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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