|
I don't think the problem is NAT-related. Looks like "To" header in SIP INVITE message do not match to "User ID" in sipura settings. On Thu, 2005-10-27 at 01:57 +0330, mohammad mirzaee wrote: Hi ALL; I have users with Sipura/Linksys phones regsitered behind Nat( useing STUN at phone not portforwarding ) in my Asterisk box, when I try to call them with another phone i got: Got SIP response 404 "Not Found" back from 217.6.190.4 SIP/217.6.190.4:5060-666d is circuit-busy Is above mentioned problem relates to "Nat", Is there anybody who use sipura with STUN method and can recive calls? My asterisk Sip.conf for Nat has the following: [sipura] .. nat=yes canreinvite=no qualify=1000 Appreciate any help Mohammad _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users |
_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
