Dial the whole wide world

My asterisk FreePBX is now on the Internet. Extensions to my PBX can connect from anywhere in the world. But other than a SIPURA SPA3102 (1FXO+1FXS), the PBX is an island of its own.

What does it now take for my extensions to dial any globally unique sip address in the world (those take incoming calls from anyone)?


You are in way over your head.

  1. Your system should not be sitting on the internet. If you keep it there, in a few days, it will likely be hacked. Put it behind a NAT firewall, or at least configure IPTables immediately.
  2. You don’t really need an Asterisk system to place SIP URI Calls.
  3. To receive SIP URI calls, just tell people to direct calls to EXTENSION#@YOURIPADDRESS. You’ll also need to enable allow anonymous inbound calls in the General Settings module. So, for example, if your IP address is and your extension number is 100, your SIP URI is:

[email protected]

  1. To place SIP URI Calls, you’ll need to create a custom trunk with the dialstring:

SIP/[email protected]

Change IPADDRESS to the IP Address at the end of the SIP URI. $OUTNUM$ is a variable that represents the number dialed by your extension. You can replace it with something else if you want, i.e.

SIP/[email protected]

Would place a SIP URI call to [email protected] (which is not real, so don’t use it).

Are you sure you want to make SIP URI calls? Most people just sign up with an ITSP like Callcentric or VOIP.MS or flowroute and make calls using those services…


My PBX is NAT behind a firewall. But the necessary ports have been opened and mapped so that it is now accessible for SIP and RDP traffic.

Your second point 3 is what I was looking for. Thanks and I will follow up on it.

Why I want to dial SIP because I read that Google Voice numbers can be dialed directly.