Background
In the fall of 2004, an email conversation began on the
Telephone Collectors International
mailing list.
Several folks who had restored older PBXes and CO switching systems had begun to wonder, now that Voice Over IP was beginning to be viable, if there was a way to connect their switches together via the internet.
Someone mentioned the Asterisk VoIP PBX , and, with a little experimentation, the project took off, with several of the switchers creating Asterisk switches as tandems to their switches. A central switching scheme was established, office codes were assigned, and an automated method of looking up call routing was put in place. The Collectors' Net , or C*NET , was born.
Since then, we've built a private network that includes more than just electromechanical switches connected to Asterisk PBXes. So far, we have older PBXes and CO switches, key systems of various sorts, and individual telephones.
These are connected not only by Asterisk VoIP PBXes, but also by other VoIP PBXes, Analog Terminal Adapters, VoIP telephones, and soft phones. There has also been some successful experimentation involving embedding Asterisk, pre-configured, into a standard residential firewall/DSL Modem.
We have our own mailing list , where we can cuss and discuss challenges and solutions, as well as pretty much anything telephony or VOIP related. The discussions so far have produced a wealth of information, available in the mailing list archives . You'll need to register with us in order to view the archives.
How It Works
There are various ways of connecting to the
C*NET
.
Here are some:
- A single telephone
- A softphone on a PC
- A Key system
- A PBX
- A Central Office Switch
Except for the soft phone, all connection types require a piece of hardware at your location. Here are two different call flow sequences, based on the most common types of connection methods: Asterisk PBXes and Analog Terminal Adapters (aka ATAs).
Typical Call Flow for the "Asterisk Tandem" means of connection to the
C*NET
- Telephone comes off hook at the originating end, and receives dial tone from the legacy PBX.
- User dials 8, and receives dial tone from the Asterisk tandem switch
- User dials the country code and number of a telephone at the terminating end.
- Asterisk switch makes DNS query to see how to route the call.
- DNS Server responds with routing information.
- Originating Tandem contacts the Terminating Tandem via an IP connection.
- Receiving tandem looks in its own routing tables and determines that the call should be passed via an FX connection to the terminating legacy PBX.
- Terminating PBX, supplying station dial tone or direct trunk connection, accepts digits from Terminating Tandem.
- Station at terminating end rings.
A simple graphical representation of this call flow is available by clicking here.
Typical Call Flow for the "Analog Telephone Adapter" means of connection to the
C*NET
- Telephone comes off hook at the originating end, and receives dial tone from the legacy PBX.
- User dials 8, and receives dial tone from the Analog Telephone Adapter (ATA).
- User dials the number of a telephone at the terminating end.
- ATA calls its host Asterisk Tandem, and sends the dialed digits to it.
- Host Asterisk Tandem makes DNS query to see how to route the call.
- DNS Server responds with routing information.
- Host (Originating) Tandem contacts the Terminating Tandem via an IP connection.
- Terminating tandem looks in its own routing tables and determines that the call should be passed via an FX connection to the terminating legacy PBX.
- Terminating PBX, supplying station dial tone or direct trunk connection, accepts digits from Terminating Tandem.
- Station at terminating end rings.
A simple graphical representation of this call flow is available by clicking here.
Which Connection Method Should You Use?
Let's Compare
- Using a PC to create a tandem of your own:
-
Pretty much any old PC will do, as long as it is modern enough to support the FXS/FXO card(s) that you choose.
Also, any
Dahdi-compliant PC telephony FXO and FXS cards will work -- one of each.
These cards are available, brand new, from
Digium
, the folks who wrote the Asterisk program.
Used Digium cards and brand new Zapata-compliant clones are available on eBay, in case you're the type who doesn't feel the need for tech support from the hardware manufacturer.
From the software and operating system perspective, it's all free.
Linux and
Asterisk
are the two necessary elements here.
A
detailed tutorial
will get a generic VoIP switch up and running for you, and the
sample configurations on our site
will give you a template to use to configure your switch to join the Collectors' Net.
- Using an Analog Terminal Adapter to home back to the ckts.info tandem, or someone else's PC:
- This method invovles the use of a combination FXO/FXS device, also known as an Analog Terminal Adapter or ATA. Once your ATA is configured, your legacy switch will make outgoing calls through the FXS portion of the device, and receives calls on the FXO portion. The FXO/FXS device is programmed to connect through a host tandem (Asterisk PBX) at ckts.info or another switcher's site.
Here we get into the nuts and bolts of getting your legacy PBX (or CO Switch) onto the
C*NET
network.
You've probably decided by now which connection method will work best for you.
Now is when you make your choice.
- Choose this link for instructions on how to install Asterisk.
- Or Choose this link for instructions on installing an ATA.
- Or choose this link for instructions on installing the nuts-to-bolts "Asterisk@Home" method.
Okay, at this point you want to be able to receive incoming calls on the C*NET . For North American folk, here is the dialing scheme:
- The Country Code plus
- The Office Code plus
- The 'line' digits .
UK switchers should refer to the UK Policy White Paper. If you live elsewhere, email the System Administrator for guidance.
In the North American C*NET scheme, the country code is the digit '1,' the next three digits are the office code, and the last four digits are the 'line' digits. This gives North American switchers a "1+7-digits" dialing pattern. (We don't use area codes... yet). The line digits are switchable at the collector's site; e.g., extensions, milliwatt, silent term, time-of-day, redirects to cell phones, etc.
Similar structures exist in other countries as well, with the first few digits comprising the 'city code' and 'exchange code", and the trailing digits switched in the individual PBX, CO switch, or key system.
These are the steps you need to be assigned an Office Code:
- First, decide if you actually need an entire office code.
- If you just have a single phone (or 2 or 3) to hook to the network, there are numbers available to you on an individual line basis. In North America, these numbers exist in the 999 office code. If you just want to hook up a line or two, this is the easiest step for you to take. Contact the System Administrator with your request.
- If you will be needing several lines, but not entire block of 10,000, you should request an office code and "thousands group" for your switch.
- If you have a larger switch, or one that uses numbers from all over an office code, you can request an entire NNX.
- Check for availability at the Collectors' Net Reservations Page.
- Register yourself as a member of the group. You'll need to do this in order to place a reservation for an office code, among other things. This is how we tie it all together. The username you choose will link your office code to your directory entries to your switch to yourself. It's the glue behind the rest of it. (Just so you know, we do NOT share any of your contact information with anyone else, even other telecom-related sites.)
- Register and Activate your Office Code in our ENUM (DNS) registry. It's a manual process, though. Once you register, it will take an actual human to put it into the system.
Testing Your Installation
At this point, you've assembled your Asterisk box and programmed it, or you've obtained an ATA and programmed it. Now, it's time to test it. Here's how to test your connectivity on the net.
- Test your connectivity . Look up a test number , and call it. Verify that you get what it says that you should.
- Look someone up in our directory , and call them. Ask them to call you back. ***OR*** Call yourself through one of our PSTN gateways to verify your connectivity.
Additional Resources / External Links
These are a few of the outside sources we have found helpful. If you stumble across another one that is not listed here, please email us with that information, and we'll add it to this list.
Helpful Links
Asterisk.ORG . | - - | The Asterisk website. Includes hardware compatibility list as well as downloads of Asterisk |
---|---|---|
Digium.COM . | - - | The folks who wrote Asterisk. They gave Asterisk away in hopes that you would buy their FXS, FXO, and T1 PC plug-in cards. |
Guide To Asterisk | - - | An excellent step-by-step tutorial on getting Asterisk up and running. Add the C*NET configs, and you're off and running. |
Max's Hacks | - - | Max's hardware schematics and Asterisk (zaptel) hacks to make it all work. |
VoIP-Info.ORG | - - | A TIKI on VoIP in General, but with emphasis on Asterisk |
Zapata Telephony | - - | The authors of the "Zapata" voice-to-PC drivers, used in Asterisk. |
C*NET is a project of participating members of TCI, ATCA, the UK's Telecom Heritage Group, and interested others.
Voice Over IP Tandem for Analog Switches 172.104.9.200