Pri Line with FreePBX

I am looking at getting a pri line for our school’s dormitory phones. I was going to install the PRI line into the freepbx box and then assign extensions to each room by assigning 2 virtual extensions to each physical extension for each student to have his own voice mail. Any ideas if that will work and any problems I may run into on that scenario??

Not to be doubting your tech skills, but probably you want a SIP trunk over a PRI trunk, as it is so much easier to configure and troubleshoot. Regarding extensions, why do you want to go with virtual extensions? I’m trying to understand why not go with real extensions.

I guess I thought that way we only needed one jack per room and the students could have their own voice mail.
As far as the pri goes please doubt my skills I have little Idea what I am talking about.
I was thinking of getting a PRI line which would give me 20 or so outgoing/incoming lines to spread out to out 75 students. How would one go about this.

OK - let’s talk about network engineering for a second.

A true Primary Rate Interface (in the US) is effectively 23 POTS telephone channels and a control channel. These are delivered on a 1.5Mbps line from your telephone company to a special card or PRI Interface external to your PBX. This has almost no relationship to SIP. While being precise isn’t in vogue, using the term PRI for anything but incoming and outgoing telephony channels is just going to confuse things.

When talking SIP, it is almost always best and cheapest to go with Internet Service trunk from an Internet Telephony Service Provider (there are hundreds). These can be through a DSL connection, cable modem, or OC1 (Optical Carrier) or DS1 (Data Service 1) line. All of them deliver Internet connectivity to a router/gateway/cable modem. From a theory perspective (which is what’s important right now), all of these are functionally equivalent - the only difference is speed and price.

One piece of confusion that happens here: the sales people at your TELCO may want to refer to your DS1 lines as a “PRI” line. Let them, but remember that you have to set everything up based on an Internet connection - using telephone lines to connect Asterisk to the world is (IMHO) the most expensive way to get connectivity. It also forces you to manage each channel on the PRI instead of just using the trunk delivered from your ITSP.

So, let’s assume you have an IP Network in your dorm rooms. If you install phones with a built-in “IP switch”, you can plug the phone into the network jack on the wall and plug a computer into the phone. Each of these devices will get its own IP address and be connected “to the Internet” (in quotes because your server, router, and firewall setup will actually determine what “to the Internet” means.)

OK. you now have phones connected to the network. The DHCP protocol is pretty much required, and can be used to configure the phones. The phones will exist as extensions off your PBX. The PBX will handle the incoming DID (Direct In Dial) numbers and will handle the outgoing calls from the extensions.

With this set up, and using the standard 10:1 rule, you should be able to get 20 phones onto a DSL line, or a couple of channels on a DS1. SIP protocols (even the fat ones) use less than 32K per channel, so you can get by with LOTS less connectivity than with a PRI or T1.

The other advantage of using this type of connection (instead of a telco connection) is that you are likely to get service at a much cheaper per-minute rate.

If you are asking me, I’d go with SIP Trunks, IP Phones that supports at least two accounts.
One IP Phone per room.
In Endpoint Manager I’d set the default Voicemail button to *98
Then set two BLF keys

Like this, calls for each student gets to the phone, and they have a button that monitors their Voicemail Box.

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