Using FreePBX for routing calls from one PRI to another PRI


I am new to PABX and telephony so I feel overwhelmed when seeing all the terms and acronims. It would be great to have an initial confirmation that FreePBX is a viable solution for what I am trying to achieve.

The context is probably non-typical : for testing lab purposes , i need to connect 2 devices that have isdn PRI interfaces. The first device is always the caller, and the second device is always the receiver. So the goal is to connect them to some kind of PABX that redirects the connection from caller to the receiver. This needs to support multiple channels in parallel (data channels). The receiver will have a single phone number that will be used by all channels.

In future i’ll want to extend this to have 1 caller and 2 receivers. So depending on the phone number the call would be redirected to one receiver or the other.

I have already installed FreePBX on a pc that has 2 dual Digium PRI adapters ( so in total there are 4 PRI connectors that are recognized ok - they appear in the DAHDI configuration ). However am not sure what exactly do I need to configure next. So far I have set a different “group id” for each PRI interface, and created 2 trunks of type DAHDI. I expect that i’ll need to configure some incoming/outgoing rules.
Clearly I need to study more the topic before becoming able to know what am i doing…

Please share any thoughts about this configuration : does it seem feasible to use FreePBX for this setup ?

Thank you !

Let’s go step by step here.

Can you confirm that the signalling on your T1/E1 is PRI ?

Yes, i am sure that it is isdn PRI t1/e1.

A) E1 or T1 ?

On what hardware does it originate ?

Do you have both ends directly connected over a properly arranged RJ-48 if T1 or presumably coax if E1 ?

Don’t push your luck trying to debug two complex interfaces at the same time. I recommend:

  1. Create a pjsip extension and set up a softphone to use it. Call *43 (echo test) to confirm that it works.
  2. Set up a default Inbound Route (DID Number and CID Number left at ANY) with the extension as destination.
  3. Make a call from the ‘caller’ device. The softphone should ring and you should be able to talk between them. If not, if anything appears in the Asterisk log for the attempted call, post that. Otherwise post any errors logged by the calling device.
  4. Once (3) is working, set up an Outbound Route that recognizes a number on the ‘receiver’ device and sends it to the receiver’s trunk. Call that number from the extension. If it doesn’t work, post Asterisk log (if present) or device log.
  5. Set up call forwarding (or Follow Me) for the extension to ring the receiver’s number. If no luck, post logs.
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I’ll need to do some research to be sure and clarify all the correct parameters of the PRI interfaces. I expect it is E1 (Europe).

The two devices are Multidata Hermes Pro/P1 routers.

I haven’t connected them yet to the FreePBX . The routers are still connected to each other with a direct cable that looks similar to a RJ45 but has a different wiring ( isdn S2M wiring ).

Before we do that, is it ok if we first find out if he is even using FreePBX?

Why do you question that?

Because he said " I haven’t connected them yet to the FreePBX"

If you wan’t to ‘clever as shit’ and you have dahdi installed on both ends, then you can have dahdi ‘cross-connect’ using DACS all 30 channels between the local and the tie spans at both ends , the overlayed PRI driver will then transparently pass the originating PRI so it will appear at the other end on the DACS’ed span. This solution does not involve asterisk of FreePBX though (yet)

Sure, but I assume that the OP knows to physically connect the devices with suitable cables. He may need crossover cables if neither end supports switching between NT and TE.

I assume that he wants FreePBX in the path to eventually choose a destination based to the called number.

I assume so also, I offer a path to a solution to his request , you offer an off topic alternative, not that I will ultimately disagree with you.

I need that the caller really calls a phone number. For example this allows testing what happens if caller uses a wrong number.

By the way : does FreePBX have the possibility to kill an active call from the web interface ?

Sorry, you have moved from a very low level request for connectivity how-to to a very high level question about FreePBX’ function. For me , too much of a disconnect.

Just go with two E1 gateways and SIP.

My initial question was just : can it be done with FreePBX ?

My answer was ‘probably not in any way easily’ as the DAHDI interface available to FreePBX is not that clever, but it could be done with the raw abilities of dahdi

Are you suggesting that a RJ45 crossover cable would universally work for RJ48 or that we can assumed the OP knows the difference?

An Ethernet crossover cable won’t work – it crosses pins 1&2 with 3&6. I believe that S2M uses pins 4&5 for one pair and 3&6 for the other.

That’s better known and well recognized as an RJ48 in ‘Registered Jack land’ , particularly by us phone folks

Is very complicated question to get a simple answer, because part of ISDN bearer capabilities and Information Transfer Capability, as on post it mention ”…needs to support multiple channels in parallel (data channels)” if all involved components it could manager and handler.
First of all should check what kind of setup message element will send caller device, if is circuit (information transfer “speech unrestricted data, restricted data… etc” structure “8 kHz integrity” or packet “Unrestricted data” Service data unit integrity.
Afterward could go with others devices as Digium card which main is develop for circuit mode and so on, I guess probably it could resolver bearer capability on setup message element but it will handler as a unique voice channel.