I just set up my first FreePBX installation with one trunk, one simple outbound route and one simple inbound route. 10 digit dial rule with 1 prepend. If, for example, I make a call to the 303 10 digit time number I get the following error message saying it cannot find the extension. It should go to the trunk… 710 is the extension making the call.
Everything shows registered correctly.
If I make an incoming call from an outside lthe console shows,the following error message with %%%%%% being the correct incoming CID.
@lgaetz is of course correct. The text to-pstn should not appear in any config file. If you manually typed it somewhere, you likely want from-internal, but please explain what you did.
Otherwise, you may have installed a third party module that uses the name to-pstn. Did you do something related?
Not that I am aware of. So what can I do to fix the issue to have dialed numbers recognized and sent to my one trunk, rather than treated as an internal extension call and go nowhere. I have tried various settings for the context variable, but none seem to direct calls to the one current trunk I am using.
With the default installation, you just need to creare the extensions, the trunk and the routes. However if you have manually modified the default contexts, I think the easiest solution would be to undo what you have done.
There’s a “thinko” in here somewhere that we need to talk about. It’s kind of fundamental to how the system works.
When you dial a phone and create a SIP call, the call goes to the PBX which answers your call. The PBX routes the call through the “context” of the extension is set in the Extension definition tab. Without getting into a lot of detail, the context is a little program that instructs the PBX what to do with an incoming call. For purposes of what’s happening here (dialing out) you should be use the “from-internal” context because it easily connects to the outbound routes.
This handles control of the call through your outbound routes. The outbound routes can process or edit the dialed number in several ways, but if the called number matches what the outbound route can handle, the route sends the call to one of its associated trunks.
The trunk then tells the PBX to place a call over the associated trunk to the number you dialed. This call is then merged (bridged) into your call and the two endpoints create a little “conference” (it isn’t really, but thinking of it this way can clarify the theory).
So, in order for your outbound call to work, you need to have a trunk (which communicates with your ITSP) that is associated with an Outbound Route, and the number that the Outbound Route call needs to work with that trunk. Remember, the PBX is the one placing and answering the calls - Asterisk is a Back to Back Agent, so the phones never talk to each other.
Hopefully, that will help you visualize what the call is supposed to be doing.