Before I get started - a few questions

Discussion in '3CX Phone System - General' started by mthomas, May 2, 2011.

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  1. mthomas

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    I was looking to get started with 3CX , as it looks to be a great system. However I have a few questions.

    I have 2 analog phone lines, and was looking to buy a router, with 2 FXS analog ports. Does this mean I can plug my 2 phone lines into the ports, and configure 3CX to use the router as a FXO gateway.

    Then provision my SIP phone, to the 3CX server, and as such, the SIP phone will have access to the 2 analog lines?

    My goal is to have 3CX configured with a few SIP providers and still retain access to my analog lines via my SIP phone, without spending a large amount of money.

    I've been unable to find a SIP phone that is combination analog + VoIP... I'm pretty sure they don't make them , correct me if I'm wrong...

    If anyone has any advice on how to go about doing this.. .I'm open to suggestions. :) Thanks so much ...
     
  2. leejor

    leejor Well-Known Member

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    No, phone lines plug into FXO ports (a Gateway), FXS ports go to a telephone. Don't expect to be able to use the two FXS ports with 3CX behind that router, those ports are meant to be accessed from the WAN side, not the LAN side.

    I believe there are (is) a Siemens cordless phone that does both, but you are probably better off (cost wise) with an ATA (FXS ports(s) ) and a standard phone.

    Costs= PC, router (if you don't have one), perhaps a switch, ATA(s) with a port for each telephone & gatewway(s) with enough FXO ports (one for each PSTN phone line)
     
  3. mthomas

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    This is the router http://www.draytek.com/user/PdInfoDetail.php?Id=114

    that I was thinking of using . It has 2 FXS Ports. So I imagine they are only configurable from the web interface of the router, not 3CX.

    This ATA has 2 analog lines.. comes up time and again : http://www.voipsupply.com/linksys-pap2t-na

    Could I plug my analog lines into that, then into the switch, and add them into 3CX ?

    Appreciate the help .

    I'd like to use this phone: http://www.snom.com/en/products/ip-phones/snom-870/
     
  4. smb1

    smb1 New Member

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    You should be fine with the Draytek router.

    For the PSTN gateway and handsets, I would go with hardware supported by 3CX, which with 3CX Version 10 neither the PAP2T nor the SNOM are.

    I would recommend the Patton 4112 for the FXO and either Grandstream, Yealink, Cisco or Polycom phones.

    3CX has built in support for these devices. 3CX has supported SNOM phones, but will not be doing so in future.
     
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  5. leejor

    leejor Well-Known Member

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    This is an ATA (Analogue Telephone Adapter) , you can't plug lines into it, you can plug telephones into it. it will work with 3CX, it is a basic ATA. Linksys also makes the SPA2102 which has 2 ports for phones (and a built in router), it has a few additional features over and above the PAP2T, but it may not be worth the additional cost for your needs. They also make the SPA3102, which has one port for a phone and one port for a telephone line. It is both a Gateway and an ATA. Other companies such as Grandsteam make similar devices.

    You require a Gateway if you wish to use PSTN telephone Lines with 3CX.
     
  6. mthomas

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    Ok. Sorry for the newbie questions.

    So the ATA won't work... and the Cisco/Linksys one recommended a post above me... I'm not a fan of Linksys products - I don't allow them in my home :)

    As far as the Draytek router... What would I need in regards to a gateway (FXO Gateway) to get it to function? The router would also do this on it's on (since it has SIP functionality built in, without 3CX of course) or - the Patton 4112 that was mentioned.

    I get confused... can't I plug my PSTN lines directly into the FXO gateway? Isn't that the point?
     
  7. mthomas

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    So after speaking with VoIPSupply.com , they advised me, I can purchase the Patton 4112, and connect my 2 PSTN lines, then configure that with 3CX.

    Then configure my SIP phone. The SIP phone will be able to place/receive calls using those 2 lines, as well as utilize any additional lines, using SIP accounts configured in 3CX console.

    Correct? :)
     
  8. leejor

    leejor Well-Known Member

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    That's pretty much it. Any phone you wish to use will have to be a SIP phone or a standard analogue phone in conjunction with an ATA. You can also use a softphone run on another PC.

    You have obviously had a horrible experience in the past, that has turned you against Linksys products. I, personally have never had any issues/failures with them. I do understand that they can be a bit daunting to the the uninitiated (especially the 3102 due to the overwhelming options), but once configured correctly, they perform (for me anyway) just fine.

    Many VoIP providers rely on their ATA's for residential service.
     
  9. mthomas

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    Well, will the SPA2102 take the place of the Patton device ? It's considerably cheaper :)

    Also the free version of 3CX only supports 2 simultaneous lines right ?
     
  10. mthomas

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    I've also come across the Zoom 5801 VoIP Gateway ..... which is considerably cheaper than the Patton 4112 ... would I be able to interface this with 3CX ?

    Sorry for the n00bish questions... I certainly appreciate the help .
     
  11. smb1

    smb1 New Member

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    You can have a try with the 2102, but it is not a supported phone. This means that 3CX does not provide documentation or automated configuration for that device.

    You can have a look at http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE and see if you can adapt the instructions for the 2102.

    The same would apply for the Zoom.
     
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  12. mthomas

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    Thank you ! So much for the answers to my questions :)
     
  13. leejor

    leejor Well-Known Member

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    If you've applied the Demo code, otherwise it is four concurrent calls (not lines).

    The 2102 is a very good ATA. Basic settings are pretty simple. It does have a built in router which most users will want to disable and just bridge the Ethernet ports to allow daisy-chaining of units. There are a lot of options, and it can be a bit overwhelming if you haven't configured one before, but, if you are happy with the tones and ringing cadence then there are just a few things that really need to be changed to get it working. A lot of other things can be changed later, after you get basic calls working and you become more familiar with the unit.
     
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