free.fr> To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, May 22, 2007 2:52:12 PM Subject: [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. _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7