Remote SIP access

Hi! I have tried EVERYTHING to get a client across Melbourne to be able to remotely connect to a extension! (Yes I understand the risk). I have allowed port 5060 to the PBX and have nat=yes on the extension. What am I missing!?

Have you check the Asterisk log file /var/log/asterisk/full? Is there anything in there about the remote extension trying to register, etc?

How would we know what you are missing since you did not tell us what you did (other than set NAT and open one port).

What kind of firewall?
What version Asterisk/FreePBX?
Have you read one of the 100’s of guides to NAT and Asterisk on the Internet?
Did you forward UDP 5060
Did you forward the audio ports as defined in /etc/asterisk/rtp.conf (you only need roughly 4 per concurrent call path).
Did you configure the NAT section in SIP settings to match your network environment?
Are the packets even making it to your server?
Go back and reread your question and ask yourself, how could anyone answer that?

I found the issue, I have to run iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT when I start the server for it to work, as soon I restart it stops until i run it again. How can i make it permanent?

You can add this setting to /etc/sysconfig/iptables. However, we very careful opening up SIP to the Internet generally. This can lead to hackers getting into your system and hijacking your trunks (and costing you a lot of money in the process)!