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 iâll 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 âgroup idâ for each PRI interface, and created 2 trunks of type DAHDI. I expect that iâll need to configure some incoming/outgoing rules.
Clearly I need to study more the topic before becoming able to know what am i doingâŚ
Please share any thoughts about this configuration : does it seem feasible to use FreePBX for this setup ?
Donât push your luck trying to debug two complex interfaces at the same time. I recommend:
Create a pjsip extension and set up a softphone to use it. Call *43 (echo test) to confirm that it works.
Set up a default Inbound Route (DID Number and CID Number left at ANY) with the extension as destination.
Make a call from the âcallerâ 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.
Once (3) is working, set up an Outbound Route that recognizes a number on the âreceiverâ device and sends it to the receiverâs trunk. Call that number from the extension. If it doesnât work, post Asterisk log (if present) or device log.
Set up call forwarding (or Follow Me) for the extension to ring the receiverâs number. If no luck, post logs.
Iâll 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 havenât 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 ).
If you wanât to âclever as shitâ and you have dahdi installed on both ends, then you can have dahdi âcross-connectâ 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 DACSâed 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.
Sorry, you have moved from a very low level request for connectivity how-to to a very high level question about FreePBXâ function. For me , too much of a disconnect.
My answer was âprobably not in any way easilyâ as the DAHDI interface available to FreePBX is not that clever, but it could be done with the raw abilities of dahdi
Is very complicated question to get a simple answer, because part of ISDN bearer capabilities and Information Transfer Capability, as on post it mention ââŚneeds to support multiple channels in parallel (data channels)â 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 âspeech unrestricted data, restricted data⌠etcâ structure â8 kHz integrityâ or packet âUnrestricted dataâ 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.