Can I Stage a PBX In Parallel With an Existing POTS System?

Hello Community,

I have a strange question. I am working towards switching over my business’ Vodavi phone system to a FreePBX system. We have 8 analog lines coming in from our telephony provider. I plan to use DAHDI to interface these lines with the PBX server. These lines come into the building, out of the interface box as twisted pairs. The pairs are punched down into a 66 block with CHAMP connectors that go into our Vodavi system.

The plan I would like to proceed with involves staging the server and getting it set up nearly to completion up until I reach the point of needing to enable the DAHDI interfaces. Ideally, I would like to be able to have the PBX server completely up and running in parralel to the Vodavi system by using the 66 block to distribute the incoming lines to the PBX. In this configuration, both the PBX and Vodavi system will be connected to the POTS lines simultaneously.

Will this cause any problems with the Vodavi system? Does either system “pick-up” the line before a handset is lifted from a desk phone?

I realize these two systems will not be able to communicate with each other - but if I am able to test incoming calls to the PBX system while the old system remains active, I will be able to iron out any kinks long before we fully switch over.

Thanks for your time!

If you setup inbound routes for all channels (or have one that covers all of them), and use the “pause before answer” setting on the inbound route(s) to some very high number, then you can be very sure that Asterisk will not answer the inbound channels. That’s definitely the most reliable way forward, because there are a number of situations when the channel might be answered that you don’t anticipate (failover to vm etc). Otherwise you could set up inbound routes to ring direct to an extension and the channel probably won’t be answered until the extension is answered.

For outbound you could set up some sip trunks or you could use your analog lines, provided you config DAHDI to check for dialtone before dialing, otherwise you’re sure to seize an in-use line at some point.

You didn’t ask, but I’ll offer an opinion. Using analog POTS lines works, but it’s a PITA. Initial config can be difficult if you have echo that needs tweaking, CID shows up after the first ring so there is delay before the call reaches the extension, outbound calls carry the CID of the channel whether that’s what you want or not. There are use cases where analog is cost effective, but much more often it’s not. The experience all-around is inferior to using pure SIP.

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Okay, makes sense.

I figured I would need to use that ‘pause’ feature. I also know about the outbound feature that checks for a busy line. I think FreePBX gives an example of a fax machine being on the same line?

But about what you say with the inbound CID,

CID shows up after the first ring

that’s not a problem. The intent (once fully deployed) is to use an IVR to filter calls and help with robo calls. I think if I allow the line to ring once before the IVR answers, I should get a CID yes?

As for the outbound,

outbound calls carry the CID of the channel whether that’s what you want or not.

4 lines are in a hunt group on the main number and 4 other lines are dedicated to specific numbers that are used for separate functions of our company. As of right now, our calls outbound do reflect the CID of the line we use, not the primary. So I don’t think that will be an issue but regardless I am going to research how difficult it would be to just hand over our current lines to SIP. My superiors are not fond of that route though because of their reluctance to make phone service internet-connection bound. Not sure why that is, but perhaps they had a bad experience in the past? I’m unsure.

At least with POTS, our phones will remain active if internet fails. I think that’s a safety blanket in their eyes.

That’s fair. In this same situation I once did inbound via DAHDI and outbound via SIP, then told them 6 months later that every outbound call was using SIP. They didn’t notice. You can also get away with fewer analog lines if outbound is not using them.

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