On Wed, 16 Jan 2008 18:08:23 +0000 (GMT), Gordon Henderson <[EMAIL PROTECTED]> wrote: >However, you'll need to do similar things to your asterisk box & router if >it's behind NAT for IAX as you do for SIP. (You will need a static IP >address on the NAT router and port-forward 4569 to the asterisk box, just >as you'd port-forward 5060 and 10000-20000 for SIP)
Am I wrong to understand that IAX only needs one port, TCP4569 by default? So I only need one port for each phone, while SIP requires at least 3 (SIP, and one RTP each way)? >And a SIP phone behind a NAT router is also solvable if it supports STUN. But not all NAT routers support STUN, ie. keeping UDP ports open so that incoming packets can make it. >I know that SIP behind NAT isn't perfect, but with care, it's very usable >and workable But unless I'm mistaken, when NAT is involved, canreinvite must be set to no, ie. all RTP packets must go through Asterisk instead of flowing from one phone to the other? Thanks guys. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
