I am using a SIP Trunk from sipstation. I do not have a static IP address on my internet line. Is there anyway that I can use the sip trunk without having a static IP address?


A static IP is not required to use SIPStation. FreePBX registers your IP with the SIP Station servers.

Thats what I thought but I am not getting any audio. I have NAT setup in my router. Does anything need to go in the SIP_NAT.conf if the external IP is Dynamic?

How could I possibly know? You did not provide us with any information on your system. sip_nat.conf was deprecated many releases ago in favor of the sip settings module.

No matter what the method you need to set the externhost and localnet directives if you want audio to work.

Please start your exploration of FreePBX by reading the documentation that I painstakingly wrote. You can find it here:

“Tools” > “Asterisk SIP settings”

set “NAT” to Yes
set “IP Configuration” to Dynamic IP (you need to register a free dynamic host at
set “Dynamic Host” to - 120 refresh rate
set “Local Networks” to match your lan IP range (eg.

install and configure ddclient to your Asterisk linux box to update your dynamic host when external IP changes ( yum install ddclient - see for configuration)

Configure your router to forward UDP Ports: 5060 and from 10000 to 20000 to your PBX IP address (lan side. eg.

That’s all.