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.
----- 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:
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?