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Thread: Voip between two offices

  1. #31
    Super Grandmaster Madman88's Avatar
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    Quote Originally Posted by leonb View Post
    I still cannot get my head around this.

    If both offices is on the same lan, and I have one voip gateway in each office, why is SIP needed/relevant? Both gateways has got an IP address, so why cant they communicate directly, and therefore sens the voip over the lan?
    Ok, I see why youre confused. Sip is a protocol. Its used on an outbound trunk (Sip trunk), but its also used for the extensions on the PBX (Sip extensions). When you have an extension running over the lan you would be using the Sip protocol for that extension. No Internet or trunk required for that.

    Quote Originally Posted by Superspeed View Post
    You don't need a sip account with any provider. Simply download the latest build of Elastix from elastix.org. Install it on an old PC with 1Gb of ram. Then setup the side b as an an extension(s). On side a you setup your outward trunk as an extension on your analogue PBX. Bob's your uncle. This will cost you - nothing. As long as you have an old PC with a network port. Install time - less than an hour.

    If you need help - holler
    I think you would at least need an internal voice modem to plug it directly into the analogue box, but otherwise if you're willing to do the install that will work fine.
    I am 13531

    "Balance is the key to everything, without it we would just keep falling over."

  2. #32

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    Quote Originally Posted by paulcolmer View Post
    OK my friend this is how you do this if you want the calls to go PBX to PBX

    1/ VoIP enable your PBX s @ 13k as quoted

    IF you have spare extensions or trunk ports on the PBXs you may look at this as well, if you dont have you can always add.

    2/ Put an FXS or FXO gateway at each end and get a VoIP account with a provider that allows free on net calls and rout via DSL or whatever.

    3/ / Put an FXS or FXO gateway at each end and build an Asterisk box one end to drive them, Digium or Sangoma cards for the Asterisk box will be more expensive than a gateway, so best go the gateway route.

    4/ Put an FXS or FXO gateway at each end and load a free soft switch like 3CX, Snom 1, BizPBX ect on a Windows PC to drive the gateways, these free edition soft switches will allow you between 2 & 4 concurrent calls depending on which one you choose, before you have to shell out for the paid versions. The lack of g729 codec is not an issue because you have so much available bandwidth on your link so g711 is ok.

    5/ Put a gateway one end and cheap Yeastar PBX the other, the Yeastar can talk to the PBX at its own end and the gateway at the other.

    These are your options if you want to make your calls from your existing extensions, other options are of course using dedicated IP phones or soft phones at each end.

    So some food for thought, I can give you more detail on each solution if required.
    Can I use two fxs ata, or does one need to be a fxo? I have a spare extension on both side and was thinking of putting a fxs ata on both. My pbx tech says we need a fxo on one of the sides. He will then program them to connect directly without need for sip account. But why a fxo?

  3. #33

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    anyone used dundi before? going to be interconnecting about 20 sites together and was wondering if this was worth my time.
    (and yes, all sites have leased lines and more then enough bandwidth)

  4. #34
    Super Grandmaster Madman88's Avatar
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    Quote Originally Posted by avert View Post
    anyone used dundi before? going to be interconnecting about 20 sites together and was wondering if this was worth my time.
    (and yes, all sites have leased lines and more then enough bandwidth)
    is dundi not kinda dead now?
    I am 13531

    "Balance is the key to everything, without it we would just keep falling over."

  5. #35

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    no clue, but if thats the general feel. i wont use it ^.^

  6. #36
    Super Grandmaster Madman88's Avatar
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    I never actually used Dundi, but in my understating it was a form of asterisk that used peer to peer protocols to control its sip extensions.

    I would suggest some kind of normal asterisk setup for something that large. Have a box at each site and then just link them to each other.

    You could probably run the whole shebang from a central location, but then if a line when down so would the phones.
    I am 13531

    "Balance is the key to everything, without it we would just keep falling over."

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