OK - there are a few things you’ll need to get an understanding of before your quest for telephony nirvana can be achieved.
- The nature of IP address networking. This is a very basic skill and requires a little bit of “out of classroom” work on your part.
- Some of the network management tasks you will be undertaking are outside the scope of Asterisk (and its management program FreePBX). Because of that, at some point you will need to avail yourself of the “root terminal” login.
- While it is not necessary to be a complete Unix wizard, having a basic understanding of the underlying operating system (CentOS - a distribution of RedHat Linux) is going to make your life considerably less challenging.
So, having said that, you need a few things that will make your life easier.
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A network map. This is the document that tells you where everything is. No two devices can have the same network address, so a map is indispensable. There are automated tools that can help you do that if you don’t have one from your predecessor. For most installations, a detailed map is really critical AS LONG AS you make sure nothing is talking on someone else’s address. For example - you are probably using a private network in the range of 192.168.0.1 through 192.168.0.254. Of course, this is just an example, but this address block is chosen a large percentage of the time.
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The “root” password. At some point, you will have to log into the main PBX computer to do something. I don’t know what yet, but I guarantee you will need to. You can do this from the console of the main PBX (there should be a Login: prompt) or you can do it using one of the myriad SSH client programs.
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The server IP address. For most installations, this is a specific number at one end of the address block. There are as many models as there are network guys. I usually use x.x.x.1 as my gateway (so, if you are using 192.168.0.something as your network, that would be 192.168.0.1), 192.168.0.2 as my DNS/DHCP/OBX/Network Management server (if I have one) and then starting numbering workstations at 192.168.0.50 and phones at 192.168.0.150. This gives you space for 45+ servers and printers, and 100 phones and computers, with a couple of spots left over for things like wireless access points.
Since you are using Asterisk 13, you will need to know that there are TWO different ways to connect SIP phones to your system. One is called PJ-SIP and is the newer technology. It offers a lot of really cool features that you don’t need yet. The other is called “Chan-SIP” and is the Asterisk 11-and-before SIP interface. When you set them up, you tell them which “port” you want them to listen on. Common numbers are 5060, 5061, and 5160.
Having said that, it’s at this moment that your brain may lock up, especially if you’re not done with that networking homework I gave you. Don’t worry - we’ve all been there.
So, log into FreePBX and look around. Let’s assume for a minute that your PBX is set up in the local network. For now (not long, but for now) disable the firewall that comes with FreePBX. It’s on the “Connectivity” menu - you’ll want to know where it is because we will be re-enabling it later.
Now, look at the “settings” tab. There should be an entry called “SIP Settings”. This set of three pages will tell you which driver you are using and on which port. We can assume, for now, that you are using Chan-SIP for your phones. That’s fine - just look at the port number you are using. You will need this to configure your phones.
Next, set up your extension. Start with one, make it a Chan-SIP extension (the extension types are managed separately, so this step is kind of critical). Set the extension number, the user (which is usually the extension number, but doesn’t have to be), and the password. The system will provide a ridiculously long, excellent password which you can either use or discard and replace with something simple (for testing). Once you get good at running the system, I actually recommend setting the phone’s SIP password to the one provided by the channel driver - it will save you money in the long run.
Now, set up your phone. Set the SIP user and password. Also, set the IP address of the SIP Server.
On the port number, there are lots of ways that can go. The first is that the phone will actually give you a place to put it. The second is that it needs to be included in the IP address using a colon (192.168.0.2:5060)… Other phones simply will not use anything but 5060. Your mileage may vary and it is a good exercise for you.
Set everything up and try to reset the phone.
In the “Reports” tab, there is a “View Log Files” entry. Use that and look to see why the phone didn’t connect. If you can’t figure that out, you could also log into the server console as “root”. You will end up at a prompt that looks like this: “#”.
At that prompt, type “more /var/log/asterisk/full”. Use the ‘/’ key to find the extension number of your phone. Use the ‘n’ key to find the next occurrence. The ‘q’ key will quit out of more.
You can not try using the system’s runtime component. Type ‘asterisk -vvvr’ to connect to the Asterisk program that’s running on the server. Reset your phone and watch the screen - you should see the phone attempt to connect and either work or error out.
That’s enough for now - if you get all of that done and still can’t get the phones to connect, come back and ask specific questions…