FPBX between T1 and existing Phone system

I am in the process of replacing a Mitel PBX system with a FreePBX system. The main reason for this is to allow multi-site dialing. I have the PBX system, with a Sangoma A104D interface. The current Mitel has 2 T1 interfaces, connected to the provider. To ease the migration, we are going to be running dual systems for a while, so I need to configure the FreePBX box as a man in the middle.

I know I need to configure the Interface for PRI_CPE connected to the provider, and PRI_NET for connecting to the Mitel.

I have set the PRI_CPE (Group1) as from-pstn, and PRI_NET (Group2) as from-internal.

My question is where to go from here for the basic configuration of just “pass everything thru” and “do nothing else”.

All calls coming from the Group 1 should connect to Group2, and vice-versa. This seems like a very simple, basic problem, but for the life of me I cannot figure out what I should change. I would appreciate any ideas.

If you want to start just send your catch all route to the PRI attached to the MITEL.

Is it truly as simple as this:

Create 2 Trunks (Provider Group0 & Mitel Group 1)

Inbound Default - Send all traffic to the trunk for mitel

Outbound Default - Send All traffic to the trunk for provider

Using the contexts as from-pstn & from-internal seem to allow for this minimalist approach, but this really seemed to simple

One more setting I should question: Since this is pri_net, I set the clock to master, but wasn’t sure if I chose the options right. My basic thought was chan 3 source is chan 1, and chan 4 source is chan 2.



The clock setting depends if the MITEL is set to receive clock (as it should be)

The clock on the T1 should clock from the line and the clock to the mitel should send clock.

The T1 should be the master clock

If the Mitel is connected to a working external service you MUST use the Mitel as your clock source and only that, (pri_cpe and pri_net are independent of clock source).

If you reconnect your provider to the Sangoma for pass-through to the Mitel, you MUST use that provider as your clocksource and the Mitel must derive clock from the Sangoma, so make sure if you use the “other span” on the Mitel while experimenting, it is set as a clock source even if number 2 and nothing should be connected to your Mitel clock source 1, the Mitel should recover quite quickly

Actually for testing, if the Mitel is still connected to the PSTN then the Mitel should supply clock towards the Sangoma just for safety.

If not it’s a SLIPery slope.

Surely it would be hard to test the pass-through concept if the Mitel was still connected to the Mitel, no? .

But the underlying arguments remain the same, whoever says them, you should only have ONE clock source and if the “network” is involved, it must come from the “network” or your SLIP’s will lead to a fall. Having two available interfaces on the Mitel allow you to set LBO’s etc. without disturbing the underlying function of the Mitel. You could start by setting up another route in the “Call Routing. ARS” section, perhaps prefixed by an * (see a joke there too) that will send your calls from the Mitel to the Asterisk Box and hence presumably to a SIP carrier,

Replacing any internal Voicemail is harder as you can only turn the lights on and off with an FXS/FXO interface and you will need to use the {E} concept in the dialplan and matching extensions in the FreePBX box when forwarding an unanswered call to an external VMail application, but you will end up with a much nicer Voicemail system.