We have to manually change each SIP extension everytime FreePBX reloads asterisks and re-writes the files.
I recommend putting a NAT option in the web form and obeying it’s precense.
We have to change:
[code:1]nat=never[/code:1]
To:
[code:1]nat=yes[/code:1]
Every time in order to get the SIP extensions to work behind the NAT in our office.
Regards,
phpfreak