TalkSwitch 284vs

I have TalkSwitch 284vs. It's nice unit with 2 analog and 4 VoIP trunks and 8 anolog and 16 VoIP extensions.

I want to register PBXes extension as trunk on TalkSwitch. Problem is that TalkSwitch needs numeric UserID that is same as phone number of trunk. Auth ID can contain letters and numbers but UserID can't.
PBXes has UserID in format User-Ext and SIP registration fails.

Does anybody know solution for this problem?
Can I create PBXes extension that has only numbers in UserID?

Sergey.

RE: TalkSwitch 284vs

Hello Sergey,

I might not have the solution to this issue, and I doubt the PBXes code can be modified to accommodate the expected credentials of the TalkSwitch, but I have a more general question regarding what you are trying to achieve.

As far as I understand, you are using the TalkSwitch as a on-premise PBX platform, in contrast with PBXes which is an off-premise PBX platform. My question is, why do you wish to register a PBXes extension as trunk on TalkSwitch? I would understand if you were trying to register a PBXes trunk to the TalkSwitch, in case you wished to receive and send calls through it, to the local land-lines attached to it, but I fail to see the reason for the reverse.

RE: TalkSwitch 284vs

There are 2 issues I want to address:

1. If my DSL is down I want to route incoming VoIP calls to land line. My main DID is VoIP line. I have 1 landline for fax machine.

2. If my main VoIP provider is down (it happens sometimes) I want to route calls from TalkSwitch to different provider automatically. PBXes has this functionality but TalkSwitch doesn't.

Sergey

RE: TalkSwitch 284vs

Hey Sergey,

Let me start with a question: Are all your extensions in one location, or in multiple locations?

Regarding your first requirement, if your DID is registered as a trunk on PBXes, inbound calls to it can be routed to your land-line via another SIP trunk. Depending on how your Inbound routes are setup, inbound calls can normally ring on one or more SIP UAs and when in fail-over mode ring on your land-line.

The above scenario can be implemented without the direct involvement of your TalkSwitch, if and only if your DID provider supports concurrent SIP registrations. If they do, the incoming calls will be send concurrently to both PBXes and the TalkSwitch (SIP Forking). Under normal circumstances, calls will be answered by the extensions registered to the TalkSwitch, but when it is no longer reachable, the calls will be delivered to your land-line via PBXes.

If the above feature is not supported by your DID provider, you might be able to try a few other techniques, if they are supported by the TalkSwitch.

Regarding your second requirement, it really depends on the SIP User Agents (soft-phones, ATAs, IP-phones) you are currently using. If they support multiple registrations, routing calls via multiple VoIP providers though PBXes is trivial. If they don't, things get more complicated, but there should be a way to achieve that, if you are willing to alter your setup.

Have a good day.