New Implementation advise from experienced freepbx users

I’m planning to install freepbx distro on a private clound and set it up with a SIP service for few lines. I’m having bit of hard time deciding phones and other equipment choice I should get for remote locations.

Then the goal is setup 2 locations with 10-15 IP phones each. I’ll be assigning 6-8 SIP ports to each location initially. One of the location will have single DID and other will have two DIDs.

Here are some of the requirements for each location that I really need your opinion (I need to basically make IP PBX mimic traditional pbx):

  • When a inbound line is picked up, all other phones should show a busy light or some other way to show that this particular line/port is busy. I know this is more of traditional pbx thing.

  • optionally, When a particular phone/extension is busy, other phones in that location needs to show that its busy.

  • Be able put a line on hold/park quickly preferably by pressing a button.

  • Be able to see all lines on hold/park from any phone and pickup that line. This is very important.

  • intercom announcement like “call on line 71”

  • overhead paging speaker announcement. (we’ve analog speakers currently setup on analog pbx via a paging amplifier).

Keeping above requirements in mind, which phones will be the best to use? Grandstream, Aastra, Snom? Which model? Budget is under $80/phone if possible.

Each location should have at least couple of phone geared towards handling lot of calls and rest simpler desktop/floor use.

what would be a most economical paging solution while keeping same analog overhead speakers/horns.

When dealing with SIP/asterisk, there is really no concept of “Lines” . Each telephone has it’s own extension number. If you’re trying to make a PBX behave like a key system, you’re pretty much going to have problems. I know of several instances where the end user really wanted a 1A2 key system and ripped out the Asterisk system even though it was working perfectly!

The first two requirements can be taken care of with a BLF programmed for each extension. You’ll need e large enough BLF to handle all of your extensions.

A single button in the BLF can be programmed for parking a call. Particularly easy to do on AASTRA phones.

You can program each parking slot in a BLF.

Paging/intercom available.

You can use your existing paging system. Get something like the Algo paging speaker. It’s a sip device with a built in amp and speaker and terminals for connecting it to PA amp.

You’re not going to get a quality phone for $80.00. Coupled with your need for a very large BLF, You could easily be talking 2 to 3 times that amount.

Oh…by the way…a DID is not the same as a trunk.

You also need an analog paging gateway for the speakers. There really is only one good one. It’s made by Valcom and runs about 600.

Bill is also right I don’t thing you have a good grasp on how I trunks work. SIP is just a protocol. It has nothing to do with concurrent calls. That is dependent on how your provider sells service.

Lastly 80 is really not a reasonable price point. The Aastra 6731i is a about 120 and has 7 usable programmable keys with LED you also need a PoE switch and a UPS. If you try and use power bricks you will be hating life.

Let us know how your deployment goes.

Bill - Thank you for your insight help. This is great!!

Sorry if I’m not using the right words… I do tend to mix analog/IP lingo… My SIP service sells $/channel (i’m referring these channels as lines).

The locations have been using analog pbx for so long that If i just replace analog to ip pbx, they’re going to have a hard time getting use to the new system. So I really want to ease in the new system and therefore the need for some feature matching.

you mentioned that extensions can be programmed to BLFs. Can SIP channels be programmed to BLFs?

Thank You very much!!

@SkykingOH - Thank you for the comment.
I already have analog paging setup that works with the analog pbx. I’m thinking about testing ATA device and feed the analog signal to my paging controller/amplifier. I’ve read many forums suggesting this route for existing equipment.

I’ll be able to PoE switch for some of the phones. But large number of phones will be pretty far from each other and will be using existing network cabling.

Thank you.

You can save a buck and use chan_console as a channel driver if you have audio hardware on your server.

my server is sitting in a data center. so this won’t be possible. Cheapest option seems like setting up ATA adapter and feed that to my paging controller.