[asterisk-users] NAT

[Shaun]

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



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