I´ve just set up a copy of FreePBX on a VPS, and am going through the configuration at the moment.
I have a few DID´s with a couple of compnaies which I want to route to my FreePBX. The SIP URI needs to be [email protected] - but what is the something part in FreePBX? Is it a particular extension number, e.g. [email protected] ?
The something (referred to as the extension in Asterisk speak) will depend on the context your calls arrive in. You will probably create a trunk to receive these calls, and this trunk will probably have a context of
from-trunk. If so, the something will be a DID defined in your inbound routes.
This is an unusual way of doing this. For individual phones, SIP URIs make sense, but for a PBX, the interface is different enough that you might not get the actions you are looking for.
As Lorne mentioned, connecting to the server with "[email protected]" would be the way you’d do that, especially if your DID provider is expecting this. A more likely situation is that aren’t currently expecting you to have a PBX, so they aren’t set up for that right now.
If it was me, I’d call your providers and ask them “What do I have to do to connect my number to Asterisk?” I expect one of three answers:
- Nope - can’t be done.
- Sure - here’s the info you need.
- Why - you’re already hooked to a PBX.
The answer you get will determine how much help you need from us and whether or not you can even start.
Although I agree completely with @cynjut 's solution, I disagree that it is ‘unusual’. It’s what virtually every ‘wholesale’ provider requires, what most high-end retailers recommend and what many other providers permit.
Assuming that your PBX has a static public IP address or is behind a NAT router with a static public address, this is the most reliable way to receive calls. The alternative is that your PBX registers to the provider, which is generally less robust against temporary network outages, router anomalies, etc.
If you choose the SIP URI method, there should be a list of IP addresses from which your provider can send calls, which you will need to open in your firewall. (If your SIP port is open to the world to allow extensions anywhere, confirm that your other security measures are in place, including strong passwords, fail2ban and denial of external calls from all unauthenticated contexts.)
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