I have a FreePBX instance that is taking care of hosting multiple SIP speakers in a school for paging/intercom. I would like to connect this FreePBX system to an on prem VoIP system the school uses for their normal phones. The simplest way to do this would be to light up a trunk between the systems but that occurs a large license cost on the side of the school district. I have been asked if I can instead register as a SIP ‘station’ to the phone VoIP server.
From some googling, it seems possible to configure a ‘SIP trunk’ to register as a SIP station, but its a little out in the weeds past what I really understand. Anyone able to shed a little light? Thank you!
You could do it that way but if you run into problems another way would be to buy 2 gateways, one an FXO gateway and one an FXS gateway, and put the FXS on the school’s system and the FXO gateway on the FreePBX system, then plug them in back to back.
(this post mainly for amusement value to illustrate the futility of the “license every last little component separately” approach)
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There’s no difference between registering as a SIP trunk and as a SIP station, as SIP doesn’t have a concept of a trunk.
Typical “extensions” expect the PABX to outbound register and outbound authenticate. They should reflect the contact user, as the request URI user, to follow the SIP specification. This is what FreePBX treats, by default, as the “DID”.
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@david55, this is what I have read online, and seems to make sense.
When it comes to registering a typical SIP device, I’ve typically only had to point the device at the VoIP server IP, enter the extension number, and the SIP Auth Username and Password, and thats about it for a simple setup. When it comes to the SIP trunk config area of FreePBX, there are a LOT of options and I think I’m a bit confused on what I need to config, along with anything else on the Asterisk side that might be needed. Any insight, examples, or other helpful hints to get me going?
I would take the defaults on almost everything.
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