Welcome back to the Transparent Telephony series. If you're a new reader, you may want to start at the beginning: Part 1 - An Introduction.
In the previous installment, we walked through installing Asterisk. In this article, we'll be picking up where we left off and configuring Asterisk to make and receive phone calls using VoIP!
Specifically, we'll:
- Register an account with a VoIP (voice over IP) provider.
- Configure Asterisk to connect to our VoIP provider.
- Create a SIP extension to make and receive calls from.
- Configure Asterisk rules for making calls.
- Configure Asterisk rules for receiving calls.
- Connect a softphone (software phone) to our Asterisk server as an extension and use it to make real calls.
So let's get started!
Getting a VoIP Account
The first thing we need to do is sign up with a VoIP (voice over IP) provider. To quote from Part 1 - An Introduction,
VoIP (voice over IP) is a relatively new way to connect to the PSTN using the Internet. VoIP is extremely low-cost compared to other methods, and very easy to set up as it only requires an active Internet connection to use. More and more businesses are switching over to VoIP for cost reasons.
There are two main types of VoIP protocols that people use with Asterisk. There is SIP (session initiation protocol) and IAX (inter-asterisk exchange). While both of these protocols are still widely in use, SIP has become the dominant VoIP protocol in today's world. In fact, the term SIP has mostly replaced VoIP. When speaking to people in the telephony industry, it is common to hear people say 'I have so and so as my SIP provider'. So from now on, we'll use that terminology as well.
Picking a SIP provider is a very important task. If you Google for 'SIP providers' you'll find thousands of options. Here are the top few I recommend (I've experimented with tons of different SIP providers):
Flowroute is my favorite SIP provider because they have a great website, low prices, and business-class reliability. So instead of walking through the set up for each of the above providers (that would take forever) I'll just cover Flowroute. If you choose to go with another provider, you can still follow along with this article and get a sense of what to do.
Create an Account
To create an account with Flowroute, visit their sign up page and fill in your information. You don't need a credit card, and they'll give you 25 cents of credit (enough to make approximately 30 minutes of calls). If you want to be able to receive calls as well as make them, then you'll have to deposit some money into your account as you'll need to own a DID (more commonly known as phone number) which costs a bit more than 25 cents.
Once your account has been created, you'll be directed to the account dashboard:
Where if you want, you can deposit some money via Amazon Payments.
A Brief Note on Terminology
In the telephony world, SIP accounts are commonly referred to as SIP trunks. Trunk meaning a single account connection to the PSTN.
Connect Your SIP Trunk to Asterisk
Now that we've got a SIP trunk ready for us to use, we need to hook it up to our Asterisk PBX. To get started, go to your Flowroute account management page, and navigate to the 'Interconnection' page, then click on the 'System Configurator' link towards the middle of the page. Next click on the drop down menu and select 'Asterisk' as this is what we're using. The System Configurator will give us all of the Asterisk settings we need to put into our PBX (with only slight modifications required). You should now see a page that looks like this:
A Brief Note on Security
Never ever give out any of your SIP account information! If someone knows your secret and account number, they can easily connect your SIP trunk to their PBX, and begin placing calls on your tab! This happens frequently as many system administrators are not familiar with VoIP security, and don't know what information is valuable, and what isn't.
A good deal of phone scams and telemarketing campaigns originate from hacked SIP accounts, so be careful!
Define Your SIP Trunk Registration String
If you read your System Configurator page, the first thing you'll notice is a section labeled sip.conf. The file sip.conf (located at: /etc/asterisk/sip.conf) is the SIP configuration file that Asterisk uses to define SIP trunks, and other SIP settings. In order to connect our Flowroute SIP trunk to Asterisk, we'll need to edit this file and add in the SIP trunk information specified on the System Configurator page of the Flowroute account dashboard.
The first bit of the Flowroute instructions say:
Add the following to the sip.conf right after general settings. allow=ulaw allow=g729 register => xxx:xxx@sip.flowroute.com
So go ahead and open /etc/asterisk/sip.conf in your favorite text editor (I prefer vim). You'll notice a lot of text, but ignore it for now. These configuration options will be covered in another article in this series.
Scroll down until you see a line that says only:
[authentication]
This line marks the end of the [general] section (which is where we need to put the settings mentioned above). Now go ahead and insert the settings Flowroute gave you directly above the [authentication] line. Your configuration file should now show something like:
allow=ulaw allow=g729 register => xxx:xxx@sip.flowroute.com [authentication]
Define Your NAT Network Settings
If you are running your Asterisk PBX behind a router (eg: your Asterisk system has a private IP), then you'll need to add the following configurations to your sip.conf file directly above the [authentication] line:
nat=yes externip=xx.xx.xx.xx localnet=192.168.0.0/255.255.255.0
Where xx.xx.xx.xx is your external IP (visit IP Chicken if you don't know it), 192.168.0.0 is your subnet, and 255.255.255.0 is your subnet mask. In this example, I'm allowing connections to my Asterisk PBX from any device on the 192.168.0.x network.
The reason we need to add our networking information here is because Asterisk needs to modify the SIP packet headers to ensure that our SIP packets get to the right destination.
Define Your SIP Trunk
The next thing the Flowroute instructions tell us to do is:
Add the following near the bottom of sip.conf [flowroute] type=friend secret=xxx username=xxx host=sip.flowroute.com dtmfmode=rfc2833 context=inbound canreinvite=no allow=ulaw allow=g729 insecure=port,invite fromdomain=sip.flowroute.com
So go ahead and scroll to the bottom of your sip.conf file, and then insert those settings along with a single addition: at the very bottom of your configurations (on a new line after the fromdomain=sip.flowroute.conf) line, insert the following:
qualify=yes
As this will give us some additional information about our SIP connection later on.
Create a SIP Extension
Now that we've set up our SIP trunk, it's time to create an extension. An extension is just another term for 'phone'. SIP extensions (IP phones) are the most common type of extensions in use on modern phone systems. Sure, you can still hook up those old analog phones to your Asterisk PBX, but we'll save that for another day. Today we'll create a simple SIP extension which we will later hook up a soft phone to and use to make and receive calls.
To define a SIP extension, we need to pick several things:
- An extension number. This is typically a numeric number, several digits in length. I'll be using 1000 in the following examples.
- A secret password. This is generally an alphanumeric password of any length used to 'secure' your extension. Any device which wants to make or receive calls from your extension need to know this password in order to authenticate. This password does not need to be remembered, and should be globally unique.
Creating the Extension
Now that you've picked your extension number and secret, let's create it!
Scroll down to the very bottom of your /etc/asterisk/sip.conf file, (below the SIP trunk we created in the previous section), and insert the following (make sure you swap out my extension number for yours, and my secret for yours):
[1000] type=friend nat=yes canreinvite=no secret=mysecretpassword qualify=yes mailbox=1000@default host=dynamic dtmfmode=rfc2833 dial=SIP/1000 context=outgoing
I'll skip over the configuration options for now, I'll cover these in great depth in a later part of the series. If you want to read more about them now, check out the VoIP Info page which describes them in detail.
At this point we've fully configured a Flowroute SIP trunk and a SIP extension with Asterisk. The next thing for us to do will be configuring our network so that our SIP trunk works. So keep reading!
Configure the Network
Before our SIP set up is finished, we need to configure our network to pass SIP and RTP (voice traffic) to our Asterisk server. If your server is not behind a router, you can skip this section.
To do this, configure port forwarding on your router to forward ports 5060 UDP, and 10,000 -> 20,000 UDP to your Asterisk server.
Port 5060 UDP is used for SIP traffic. Ports 10,000 -> 20,000 UDP are used to pass voice traffic (audio).
If your router has any SIP options such as SIP ALG (application layer gateway) or SIP fixup, you'll want to disable them.
Testing the SIP Trunk
Before moving on to the actual call configuration, let's make sure our SIP trunk is actually connected to Asterisk and working.
From the command line, type the following:
asterisk -rx 'sip reload'
Which will force Asterisk to reload our SIP configuration file that we just edited (/etc/asterisk/sip.conf).
Next, let's check to make sure that our Flowroute trunk is registered (connected and in service):
asterisk -rx 'sip show registry'
If your State shows Registered:
[root@localhost asterisk]# asterisk -rx 'sip show registry' Host Username Refresh State Reg.Time sip.flowroute.com:5060 xxxxxxxx 105 Registered Tue, 02 Mar 2010 09:34:23
Then you know that your SIP trunk is connected and working!
Configure the Dial Plan
Since we have our SIP trunk working, we now need to program Asterisk, and teach it how to route calls for us. There are two things that we need to teach Asterisk to do: how to route outgoing calls, and how to route incoming calls. Each of these requires separate Asterisk configuration to work.
Our goal will be to write the following two rules:
- When we dial an 11-digit US phone number from our soft phone, it should call that phone number.
- When someone calls our SIP DID (phone number), we should connect that call to our soft phone so we can talk to the person who called us.
There are several ways to program Asterisk to handle call routing, but the simplest (and native) way to do it is via 'dial plan'. Dial plan is the name of Asterisk's own scripting language. It is very simple, and easy to understand (even for non-programmers).
In the next few sections, we'll write dial plan code to route calls according to our rules above.
Asterisk Dial Plan Basics
All Asterisk dial plan code resides in the file /etc/asterisk/extensions.conf. So when working on dial plan code, always have this file open.
There are two main sections of the extensions.conf file that are always present. There is a [general] context which contains dial plan settings. These settings control Asterisk dial plan parsing options for the most part. The other context is [globals] which contains a list of global variables that can be accessed by any dial plan code.
All Asterisk dial plan code is kept in contexts (much like the SIP configuration we walked through earlier). You can create as many contexts as you like.
Each context should have a clear, meaningful name, which describes what the code inside of it does.
Each line of dial plan code is of the form:
exten => extension,priority,application
Where:
- extension is a regular expression or literal value which is used to determine routing logic.
- priority is a numeric value (or variable) which tells Asterisk in what order to execute the code.
- application is a function that Asterisk executes to perform some functionality.
If you don't understand all of this now, don't worry!
Start With a Clean Slate
Before we begin writing our code, let's quickly create a nice clean extensions.conf file to work in. Remove your current dial plan file (/etc/asterisk/extensions.conf) and replace it with the following:
[general] static=yes writeprotect=no clearglobalvars=no [globals] TRUNK=flowroute
All I did here was remove all comments and clutter from the file, so that we can clearly see what we're doing. I also added a global variable called TRUNK which contains the name of our SIP trunk defined earlier in the article. We'll use this global variable later on to make outbound calls.
Configure Outbound Routing
The goal here is to allow our extension to dial an 11-digit US phone number, and connect it to the PSTN via our Flowroute SIP trunk.
Remember when we configured our SIP extension? One of the keys we specified in the configuration file was:
context=outgoing
The context field of an extension determines what dial plan context the pattern matching will start at. This is similar to the main() function in most programming languages.
The context specified in our SIP extension plays an important role in routing outbound calls. It says (in human English):
When any extension who's 'context' key is equal to 'outgoing' dials a number, send the number that was dialed to the [outgoing] context to be processed. Let the [outgoing] context determine what to do at this point.
So now that we know what context we need to create (the [outgoing] context), let's make it!
Open your extensions.conf file and add the following code to it at the bottom:
[outgoing]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@${GLOBAL(TRUNK)})
Let's go through the code and analyze exactly what is happening.
The first bit:
exten =>
Is standard. All Asterisk dial plan code starts with this bit.
The extension:
_1NXXNXXXXXX
Is a regular expression that Asterisk will use to pattern match the number dialed. Remember above how we said that we're going to route all 11-digit US telephone numbers outbound? This pattern represents an 11-digit US telephone number. Since all 11-digit US telephone numbers begin with the number 1, we hard-code that number. The variable N represents a number (2 through 9), and the variable X represents a number (0 through 9). Therefore, the pattern _1NXXNXXXXXX will match any 11-digit US telephone number.
After the extension (pattern), you'll see a comma character followed by the number 1. The number 1 is the priority. Asterisk dial plan code can contain more than a single line of instructions. In cases where there are multiple lines of code that need to be executed, Asterisk relies on the priority number to determine what to do first.
Imagine that we had code which looked like:
exten => _1NXXNXXXXXX,2,Dial(18002223333@flowroute) exten => _1NXXNXXXXXX,1,Dial(19999999999@flowroute)
In this case, Asterisk would execute the code at priority number 1 first, then the code at priority number 2. So always check for priority numbers when looking at code as they will help determine what is going on.
The last part of the code you see is the called the application:
Dial(SIP/${EXTEN}@${GLOBAL(TRUNK)})
The application we're using here is the Dial() command. This application instructs Asterisk to dial the phone number out of our Flowroute SIP trunk and connect the call to our extension (eg: make an outbound call)!
Everything inside of the ${} characters is a variable reference. Asterisk has several pre-defined 'channel variables' which are always accessible. In our Dial() code, we reference the ${EXTEN} channel variable, which contains the number that was dialed and pattern matched. If we dialed the number 18002223333, then ${EXTEN} would expand to 18002223333.
The ${GLOBAL(TRUNK)} code references the global variable TRUNK which we defined at the top of our extensions.conf file. This code will expand to flowroute before executing the Dial() application.
So after all of the variable expansion is finished, Asterisk actually sees the following (assuming that we dialed the number 18002223333):
Dial(SIP/18002223333@flowroute)
Which is a lot easier to understand! The first part of the line that says SIP/ tells Asterisk that the number we're going to dial should be sent out of our SIP trunk. The @flowroute part tells Asterisk to dial the number 18002223333 on the flowroute SIP trunk, specifically. Doing it this way gives us flexibility. Imagine if we had many SIP trunks on our Asterisk system, and wanted to route certain calls through certain SIP trunks.
Outbound Routing, A Full Walk Through
Let's quickly perform a full walk through of what happens when we dial the number 18002223333 on our extension. (Don't worry that we haven't set up the soft phone yet, we'll get to that later.)
- Asterisk receives a SIP call from our extension (1000) which has dialed the number 18002223333.
- Asterisk looks at the context key in our extension's configuration settings, and sees that it is set to outgoing.
- Asterisk sends the number 18002223333 to the [outgoing] context, defined in /etc/asterisk/extensions.conf to be pattern matched.
- Once Asterisk finds a match for the number, it begins looking for the first application to execute by looking for priority 1.
- Asterisk finds priority 1, and then beings performing variable expansion in the application field to prepare for execution.
- Asterisk finishes variable expansion, then executes: Dial(SIP/18002223333@flowroute).
- Asterisk then calls the number 18002223333 on the flowroute SIP trunk, and connects the call to our extension so that both ends can talk to each other.
That's it for outbound routing! Simple right?
Configure Inbound Routing
In order to route calls inbound, you will need a DID (phone number) with your SIP provider. This way, you can call your number, and it will direct to your Asterisk PBX system. A phone number is a lot like an IP address, each one is unique, and routes to a specific location.
Unfortunately, DIDs cost money (just like you pay for cell phone service and house phone service, you need to pay for VoIP service to rent a DID). If you would like to try out the code that follows, you'll need to deposit some money into your Flowroute account, and then purchase a DID from the web panel. At this point in time, a single DID from Flowroute costs approximately $1.39 per month.
Once you've purchased a DID, write the number down somewhere. It will be an 11-digit phone number. For the code that follows, simply substitute in your DID where necessary, as I will be using the ficticious DID 18182223333.
Open your /etc/asterisk/extensions.conf file and add a new context called [inbound] to the bottom of your file (it can go beneath the [outgoing] context we created in the last section). The context should look like:
[inbound] exten => 18182223333,1,Dial(SIP/1000)
The code here should look familiar to the code in the previous section. All we're doing is dialing the SIP extension 1000 (which is the extension we created earlier). You've probably noticed that there is no @trunk syntax at the end of our Dial() application. If you look back at your SIP configuration file (/etc/asterisk/sip.conf) you'll notice that when we created our extension, one of the keys we defined was:
dial=SIP/1000
Which tells Asterisk that in order to connect a call to that extension, we have to dial SIP/1000.
Inbound Routing, A Full Walk Through
Let's quickly perform a full walk through of what happens when someone calls our DID (18182223333) from the PSTN.
- First our SIP provider will receive the call, and check to see what customer the DID belongs to.
- The SIP provider will then check to see if we are registered to their SIP server.
- Once they've confirmed our registration, they'll forward the call to our Asterisk system.
- Asterisk will receive the call from our Flowroute SIP trunk, and check to see what context it should use to route the call.
- Asterisk will see that our [flowroute] trunk is configured to use the context 'inbound', and will send the DID number, 18182223333 to the [inbound] context for pattern matching.
- Asterisk will match the pattern 18182223333 in our [inbound] context, and begin looking for priority 1.
- Asterisk will find priority 1, and then execute the Dial(SIP/1000) command.
- Asterisk will connect the incoming call to our extension, so that both parties can talk.
Not too bad! We've set up the ability to receive incoming calls in only a single line of code!
At this point, we've created dial plan rules for both the outgoing and incoming call routing. Save your extensions.conf file, and let's move on.
Apply Your New Dial Plan Rules
Now that we've finished writing our dial plan code, we need to reload Asterisk to have it re-scan our extensions.conf file. To do this, simply type:
asterisk -rx 'dialplan reload'
From the command line.
The next section which will teach you how to hook up a soft phone to your Asterisk system so you can actually test your system!
Set Up a Softphone
In this section, we'll set up a soft phone to use to make calls. Soft phones are just SIP clients that can connect to Asterisk and act as normal phones. Asterisk can work with normal analog telephones as well as fancier (and more expensive) SIP phones, but SIP phones are much easier to set up, so we'll be configuring a soft phone today. If you want to hook up your analog phone to your Asterisk server--don't despair--that will be covered in another article in this series.
There are tons of soft phones to choose from, but I'll be walking you through using X-Lite, as it is one of the most popular and widely used.
First thing you'll want to do is download and install X-Lite on your computer (it runs on all platforms).
Once you've got it installed, open it up. If you don't have your extension information in front of you, open up your SIP configuration file (/etc/asterisk/sip.conf) and look at your extension definition.
Right click on the main window display in X-Lite and click on 'SIP Account Settings'. Now click 'Add...' to add a SIP account.
Here are the values you should fill in:
- Display Name - Set this to anything you like. I prefer 1000 (the extension I'm using). 2 .User name - Set this to your extension number (1000).
- Password - Set this to your secret.
- Authorization user name - Set this to your extension number (1000).
- Domain - Set this to the IP address of your Asterisk server.
- Make sure the box next to 'Register with domain and receive incoming calls' is checked.
Once you've configured all those settings, press OK and then close out of the configuration menu.
If your network and SIP configuration files have been properly configured, your X-Lite phone will now say 'Registered' at the top of the window! If it is not working, read back through the setup instructions and make sure you didn't miss anything. If you are using a virtual machine, also make sure your host network settings have been configured to receive incoming traffic.
Making And Receiving Your First Call
Now that we've gotten everything set up, let's actually make some calls!
On your X-Lite soft phone, go ahead and dial any 11-digit US telephone number (since this is the rule that we configured for outbound routing). When you hit dial (the green button) you'll make a call just like you would on a normal phone, and you'll be connected to the phone number you dialed!
If you want to receive a call, use your cell phone to call your DID, and Asterisk will route the call directly to your X-Lite phone. You will see and hear you X-Lite phone ring, and you can pick up the call by clicking the green button.
Conclusion
This was a very large article, and took a considerable amount of time to write. I hope that if you got this far, you were able to clearly configure and understand the basic Asterisk setup required to make outbound calls, and to receive inbound calls.
The material covered in this part of the series is bulky, but very important in understanding the way Asterisk works. In future parts of the Transparent Telephony series, we'll have significantly shorter, more targeted articles, so following along should be easier.
If you have any questions, comments, or suggestions about this article, please leave a comment and I'll be happy to respond.
