- Q: What if I don't have a 'system' to hook up. How can I attach a regular phone to the network?
- Q: My step switch has two-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. *
- Q: My step switch won't know that it needs to access the trunks going to the Asterisk switch unti l after my internal user dials "1." How will the Asterisk box deal with the first digit being missi ng?
- Q: Do the asterisk cards (and asterisk) support rotary pulse dialing, or does one need tone->pulse converters on his switch?
- Q: What do you recommend as a minimum (and cheapest) hardware complement for the Asterisk interface to a private *PBX*?
- 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?
- Q: Do I need to subscribe to a service such as vonage?
- 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?
- Q: Should I use SIP or IAX to connect to the Collectors' Net?
- Q: I'd like people to be able to hear call progress as Asterisk dials into my legacy switch. Is there a way I can allow this?
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 ckts.info tandem to host you, or one of our members could volunteer to do the same. Just ask the webmaster or 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 Zaptel 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:
- For the lowest possibility of hassles, get a new board from Digium. They'll stand by you every step of the way, from talking with you to determine what board you will need to helping you get it installed properly to standing behind it if something goes wrong.
- Other brand new FXO and FXS cards are manufactured offshore, but sold by US companies. The one with which I've had dealings is Voicetronix. My only issue in dealing with them is that they are located in Australia, so it's a bit difficult to synchronize their business hours with your free time. They definitely know their stuff, though, and were able to work with me to get some nasty echo problems cleared up.
- DiaLogic has a whole slew of products that will work with Asterisk, but I've not heard how well they work
- There are clones of Digium's Zaptel cards that are made in Asia. They are cheap, and generally work well. But they have no tech support. You buy these if you're skilled at installing the Digium cards.
- And there are others, all listed at http://www.asterisk.org/index.php?menu=hardware.
- Lastly, there is the used market. I sold my Voicetronix card on eBay, and the guy who bought it from me was quite happy with the card, as well as the price he paid. If you go that route, the usual eBay precautions apply: Deal with only those people who have perfect or almost perfect records as regards positive feedback. If you want additional peace of mind, buy shipping insurance. Do a search on TDM400P or X100P, depending on whether you want the FXO or FXS card. (Again, consult the website in the previous paragraph to see which is which).
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.
As an aside, there are services out there that are easier to integrate into Asterisk than Vonage is, and most are just as cheap, if not more so. And they don't require an FXO card in the Asterisk box to use them. Check out iConnectHere or Voice Pulse if you want a PSTN connection. Or, for a more complete list of VoIP providers, try the VoIP Catalog.
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:
- 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.
- Change directories to /etc.
[root@lizzie /]# cd /etc
- Create a backup copy of the file you're about to modify.
[root@lizzie etc]# cp /etc/inittab /etc/inittab.old
- Edit inittab using the program vi.
[root@lizzie etc]# vi inittab
- Find the line that says " id:5:initdefault: "
- Push "i" to change to insert mode.
- Backspace over the number 5 and replace it with the number 3.
- Save <ESC> :wq
- Reload your inittab.
[root@lizzie etc]# telinit q
- 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>
- 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
- Once you are satisfied that your GUI has stopped, reboot your PC.
[root@lizzie etc]# shutdown -r now
Q: Should I use SIP or IAX to connect to the Collectors' Net?
A: Either should work well. IAX is the simplest to set up, especially if your computer is NATted, i.e., using private IP space.
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 SIP and 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.
- 10.0.0.0 -- 10.255.255.255
- 172.16.0.0 -- 172.31.255.255
- 192.168.0.0 -- 192.168.255.255
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 Zaptel hardware drivers, but one of our collectors (Max) cobbled together a patch. These are the steps needed to apply the patch to your system.
- Log in as root to a command prompt.
- Change directories to /usr/src
[root@fubar root]# cd /usr/src
- 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.
- Download the patch.
[root@fubar usr]# wget http://www.lightlink.com/mhp/sf/sf.patch
Apply the patch.
[root@fubar usr]# patch -p1 < sf.patch
Recompile and re-install the Zaptel drivers.
[root@fubar usr]# cd zaptel; make clean; make install
Recompile and re-install Asterisk.
[root@fubar zaptel]# cd ../asterisk; make clean; make install
Reload the Zaptel drivers.
[root@fubar asterisk]# ztcfg
If the Asterisk console is not running, start it.
[root@fubar asterisk]# asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvdRT
fubar*CLI> stop now
- You're done.
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