I'm running Asterisk SVN-trunk-r51353... for some reason even if i set nat=yes in the sip.conf for a device when i do a show sip peers it shows N for nat. Is this a bug or am i doing somthing wrong here. I'm basically having a problem right now where i can call in/out of asterisk and talk fine using the phone but if i call another SIP phone registered to the same asterisk server it rings but when i pickup both ends i cant hear or say anything through them. Not sure if this is related to the NAT issue... i think it may be?
my setup is... Asterisk is on a public ip 2 polycom601 phones on a private network here's a sip debug... http://channels.debian.net/paste/5164 ~Shaun
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
