FXO VoIP gateways and 3CX and Meridian

Discussion in '3CX Phone System - General' started by RooDBwoY, Feb 28, 2013.

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  1. RooDBwoY

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    I have a quick query regarding hashing my 3CX test environment into our existing traditional Meridian for a couple of extensions and understand from reading the install and config guide that using a VoIP gateway to connect to some analogue phone lines may be the way forward.

    Basically, what I have achieved so far is a completely standalone 3CX implementation with 3 Cisco SPA504G handsets and no external calling at all (nicely configured by the templating system, which I have customised) and works perfectly. What I want to achieve now is to configure some analogue extensions on our existing (live, firmwide) Meridian PBX and gateway them through to some handsets on the 3CX system to utilise the Cisco handsets and test system in a 'sandboxed' environment, which will obviously allow me to route calls from the Cisco handsets out through the Meridian on our 2 x ISDN PRI lines.

    I need to buy a cheap VoIP Gateway for this, with presumably FXO ports (rather than FXS) to allow the Meridian analogue extensions to get routed into the 3CX system and then configure in-bound and outbound rules, which I'll figure out later on. What I need to know is a. whether anyone has done similar before and b. that it is definitely FXO ports I'll need, and not FXS.

    My understanding is that FXS ports would be used to plug phone handsets into but FXO ports can be used to present between the two PBX's? Also, does anyone know if the CLI will be presented from the analogue lines through the SIP traffic to the Cisco handsets? I don't actually have any analogue handsets with that feature on them to be able to test (most of our Meridian handsets are either digital or very simple analogue handsets!) beforehand...

    Cheers in advance folks!

    Awesome product by the way. Very handy for learning VoIP/SIP/RTP/etc.
     
  2. leejor

    leejor Well-Known Member

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    Unless the "other" PBX supports SIP trunking, in which case a bridged trunk may work, you are left with two methods...

    3CX->trunk->PSTN Gateway (FXO)->analogue extension of other PBX

    Calls from the other PBX to the extension number would pass to the trunk on 3CX, they will show CID if sent. Calls from 3CX will show on the other PBX as a call from the extension that 3CX connects through.

    or

    PSTN (analogue) trunk from other PBX -> ATA (FXS)-> registered as extension on 3CX

    Calls from the other PBX to 3CX will show as a call from the 3CX extension the ATA is registered as. calls from 3CX should show caller ID from the 3CX extension.
     
  3. RooDBwoY

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    leejor, thanks for that info. It was as I suspected, so thanks for confirming. I'm going to test the analogue CLI/CID today as I've brought an old phone in that supports calling identifier so will be a reasonable test.

    Looks like I'll be looking for a VoIP Gateway that has some FXO ports on it...
     
  4. leejor

    leejor Well-Known Member

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    If you plan on using a gateway to connect with analogue extensions from the Meridian, something to keep in mind. Many PBX's only supply 24 VDC on the phone line, in an idle state. As most gateways are expecting 48 to 51 VDC, as a default, the lower voltage may be seen as "line in use". This would require the "adjustment' (lowering) of the Line in use voltage setting, on the gateway.
     
  5. RooDBwoY

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    Ah, useful tip.

    Cheers, I'll bear it in mind!
     
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