3com VCX V7111 - Surely it can't be this hard to connect to freepbx

Is there anybody out there using a 3com vcx v7111 with freepbx and got it working? Did you specify freepbx as a proxy within the 3com software setup?
Am I wasting my time with a 3com box that isn’t going to work???

I’ve no experience with phones voip etc but can rtfm if I could find one to copy an example from…

I’m just wanting to get inbound + outbound calls working from my pstn analogue line to my freepbx server for a demo.

I have got a 3com vcx v7111 2 port FXS/FXO adapter to connect the analogue line to my network.

I have some SIP softphones on laptops around the network.

I have a trunk and inbound/outbound routes defined in freepbx but cannot get calls in or out.

For example - dialing in from an external line…
the 3com box answers after two rings and i can then enter a number eg ext 1012 - despite the 3com box being set to single stage dial?? I want it to just route any call through to the 7777 IVR. However, it stops and lets me enter an extension.
the 3com log says its trying to find [email protected] - which it doesnt find because my 1012 extension is on a laptop with dynamic IP
the is the freepbx box where the sip extensions are defined. When I dial 7777 from a softphone i get the IVR. How do I get the IVR from an external call?

Outbound… I’ll worry about that later…

well it was that hard, the GXW4108 was working pretty much straight away.

I’ve some new issues now that it’s working though :sunglasses:

Thanks Hint, I’ve previously found a similar document for an asterisk config with the GXW410x and tried the example with no success.
I’ll have another look with the tribox doc to see if any bells are rung.
I’ll have another look tomorrow and see what’s getting populated in the .conf files and also if there is anything in the /etc/asterisk/log/

I’m leaning toward thinking my issue is with the 3com box config not asterisk.

well I had some spare time for another play with this and still can’t get it route inbound or outbound calls.

I’m giving up on it and getting a GXW410x to play with… oooo ready for egg on my face if I still can’t get it to work.

Not Familiar with the CISCO, but with gateways of this type, you’ll probably have to set up a SIP trunk between the CISCO and the Asterisk Box, then set up at least one inbound route pointing to the IVR you want to access. Then you’ll set up outbound routes to allow for outbound calls.

Sometimes you have to find something close and extrapolate from there.


Trixbox uses a derivitive of FreePbx, and maybe you can get some words of wisdom from the GS docs.

I know this is old and I’m new here, but for anyone coming here for google, this is a rebadged audiocodes mp-1000 box. So an audiocodes guide would apply here.

Here’s the user manual for the audicodes mediant 1000 (3com v7111) box. It goes indepth into all the options. The reason there’s so many options is for worldwide compatibility. Certain ATA’s with fewer options may not work in other countries, afaik.

Sorry, here’s the link.