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  • Astrisk Confrence Bridge

    I am too cheap to buy a Shoretel conference bridge. I am beginning to tinker with making a conference bridge with a beige box Asterisk server. I am planning on connecting them using SIP trunks. I would use DNIS to map a few numbers to point to the Asterisk box. The Asterisk box would take all incoming SIP calls and point them to the built in conferencing solution.

    What we are trying to accomplish is the ability of Conference phone to call into the Asterisk box and input a leader pin code. All other users could call our phone system on these designated numbers and they would be asked for a conference PIN. Based on the PINs entered would be the room they were added to.

    How outlandish does this idea sound? Has anyone tried this yet? Any advice or tips before I proceed?

    Shoretel 7.5 Build 12.15.1800.0
    AsteriskNow 1.0.2

  • #2
    Trixbox conference bridge

    I am doing it and it has been working for about 6 months now. I have setup 2 extensions going into the conference bridge (so we can have 2 conferences at one time) and 1 DID. We have 10 sip licenses so we can have up to 10 people in a conference at 1 time. I used Trixbox and it works, no frills like the shoretel has but if all you need is a conference bridge to dial into its not shabby. The call quality is good (we have the ip8000 in our conference room) and the learning curve is steep and I built several before i got it right (the vmware appliance version of Trixbox is a good place to start testing.) I will make another post with the sip trunk config for the trixbox which will save you a ton of time.

    Regards,

    Jackl

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    • #3
      I'm doing it, too

      I've been running the conference bridge on Trixbox over sip trunks from ShoreTel for some time now and it works well for us also.

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      • #4
        You guys have been anointed to lead me on the path to righteousness.

        OK Seriously, any help would be appreciated. I will look into a Trixbox VMWare Appliance and begin testing from there.

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        • #5
          Quick question for Jackl, what kind of hardware are we talking for a 10-12 user confrence, same idea two extensions total of 12 parties?

          How much CPU and RAM should I expect to use?

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          • #6
            Trixbox trunk config

            Outgoing settings

            type=peer
            qualify=yes
            port=5060
            host=192.168.16.4
            context=from-internal
            canreinvite=no


            Incoming Settings

            type=user
            port=5060
            host=192.168.16.4
            context=from-internal


            the host in the config is the shoretel 40/60 switch that you setup the sip trunks that point back to the Trixbox.

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            • #7
              I am running mine on an old hp box that was running xp with 512 mb ram.

              Here is some reading material to help with trixbox
              Trixbox without tears (pick and choose conferencing stuff)

              Good readme on Web Meetme 3
              Last edited by jackl; 09-02-2008, 04:07 PM.

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              • #8
                We have a trixbox setup with SIP trunks on both ends. I can see that they are registering with each other.

                I have setup with DNIS to forward to an off system extension and forwarded all calls for a DID to that Extension. The extension is setup on the SIP Trunk Group. I get a reorder tone when dialing the extension directly or from an outside line.

                I am a little confused as to what I need to do with the Shoretel side to gets the calls sent to the Trixbox.
                Last edited by Crewdawg; 09-04-2008, 09:17 AM. Reason: Clarity

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                • #9
                  The shoretel part is probably the easiest, I bet you are getting the reorder tone from the trixbox. Take a look at your inbound routes in the free pbx web interface, I remember having the same problem until I setup a route for the extension to go the the conference i setup. Hope that works.

                  Jackl
                  Last edited by jackl; 09-04-2008, 01:53 PM.

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                  • #10
                    It is up and working now.

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