Frequently Asked Questions

The Collectors' Network

Q: What if I don't have a 'system' to hook up. How can I attach a regular phone to the network?
A: You can use the main tandem to host you, or one of our members could volunteer to do the same. Just pose the question to the mailing list after you join it. You will need either: (1) a telephone and analog terminal adapter (ATA), or (2) a voip-capable phone, or (3) a softphone installed on your PC.

Q: My step switch has three-digit dialing so I don't know if that's something that can be configured around in the software or if I have to put in digit-absorbing selectors in my switch.
A: Asterisk can dial into your *PBX* with any number of digits you like. For example, if one of your C*NET-assigned phone numbers is 999-4321, you could configure Asterisk to send just "321" to your step switch.

Q: My step switch won't know that it needs to access the trunks going to the Asterisk switch until after my internal user dials "1." How will the Asterisk box deal with the first digit being missing?
A: One of the beauties of a switch like asterisk is that they can dial out whatever you'd like based on what you dial in, and it doesn't have to be the same thing. For example, when I dial "123" into my asterisk box from an attached telephone, it dials my cell phone.
So, you can tell your asterisk tandem to wait for 7 digits, and then prepend the digit one when it dials out.

Q: Do the asterisk cards (and asterisk) support rotary pulse dialing, or does one need tone->pulse converters on his switch?
A: Yes, both Asterisk and the Dahdi hardware made by Digium can understand pulse dialing. There is a parameter that needs to be set in /etc/zapata.conf: pulsedial=yes

Q: What do you recommend as a minimum (and cheapest) hardware complement for the Asterisk interface to a private *PBX*?
A: What I have learned so far is this:

Q: I have several computers and have been considering loading Linux on one of them, and I have a broadband connection. What else would I need to participate, and what would I gain?
A: You would need an ethernet port on that computer, either on the motherboard, or as a card. And you would need a card in your PC that would serve as the physical interface between your computer and your phone system.

Q: Do I need to subscribe to a service such as vonage?
A: Not unless you need to connect to the PSTN through your Linux box. If all you want to do is connect to the private collectors' network, you would not need a Vonage-like service.

Q: I installed "everything" from the RedHat distribution. I almost had to do it that way in order to get all the things that Asterisk needs. But it now boots up this GUI that really slows down my computer. What can I do?
A: As you guessed, the GUI is eating up your processor.
In order to turn your PC into a sleek, fast, Asterisk machine, you need to disable the GUI. Very easy to do if you change your default run level by doing the following:

  1. Log in as root to a command prompt.
    The easiest way, if you're in the GUI, would be to do a CTRL-ALT-F2, and then log in.
  2. Change directories to /etc.

    [root@lizzie /]# cd /etc

  3. Create a backup copy of the file you're about to modify.

    [root@lizzie etc]# cp /etc/inittab /etc/inittab.old

  4. Edit inittab using the program vi.

    [root@lizzie etc]# vi inittab

  5. Find the line that says " id:5:initdefault: "
  6. Push "i" to change to insert mode.
  7. Backspace over the number 5 and replace it with the number 3.
  8. Save <ESC> :wq
  9. Reload your inittab.

    [root@lizzie etc]# telinit q

  10. You can check to see if your GUI died as a result of the telinit reload.
    Push F7. If you get back into the GUI, try one additional step to kill it. Push <CTL><ALT><BackSpace>
  11. If the GUI goes away entirely, your new inittab has done the job, and you can delete your backup copy of the inittab.

    [root@lizzie etc]# rm -f /etc/inittab.old

    If the GUI pops right back up again, re-check the editing you did on your inittab by going back to step 4.
  12. Once you are satisfied that your GUI has stopped, reboot your PC.

    [root@lizzie etc]# shutdown -r now

Q: What VOIP protocol is used?
A: We use IAX2 for routing calls between Asterisk systems.

Q: What is NAT?
A: NAT stands for Network Address Translation, and this is a brief synopsis of what it is, why it exists, and how to tell if you have it.

Several years ago, the alarm was raised that the internet was fast running out of IP addresses. And, since IP version 6 was not yet implemented in most computers and routers, it looked like some conservation of IP addresses was needed.

Before the shortage, your ISP would hand out as many IP addresses as your network asked for; now they usually only hand out one. How, then, would someone with more than one computer at home use the internet?
Would they have to pay for two or more DSL connections?

No, the answer is NAT.
Under Network Address Translation, one router (or one computer acting as a router) would be assigned the single IP address that your ISP allows you. But that router has two interfaces. The second interface points toward your LAN. And that router hands out one unique 'private' IP address to each computer on your network that wants one.

Assuming your computer is located on the LAN side of the router, when it requests a web page, the router uses its outside internet and IP address to ask for that web page on your behalf. No matter what any of your computers on your LAN request from the internet, that single 'real' IP address is used by the router to make those connection requests. The router remembers who on the inside asked for what on the outside, and acts as a proxy for your internet connection.

Additionally, if you are running an Asterisk server on your LAN, the router can be configured to forward IAX2 requests (from the internet) directly to your Asterisk machine. See your router's documentation for details.

If you know the IP address your computer is using, you can tell if you are using NAT. The following IP addresses have been reserved for private IP space, and are not routable on the internet itself. They are only routable on the LAN side of a router that is running NAT.

Q: I'd like people to be able to hear call progress as Asterisk dials into my switch. Is there a way I c an allow this?

Yes, there is. Asterisk is ordinarily set up to silence incoming audio on the Dahdi hardware drivers, but one of our collectors (Max) cobbled together a patch. These are the steps needed to apply the patch to your system.

  1. Log in as root to a command prompt.
  2. Change directories to /usr/src

    [root@fubar root]# cd /usr/src

  3. Check to be sure that you have directories named "zaptel" and "asterisk,"
    and that they contain files.

    [root@fubar src]# ls -l asterisk; ls -l zaptel

    If a list of several files was found, go on to the next step.
  4. Download the patch.

    [root@fubar usr]# wget

  5. Apply the patch.

    [root@fubar usr]# patch -p1 < sf.patch

  6. Recompile and re-install the Dahdi drivers.

    [root@fubar usr]# cd zaptel; make clean; make install

  7. Recompile and re-install Asterisk.

    [root@fubar zaptel]# cd ../asterisk; make clean; make install

  8. Reload the Dahdi drivers.

    [root@fubar asterisk]# ztcfg

  9. If the Asterisk console is not running, start it.

    [root@fubar asterisk]# asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvdRT

  10. Re-start Asterisk.

    fubar*CLI> stop now

  11. You're done.


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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