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Thread: VCF West VoIP BBS setup (ADIT 600)

  1. #1

    Default VCF West VoIP BBS setup (ADIT 600)

    Hi All,
    I saw a pretty nice BBS setup at the VCF West where there were a bunch of systems all connected to an ADIT 600, and especially, the systems were able to accept dial-in. Does anyone have information on that setup, especially ADIT 600 configuration and associated software (e.g. if FreePBX/Asterisk). Many thanks.

  2. #2
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    That was the booth I assisted Lutiana with (The IBM PC Family). The telephony part was mostly my concoction. I will do my best to explain the setup without rambling on infinitely...heh.

    The Adit 600 (also branded as a Cactus Lite) connected via T1 to a small form factor Dell OptiPlex 755 running vanilla Asterisk 16 under CentOS 7. A half-height Rhino R1T1 PCI card in the host provided the physical T1 facilities and the DAHDI channel driver provided the software T1 facilities. DAHDI/Asterisk is what does the bulk of the work and was where all of the "meat" of the configuration took place. The Adit, being a channel bank, takes the channelized T1 (24 channels) and breaks each channel into individual POTS lines (so we had a 24 line capacity at the show. I would've loved to provide service to other booths!).

    The way we achieved access to/from the PSTN was through a Netgear LB1120 LTE modem and an IAX2 trunk. Since one of my nerd rabbit holes is telephony, it is an integral part of my home lab. What I did was set up an IAX2 trunk (fancy connection, basically) between the Asterisk instance at the show and an Asterisk instance at home. Because IAX2 uses the same port for signaling and audio, it traverses NAT very well. Incoming calls would initially route to the Asterisk instance at the house (it was acting as a "tandem" switch), which it would then pass the call on to the Asterisk instance at the show. Outgoing calls would route from the Asterisk instance at the show to the Asterisk instance at the house, then out to the PSTN. Since most cellular carriers do not assign public IP addresses, this was necessary. However, it worked out well!

    The BBS was running on a vanilla 486 DX4-100 that was hiding under the tables along side the Asterisk box. It had 4 internal ISA modems and was running Synchronet BBS under DESQView under MS-DOS 6.22. DESQView was used to support simultaneous connections as it's one connection per "node". DESQview allowed us to run 9 "nodes" on a single host. Although we didn't have the setup to truly demo it, the BBS was equipped to support 4 dial-in users and 4 telnet users (8 instances running under DESQview). The remaining node was for local access on the host itself.

    The configuration on the Adit was pretty simple. Each FXS (phone line) interface gets cross-connected with each T1 channel. There is some basic setup with the T1: picking which side provides the clock source (I chose to have the Asterisk box provide the clock source), making sure the framing (ESF in this case) and line encoding (B8ZS in this case) match on both ends...and that's about it.

    The configuration on the Asterisk side was pretty simple too. A basic build-out for the channels. I made sure call-waiting was disabled, but did leave 3-way calling enabled but I don't think anyone attempted that. Defined extensions for each channel, and then the "default route" when the digits dialed did not match the local dial plan (this then sent calls over the IAX trunk to the Asterisk box at home, which then would send the call to the PSTN if the digits did not match anything in its dial plan).

    I think that's it in a nutshell. Feel free to ask questions. As you can tell, I'm happy to share.

    EDIT: I forgot to include this tidbit. Inbound access from the PSTN was via SIP trunk through DIDs I purchased from Anveo Direct. Outbound access to the PSTN was provided via AT&T POTS lines at the house or via SIP trunk through Anveo Direct if no POTS lines were available.
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  3. #3

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    Was the inbound access from the PSTN via SIP trunk through DIDs the only way?

  4. #4

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    Hi trmg,
    Thanks for the reply, any additional information would be helpful. Right now, my setup is this:

    [QuestBlue]<--internet-->[RPi running FreePBX + Asterisk]<--ethernet-->[Cisco SPA122]<--RJ11-->[Modem]

    My goal is to reproduce a similar setup with the ADIT 600 replacing the Cisco SPA122 so I could use inbound DIDs from QuestBlue for a few systems but also make calls between systems inside my home. I know that you can directly connect a couple of modems and not wait for dialtone but figure this would be a slicker system. Also for what it's worth, my ADIT 600 has the TDM controller, 3x FXS 8B, and 1x router card. My first question is if I can connect FXS ports to Asterisk via the router card or if I need to add a T1 card. Follow up question is if you have some suggestions for configuration for individual lines focusing on data circuits (albeit slow speed). I just got this thing so only partially through the manual. Thanks again.

  5. #5
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    Quote Originally Posted by Morrain View Post
    Was the inbound access from the PSTN via SIP trunk through DIDs the only way?
    The only reasonable way for the show setup, yes. I could've routed one of my AT&T POTS circuits, but 1) I specifically bought numbers local to the Computer History Museum for that extra detail (though not sure many noticed), and 2) using SIP DIDs allowed me to assign every device its own phone number (every device set up had its own DID). If I had used one of my POTS lines, 1) the call would traverse a leg of IP anyway, so some of the benefits of circuit switched goodness are lost there (though not completely), but more importantly 2) it would've only allowed for a single call and said call would've had to be dumped to an IVR or some other fixed destination.


    Quote Originally Posted by breadboard View Post
    Hi trmg,
    Thanks for the reply, any additional information would be helpful. Right now, my setup is this:

    [QuestBlue]<--internet-->[RPi running FreePBX + Asterisk]<--ethernet-->[Cisco SPA122]<--RJ11-->[Modem]

    My goal is to reproduce a similar setup with the ADIT 600 replacing the Cisco SPA122 so I could use inbound DIDs from QuestBlue for a few systems but also make calls between systems inside my home. I know that you can directly connect a couple of modems and not wait for dialtone but figure this would be a slicker system. Also for what it's worth, my ADIT 600 has the TDM controller, 3x FXS 8B, and 1x router card. My first question is if I can connect FXS ports to Asterisk via the router card or if I need to add a T1 card. Follow up question is if you have some suggestions for configuration for individual lines focusing on data circuits (albeit slow speed). I just got this thing so only partially through the manual. Thanks again.
    The best way to approach this is to get a TDM interface between your Asterisk instance and the Adit. The Router card does not have any VoIP capability. You'd need to get a CMG Router card. However, even then, AFAIK it will only do MGCP which can be a pain to do with Asterisk. It *can* work, but if you want to do modem calls with this, your best experience will be to keep as much of the setup as possible circuit switched. So, in the end, getting a T1 card for the Asterisk side of the setup would be ideal.

    The Adit is designed for something to control it whether that be via MGCP using a CMG Router card or via T1 using the TDM controller (which I'm a proponent of the latter). The TDM controller in the Adit has you covered on the channel bank side, you just need T1 on the Asterisk side.

    The first goal would be to work out proper trunking between the Adit and Asterisk. If you go the path of TDM, "Rhino T1", "Digium T1", "Wildcard T1", "Sangoma T1", and "Asterisk T1" are good eBay searches to start with. I would limit the location to North America (especially when searching for "Asterisk T1") as this lessens your chances of getting a knock-off card. Something like theses should do the job:

    https://www.ebay.com/itm/SANGOMA-AFT...D/362764864708
    https://www.ebay.com/itm/New-Openvox...x/283196360280
    https://www.ebay.com/itm/Openvox-D21...X/283197859596

    A Sangoma, or Digium/Wildcard card should work out of the box with DAHDI. The Rhino cards will require you to compile a kernel module, but there are easy to follow instructions for that which I can share if that bridge is crossed. I do not have any experience with OpenVox cards, so I'm not sure if there is native DAHDI support for those or if it requires a separate kernel module.

    I'm not sure what your budget is. If the cards above are beyond it, some patience may save you some money. Also don't be afraid to throw sellers lowball offers when Make Offer is available (you never know what they'll accept!) You could also utilize the "saved searches" feature of eBay to keep an eye out.
    Last edited by trmg; November 19th, 2019 at 10:35 AM. Reason: Some grammar corrections
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  6. #6

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    Hi trmg,
    Heh, I noticed all of the dial in numbers are 650 which was nice. These cards are reasonable but I will also need to find a machine to install it in so this might take me a couple of weeks, I'm likely going to go with SANGOMA because it's PCIe and can go into amd64 or arm system. As for the ADIT, any particular T1 card you recommend? Thank you for the information and identifying cards on eBay.

  7. #7
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    Quote Originally Posted by breadboard View Post
    Hi trmg,
    Heh, I noticed all of the dial in numbers are 650 which was nice. These cards are reasonable but I will also need to find a machine to install it in so this might take me a couple of weeks, I'm likely going to go with SANGOMA because it's PCIe and can go into amd64 or arm system. As for the ADIT, any particular T1 card you recommend? Thank you for the information and identifying cards on eBay.
    You do not need to purchase anything for the Adit. It is already equipped. You will use one of the T1 interfaces found on the back of the unit (the TDM Controller handles those among other things).

    You'll also need a T1 crossover cable. Though, I make T1 crossover "couplers" using two keystone jacks. This way I can use whatever length patch cables are appropriate and if I have to lengthen/shorten the cable between the Adit and whatever host device it's connected to, it's a matter of switching patch cables rather than having to make another custom T1 crossover cable.

    2717-attachment.jpg
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    IBM Model 80 Type 8580-321 - 386DX 20MHz, 4 MB RAM onboard, 4 MB RAM via 80386 Enhanced Memory Expansion, 400 MB Type 0661 HDD, 1.44 MB Floppy, Running MS-DOS 6.22 & Windows 3.11

  8. #8
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    Oh, and not only were they 650 numbers, they were 650-966. That is a local exchange for the area of Mountain View the Computer History Museum is located. A number of thousand blocks have been ported out to other carriers which is how I was able to obtain the numbers I used. Yeah, I'm that much of a nerd...

    https://www.telcodata.us/search-area...0&exchange=966

    https://www.telcodata.us/view-switch...li=MTVWCA11DS1
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    IBM Model 80 Type 8580-321 - 386DX 20MHz, 4 MB RAM onboard, 4 MB RAM via 80386 Enhanced Memory Expansion, 400 MB Type 0661 HDD, 1.44 MB Floppy, Running MS-DOS 6.22 & Windows 3.11

  9. #9
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    There's a version of the ADIT router card that does do SIP (or so I've read, haven't pulled the trigger on an ADIT 600 yet), but I agree you'll be better off with a T1/E1 on the Asterisk box. Personally I wouldn't bother with anything but a Digium card.

  10. #10

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    Hi All,
    I went ahead with Sangoma card because it was PCIe and made the cross over cable. I'm going to build Asterisk and FreePBX tonight/tomorrow and will report back. Planning to follow along here unless folks have other suggestions. Thanks.

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