[asterisk-users] SIP Connection Problems

[Barry Fawthrop]

Hi All

I have a Cisco 7960 which is connected remotely to an Asterisk server.

Both are unfortunately behind NAT.
The Phone registers and is show in sip show peers, with the correct 
public ip for the phone and a 100ms qualify time

(1) I can dial the phone from another phone, it will ring but no voice 
goes through in fact I get this error on * console
      SIP response 481 "Call Leg/Transaction Does Not Exist

(2) the phone can make calls outbound fine, with voice no problems ?

http://channels.debian.net/paste/3409   holds the SIP debug for the 
phone extension 650

port forwarding is also set on both sides, and sip.conf has the nat=yes 
, externip and localnet all set correctly


Thank always

Barry