OK - slow down. What you said doesn’t comport with the way the system works.
Asterisk (and by extension, FreePBX) is a back to back user agent. One of the ‘backs’ is the extension interface. You connect extensions to the system and your extensions interact with Asterisk. The other ‘back’ is the trunk interface, which is connected to Asterisk through a series of routes (inbound and outbound).
The only connection between these (extensions and trunks) is that they go through Asterisk.
So, as the tutorials and wiki pages describe, the process is something like the following:
- Set up your extension(s) so you can communicate with the PBX.
- Set up a Provider Account. This can be incoming, outgoing, or both. That last bit might not seem important, but once you understand that it’s really a thing, your conceptual curtain will be lifted.
- Set up your trunk(s). There is an incoming and outgoing “leg” that are often handled as if they are one thing, but for simplicity, it helps to think of them as if they are a coincidence - your incoming calls can come from anywhere you have a trunk, and your outgoing calls can go anywhere you have a trunk.
- Set up your outgoing so that the provider gets your outbound traffic.
- Set up your incoming routes so that the provider’s traffic has some place to go.
Calls from your providers will come into your local trunk, regardless of the ‘flavor’ of the call. It can have any of your numbers (as assigned by your provider) and the trunk just doesn’t care.
These trunk calls are sent to the Inbound Routes. These are a series of filters that allow calls to be sent to various destinations in your system. These can include Voice Mail, Announcements, IVRs, extensions, etc. To start, you should always set up at least one inbound route that has no DID or CID settings - this is your ‘any/any’ or default inbound route. Wherever you send the call from here is where any call that doesn’t meet either the DID or CID filter on the way in.
Outbound routes work the same basic way, except from the Asterisk side. When you dial a number, it is filtered through the settings in the Outbound Route, which then connects to one or more trunks.
In the GUI there are Asterisk Reports. These will list your trunks.
Your outbound Caller ID will always be in the form “Name with or without spaces” <xxxxxxxxxxx> . Note that the quotes and angle brackets are required.
There’s no way of knowing at this point.
This is a trunk setting, and if set up correctly, it will allow the incoming calls to be processed by your inbound routes.
The file /var/log/asterisk/full will tell you what is happening at all points in all of your problematic situations.
Your phones do not connect to the trunk. They connect to Asterisk, which is managed by FreePBX. If your phones are not working, it’s not probably a problem with your setup of the extensions and the PBX than it is the trunk. Let me say that one last time - there is no connection between your ITSP and your phones except Asterisk. Think of it like Asterisk being the escrow for your data. Everything goes through Asterisk.