If you run your own dialer (such as VICIdial) or a PBX (such as Asterisk or FreePBX), a SIP trunk is how calls get from your server to real phone numbers — and back. Think of it as a virtual phone line delivered over the internet, with as many simultaneous calls as you need.
The two directions
- Termination (outbound): your server sends calls to the trunk, and the provider connects them to the public phone network.
- Origination (inbound): someone calls one of your numbers, and the provider delivers the call to your server.
How the provider knows it is you
Most trunks for servers use IP authentication: you list your server's public IP address, and only calls coming from that address are accepted. Some setups also use a username and password (credential authentication). IP authentication is simpler and works well when your server has a fixed public IP.
What you need
- A server with a static public IP.
- A SIP trunk with your server's IP in the allowed list.
- An inbound destination — your server's IP or host name and SIP port (usually 5060).
- Phone numbers attached to the trunk.
- Firewall rules allowing the provider's signalling and media IP ranges.
Setting one up on ReachMeLine
- Open SIP trunks in your account and create a trunk — it is named automatically.
- Add your server's public IP under Allowed IPs.
- Add your server under Inbound server so calls to your numbers reach it.
- Attach one or more numbers to the trunk.
- Allow the listed signalling and media IPs on your firewall.
Common problems
- Calls rejected: the IP your server actually uses is not the one you allowed — check with a "what is my IP" lookup from the server itself.
- Inbound calls not arriving: wrong port, or the firewall is blocking the provider's IPs.
- One-way or no audio: almost always NAT or firewall settings for RTP media — see our firewall and NAT guide.
Tip: make one test call in each direction after every change, and keep a note of your server IP and trunk settings somewhere your whole team can find them.
