Some Asterisk setup questions


I recently installed AsterixNow with FreePBX Version

I’ve configured the extensions and voicemail, etc. so far.

So, I have a few questions:

  1. What is the best way to test out using an outside line (trunk). Sign up for SIPStation?

  2. What hardware would I need if I wanted to connect the system to outside regular PSTN analog lines?

  3. Can I interface with Vonage?

  4. I want to be able to retrieve Caller ID data and other data concerning calls (calling number, time, date, extension that picked up, etc.) from the database on the Asterisk server. I want to be able to import that data into a Windows-based application I’ve written using VB6. How do I attach to the database on the Asterisk server from Windows XP?

Any help/advice greatly appreciated.



FreePBX 2.7 is quite old. 2.10 is current. The Asterisk Now distribution is not very current.

We have our own ISO (The FreePBX distro that includes the latest FreePBX and Asterisk).

You might consider this option.

Welcome to the forums.

First, welcome to the world of VOIP. You sound like you’re in the same position that I was in about 1 year ago.

Second, wwitch to the FreePBX Distro or PBX In A Flash. AsteriskNOW had a number of issues when it was released that still have not been fixed, i.e. Call Detail Records don’t work.

Third, I’d sign up for VOIP.MS, Callcentric, and Flowroute.

Fourth, you’d want to use an FXO device, like the Grandstream GXW-4104, which is what I use.

Fifth, you can interface with Vonage, but you definitely don’t want to. Vonage is way too expensive.

Sixth, many of your next set of questions are answered here:

P.S. I don’t know anything about your Caller ID export, but I’m sure that it can be done.

Thanks for the heads-up. I believe I tried previously to install the FreePBX distro, but I found there were some issues that occurred during the install. Is there a set of recommended settings to use when setting up?

Is his also based on Centos flavor of Linux?

I also note there is no GUI interface supplied with the AsteriskNow install (well, not that I know of). Does such an interface come with the FreePBX install?

What database does the Asterisk version use?

Thanks for the help


Thanks a lot for the valuable suggestions!

Which of VOIP.MS, Callcentric and Flowroute would you recommend starting out with?

What’s the difference between the FreePBX distro and PBX In A Flash?

Is the Grandstream hardware better than Digium, in your opinion? It supports up to 4 PSTN lines, right?



Start with Callcentric. They are the easiest to use and have the best support.

FreePBX Distro and PBX In A Flash both provide you with an Asterisk-based PBX that used FreePBX to configure Asterisk.

PIAF has more options to choose from on installation (mostly choosing differing versions), has very active user forums, but allows no official way to update. PIAF is run by a retired lawyer named Ward Mundy who appears to spend most of his time traveling. He’s banned me from his forums for disagreeing with him three times.

FreePBX Distro has fewer options, but has an upgrade path when new versions come out, but has a less active user base. It is run by the same people who created FreePBX and by, which sells FreePBX based phone systems.

I’ve never used Digium hardware, but I don’t think that they offer an FXO Gateway, i.e. a device that connects your network to an FXO. I think they just offer FXO Cards. If you are going to go with a card, most people around here would recommend Sangoma. I prefer gateways because I run my PBX virtually, i.e. in VMWare Player or a VMWare ESXi, and it is very difficult to get a virtual machine to work with a card installed in your computer.

I did not know Sangoma makes external gateways.

If you know how to configure Cisco routers the best thing to do is to use a Cisco router and install FXO ports.

1760V are under $100 on eBay and the quality blows away any cheap plastic junk FXO gateway.

Thanks for the suggestions.

Just to get my terminology correct: An FXO card (or FXO Gateway) connects to the traditional phone company’s analog ports (lines), which deliver the dialtone, and on the other side connects to the PC, providing an interface between the digital (the Asterisk PBX) and the analog (the PSTN phone line/s). Is that correct?

I would need to have an FXO gateway or card to allow the Asterisk PBX to make use of my existing traditioanl POTS telephone lines, right?

But now I’m wondering: Once I get my Asterisk PBX set up, and I sign up with Callcentric, then we can make all the outgoing calls we want with the Callcentric facility. Then we don’t really need the analog lines at all any more, right?
(Except in the case that we lose our internet connectivity – then we could still use the analog lines for outgoing calls. But what about incoming calls?)

By the way, how many outgoing calls can one make simultaneously when using Callcentric, etc.?

One (hopefully) last question: What is the call quality like with Callcentric? Some of these low-cost call suppliers skimp on the bandwidth they use to carry each call, and then the quality of the call suffers.

I’m very grateful for the help and advice. Thanks to all of you.

BTW, I think I’ll find a spare hard drive and try installing the PIAF version on the second hard drive. With the AsteriskNow, there doesn’t seem to be a database where call details are stored.



Yes FXO ports connect to Analog phones.

I always recommend having a backup.

I don’t know anything about Callcentric call quality or policies, some companies charge by the minute with unlimited calls, others charge by the call path. What does their agreement say?

Once you port your numbers to Callcentric your POTS lines will be a backup and used for 911 traffic.

You sure worry more about the quality of your Internet connection that the carriers back end.

Asterisknow has all sorts of issues. It does store CDR’s in a database, it just is broken and doesn’t set the database credentials by default. However the distribution has other issues, I would not run it in production.

PBX in a flash is supported and a good choice as is the FreePBX distro.

Thanks for the heads-up. I installed PIAF. All seemed to be OK, but now when I go into FreePBX it says Asterisk is not running. I seem to recall that it said something about having to compile Asterisk manually when the install finished, because there was some problem during the installation process.

Any ideas would be appreciated.

So far, they haven’t granted me permission to post to the PIAF board. Maybe someone here knows?

Thanks in advance


If Asterisk is not running, then something went wrong when you installed PIAF. You’ll probably want to reinstall. This time, you might want to use the FreePBX Distro.

My approach is to retain ONE standard POTS line. I use this line for 911 calls and outgoing calls that are local.

I forward calls that come in on this line to the phone number assigned to me by a VOIP Provider. Incoming calls thus all come in over the internet. However, if my internet goes down, I can turn off the call forwarding and still receive and make calls.

The number if inbound calls supported by Callcentric depends upon which plan you purchase. I can’t remember how many simultaneous outbound calls they allow, but I think it is 5. If that’s not enough, you can always sign up for a second account.

The only quirky thing about CC is that they use Freeswitch and not Asterisk, so the instructions for getting FreePBX to work with CC is a little weird and involves editing text files.

The only other thing to remember about using this system is that there is a significant Asterisk bug that will cause your phones to stop working if you lose internet access. There’s a work-around, which is detailed in the instructions that I linked to. When you think you’re done setting up your system, pull the plug on your internet and try making a few calls and make sure that your system isn’t affected by the bug.