Chapter 5. Dialplan Basics
Everything should be made as simple as possible, but not simpler.
The dialplan is truly the heart of any Asterisk system, as it defines how Asterisk handles inbound and outbound calls. In a nutshell, it consists of a list of instructions or steps that Asterisk will follow. Unlike traditional phone systems, Asterisk’s dialplan is fully customizable. To successfully set up your own Asterisk system, you will need to understand the dialplan.
If you have attempted to read some sample dialplans and found them overwhelming, or if you’ve tried to write an Asterisk dialplan and had no success, help is at hand. This chapter explains how dialplans work in a step-by-step manner and teaches the skills necessary to create your own. The examples have been designed to build upon one another, so feel free to go back and reread a section if something doesn’t quite make sense. Please also note that this chapter is by no means an exhaustive survey of all the possible things dialplans can do; our aim is to cover just the fundamentals. We’ll cover more advanced dialplan topics in later chapters.
Dialplan Syntax
The Asterisk dialplan is specified in the configuration file named extensions.conf.
Tip
The extensions.conf file usually resides in the /etc/asterisk/ directory, but its location may vary depending on how you installed Asterisk. Other common locations for this file include /usr/local/asterisk/etc/ and /opt/asterisk/etc/.
The dialplan is made up of four main concepts: contexts, extensions, priorities, and applications. In the next few sections, we’ll cover each of these parts and explain how they work together. After explaining the role each of these elements plays in the dialplan, we will step you though the process of creating a basic, functioning dialplan.
Contexts
Dialplans are broken into sections called contexts. Contexts are named groups of extensions, which serve several purposes.
Contexts keep different parts of the dialplan from interacting with one another. An extension that is defined in one context is completely isolated from extensions in any other context, unless interaction is specifically allowed. (We’ll cover how to allow interaction between contexts near the end of the chapter.)
As a simple example, let’s imagine we have two companies sharing an Asterisk server. If we place each company’s voice menu in its own context, they are effectively separated from each other. This allows us to independently define what happens when, say, extension 0 is dialed: people pressing 0 at Company A’s voice menu will get Company A’s receptionist, and callers pressing 0 at Company B’s voice menu will get Company B’s receptionist. (This example assumes, of course, that we’ve told Asterisk to transfer the calls to the receptionists when callers press 0.)
Contexts are denoted by placing the name of the context
inside square brackets ([
]
). The name can be made up of the letters A through Z
(upper- and lowercase), the numbers 0 through 9, and the hyphen and
underscore.[70] For example, a context for incoming calls looks like
this:
[incoming]
Note
Context names have a maximum length of 79 characters (80 characters –1 terminating null)
All of the instructions placed after a context definition are
part of that context, until the next context is defined. At the
beginning of the dialplan, there are two special contexts named
[general]
and [globals]
. The [general]
section contains a list of general
dialplan settings (which you’ll probably never have to worry about),
and we will discuss the [globals]
context the Global variables” section; for
now it’s just important to know that these two contexts are special.
As long as you avoid the names [general]
and [globals]
, you may name your contexts
anything you like.
When you define a channel (which is how you connect things to the system), one of the parameters that is defined in the channel definition is the context. In other words, the context is the point in the dialplan where connections from that channel will begin.
Another important use of contexts (perhaps the most important) is to provide security. By using contexts correctly, you can give certain callers access to features (such as long-distance calling) that aren’t made available to others. If you don’t design your dialplan carefully, you may inadvertently allow others to fraudulently use your system. Please keep this in mind as you build your Asterisk system.
Warning
The doc/ subdirectory of the Asterisk source code contains a very important file named security.txt, which outlines several steps you should take to keep your Asterisk system secure. It is vitally important that you read and understand this file. If you ignore the security precautions outlined there, you may end up allowing anyone and everyone to make long-distance or toll calls at your expense!
If you don’t take the security of your Asterisk system seriously, you may end up paying—literally! Please take the time and effort to secure your system from toll fraud.
Extensions
In the world of telecommunications, the word extension usually refers to a numeric identifier given to a line that rings a particular phone. In Asterisk, however, an extension is far more powerful, as it defines a unique series of steps (each step containing an application) that Asterisk will take that call through. Within each context, we can define as many (or few) extensions as required. When a particular extension is triggered (by an incoming call or by digits being dialed on a channel), Asterisk will follow the steps defined for that extension. It is the extensions, therefore, that specify what happens to calls as they make their way through the dialplan. Although extensions can certainly be used to specify phone extensions in the traditional sense (i.e., extension 153 will cause the SIP telephone set on John’s desk to ring), in an Asterisk dialplan, they can be used for much more.
The syntax for an extension is the word exten
, followed by an arrow formed by the equals sign and the greater-than sign, like
this:
exten =>
This is followed by the name (or number) of the extension. When dealing with traditional telephone systems, we tend to think of extensions as the numbers you would dial to make another phone ring. In Asterisk, you get a whole lot more; for example, extension names can be any combination of numbers and letters. Over the course of this chapter and the next, we’ll use both numeric and alphanumeric extensions.
Tip
Assigning names to extensions may seem like a revolutionary concept, but when you realize that many VoIP transports support (or even actively encourage) dialing by name or email address instead of only dialing by number, it makes perfect sense. This is one of the features that makes Asterisk so flexible and powerful.
A complete extension is composed of three components:
These three components are separated by commas, like this:
exten =>name
,priority
,application()
Here’s a simple example of what a real extension might look like:
exten => 123,1,Answer()
In this example, the extension name is 123
, the priority is 1
, and the application is Answer()
. Now, let’s
move ahead and explain priorities and applications.
Priorities
Each extension can have multiple steps, called priorities. Each priority is numbered sequentially, starting with 1, and executes one specific application. As an example, the following extension would answer the phone (in priority number 1), and then hang it up (in priority number 2):
exten => 123,1,Answer() exten => 123,2,Hangup()
Don’t worry if you don’t understand what Answer()
and Hangup()
are—we’ll cover them shortly. The
key point to remember here is that for a particular extension,
Asterisk follows the priorities in order.
Unnumbered priorities
In older releases of Asterisk, the numbering of priorities caused a lot of problems. Imagine having an extension that had 15 priorities, and then needing to add something at step 2. All of the subsequent priorities would have to be manually renumbered. Asterisk does not handle missing steps or misnumbered priorities, and debugging these types of errors was pointless and frustrating.
Beginning with version 1.2, Asterisk addressed this problem.
It introduced the use of the n
priority, which stands for “next.” Each time Asterisk encounters a
priority named n
, it takes the
number of the previous priority and adds 1. This makes it easier to
make changes to your dialplan, as you don’t have to keep renumbering
all your steps. For example, your dialplan might look something like
this:
exten => 123,1,Answer() exten => 123,n,do something
exten => 123,n,do something else
exten => 123,n,do one last thing
exten => 123,n,Hangup()
Internally, Asterisk will calculate the next priority number
every time it encounters an n
.[71] You should note, however, that you must
always specify priority number 1. If you accidentally put
an n
instead of 1
for the first priority, you’ll find that
the extension will not be available.
Priority labels
Starting with Asterisk version 1.2 and higher, common practice is to assign text labels to priorities. This is to ensure that you can refer to a priority by something other than its number, which probably isn’t known, given that dialplans now generally use unnumbered priorities. To assign a text label to a priority, simply add the label inside parentheses after the priority, like this:
exten => 123,n(label
),application()
Warning
A very common mistake when writing labels is to insert a
comma between the n
and the
(
, like this:
exten => 123,n,(label
),application()
;<-- THIS IS NOT GOING TO WORK
This mistake will break that part of your dialplan, and you will get an error that the application cannot be found.
In the next chapter, we’ll cover how to jump between different priorities based on dialplan logic. You’ll be seeing a lot more of priority labels, and you will be using them often in your dialplans.
Applications
Applications are the workhorses of the dialplan. Each application
performs a specific action on the current channel, such as playing a
sound, accepting touch-tone input, dialing a channel, hanging up the
call, and so forth. In the previous example, you were introduced to
two simple applications: Answer()
and Hangup()
. You’ll
learn more about how these work momentarily.
Some applications, such as Answer()
and Hangup()
, need no other instructions to do
their jobs. Other applications require additional information. These
pieces of information, called arguments, can be passed on to the
applications to affect how they perform their actions. To pass
arguments to an application, place them between the parentheses that
follow the application name, separated by commas.
Tip
Occasionally, you may also see the pipe character (|
) being
used as a separator between arguments, instead of a comma. Feel free
to use whichever you prefer. For the examples in this book, we will
be using the comma to separate arguments to an application, as the
authors prefer the look of this syntax. You should be aware,
however, that when Asterisk parses the dialplan, it converts any
commas in the application arguments to pipes.
As we build our first dialplan in the next section, you’ll learn to use applications (and their associated arguments) to your advantage.
A Simple Dialplan
Now we’re ready to create our first dialplan. We’ll start with a very simple example. We are going to instruct Asterisk to answer a call, play a sound file, and hang up. We’ll use this simple example to point out the most important dialplan fundamentals.
For the examples in this chapter to work correctly, we’re assuming
that at least one channel (either Zap, SIP, or IAX2) has been created
and configured (as described in the previous chapter), and that all
calls coming into that channel enter the dialplan at the [incoming]
context. If you have been creative
with any previous examples, you may need to make adjustments to fit your
particular channel names.
The s Extension
Because of the technology we are using in our channels, we need to
cover one more thing before we get started with our dialplan. We need
to explain extension s
. When calls
enter a context without a specific destination extension (for example,
a ringing FXO line), they are passed to the s
extension. (The s
stands for “start,” as this is where a
call will start if no extension information was passed with the
call.)
Since this is exactly what we need for our dialplan, let’s begin
to fill in the pieces. We will be performing three actions on the call
(answer it, play a sound file, and hang it up), so our extension
called s
will need three
priorities. We’ll place the three priorities below [incoming]
, because we have decided that all incoming calls should
start in this context.[72]
[incoming] exten => s,1,application()
exten => s,n,application()
exten => s,n,application()
Now all we need to do is fill in the applications, and we’ve created our first dialplan.
The Answer(), Playback(), and Hangup() Applications
If we’re going to answer the call, play a sound file, and then hang up,
we’d better learn how to do just that. The Answer()
application is used to answer a
channel that is ringing. This does the initial setup for the channel
that receives the incoming call. (A few applications don’t require
that you answer the channel first, but properly answering the channel
before performing any other actions is a very good habit.) As we
mentioned earlier, Answer()
takes
no arguments.
The Playback()
application is
used for playing a previously recorded sound file over a channel. When
using the Playback()
application,
input from the user is simply ignored.
Tip
Asterisk comes with many professionally recorded sound files, which should be found in the default sounds directory (usually /var/lib/asterisk/sounds/). When you compile Asterisk, you can choose to install various sets of sample sounds that have been recorded in a variety of languages and file formats. We’ll be using these files in many of our examples. Several of the files in our examples come from the Extra Sound Package, so please take the time to install it (see Chapter 3). You can also have your own sound prompts recorded in the same voices as the stock prompts by visiting http://thevoice.digium.com/.
To use Playback()
, specify a
filename (without a file extension) as the argument. For example,
Playback(filename)
would play the
sound file called filename.gsm, assuming it was
located in the default sounds directory. Note that you can include the
full path to the file if you want, like this:
Playback(/home/john/sounds/filename)
The previous example would play filename.gsm from the /home/john/sounds/ directory. You can also use relative paths from the Asterisk sounds directory as follows:
Playback(custom/filename)
This example would play filename.gsm from the custom/ subdirectory of the default sounds directory (probably /var/lib/asterisk/sounds/custom/filename.gsm). Note that if the specified directory contains more than one file with that filename but with different file extensions, Asterisk automatically plays the best file.[73]
The Hangup()
application does
exactly as its name implies: it hangs up the active channel. You
should use this application at the end of a context when you want to
end the current call to ensure that callers don’t continue on in the
dialplan in a way you might not have anticipated. The Hangup()
application takes no
arguments.
Our First Dialplan
Now that we have designed our extension, let’s put together all the pieces to create our first dialplan. As is typical in many technology books (especially computer programming books), our first example will be called “Hello World!”
In the first priority of our extension, we’ll answer the call. In the second, we’ll play a sound file named hello-world.gsm, and in the third we’ll hang up the call. Here’s what the dialplan looks like:
[incoming] exten => s,1,Answer() exten => s,n,Playback(hello-world) exten => s,n,Hangup()
If you have a channel or two configured, go ahead and try it
out![74] Simply create a file called
extensions.conf, (probably in
/etc/asterisk) and insert the four lines of
dialplan code we just designed. If it doesn’t work, check the Asterisk
console for error messages, and make sure your channels are assigned
to the [incoming]
context.
Even though this example is very short and simple, it emphasizes the core concepts of contexts, extensions, priorities, and applications. If you can get this to work, you have the fundamental knowledge on which all dialplans are built.
Let’s build upon our example. After all, a phone system that simply plays a sound file and then hangs up the channel isn’t that useful!
Building an Interactive Dialplan
The dialplan we just built was static; it will always perform the same actions on every call. We are going to start adding some logic to our dialplan so that it will perform different actions based on input from the user. To do this, we’re going to need to introduce a few more applications.
The Background(), WaitExten(), and Goto() Applications
One of the most important keys to building interactive
Asterisk dialplans is the Background()
[75] application. Like Playback()
, it plays a recorded sound
file. Unlike Playback()
, however, when the caller
presses a key (or series of keys) on her telephone keypad, it
interrupts the playback and goes to the extension that corresponds
with the pressed digit(s). If a caller presses 5, for example,
Asterisk will stop playing the sound prompt and send control of the
call to the first priority of extension 5.
The most common use of the Background()
application is to create voice menus (often called auto-attendants or phone
trees). Many companies use voice menus to direct callers to
the proper extensions, thus relieving their receptionists from having
to answer every single call.
Background()
has the same
syntax as Playback()
:
exten => 123,1,Answer() exten => 123,n,Background(main-menu)
In earlier versions of Asterisk, if the Background()
application finished playing
the sound prompt and there were no more priorities in the current
extension, Asterisk would sit and wait for input from the caller.
Asterisk no longer does this by default. If you want Asterisk to wait
for input from the caller after the sound prompt has finished playing,
you can call the WaitExten()
application. The WaitExten()
application waits for the caller to enter DTMF digits, and is
frequently called directly after the Background()
application, like this:
exten => 123,1,Answer() exten => 123,n,Background(main-menu) exten => 123,n,WaitExten()
If you’d like the WaitExten()
application to wait a specific number of seconds for a response
(instead of using the default timeout), simply pass the number of
seconds as the first argument to WaitExten()
, like this:
exten => 123,n,WaitExten(5)
Both Background()
and
WaitExten()
allow the caller to
enter DTMF digits. Asterisk then attempts to find an extension in the
current context that matches the digits that the caller entered. If
Asterisk finds an unambiguous match, it will send the call to that
extension. Let’s demonstrate by adding a few lines to our
example:
exten => 123,1,Answer() exten => 123,n,Background(main-menu) exten => 123,n,WaitExten() exten => 2,1,Playback(digits/2) exten => 3,1,Playback(digits/3) exten => 4,1,Playback(digits/4)
If you call into extension 123 in the example above, it will
play a sound prompt that says “main menu.” It will then wait for you
to enter either 2
, 3
, or 4
.
If you press one of those digits, Asterisk will read that digit back
to you. You’ll also find that if you enter a different digit (such as
5
), it won’t give you what you
expected.
It is also possible that Asterisk will find an ambiguous match.
This can be easily explained if we add an extension named 1
to the previous example:
exten => 123,1,Answer() exten => 123,n,Background(main-menu) exten => 123,n,WaitExten() exten => 1,1,Playback(digits/1) exten => 2,1,Playback(digits/2) exten => 3,1,Playback(digits/3) exten => 4,1,Playback(digits/4)
Dial extension 123, and then at the main menu prompt dial
1
. Why doesn’t Asterisk immediately
read back the number one to you? It’s because the digit 1
is ambiguous; Asterisk doesn’t know
whether you’re trying to go to extension 1 or extension 123. It waits
a few seconds to see if you’re going to dial another digit (such as
the 2
in extension 123). If you
don’t dial any more digits, Asterisk will eventually time out and send
the call to extension 1. (We’ll learn how to choose our own timeout
values in Chapter 6.)
Before going on, let’s review what we’ve done so far. When users
call into our dialplan, they will hear a greeting. If they press
1
, they will hear the number one,
and if they press 2
, they will hear
the number two, and so on. While that’s a good start, let’s embellish
it a little. We’ll use the Goto()
application to make the dialplan repeat the greeting after playing
back the number.
As its name implies, the Goto()
application is used to send the call
to another part of the dialplan. The syntax for the Goto()
application requires us to pass the
destination context, extension, and priority on as arguments to the
application, like this:
exten => 123,n,Goto(context
,extension
,priority
)
Now, let’s use the Goto()
application in our dialplan:
[incoming] exten => 123,1,Answer() exten => 123,n,Background(main-menu) exten => 1,1,Playback(digits/1) exten => 1,n,Goto(incoming,123,1) exten => 2,1,Playback(digits/2)exten => 2,n,Goto(incoming,123,1)
These two new lines (highlighted in bold) will send control of
the call back to the 123
extension
after playing back the selected number.
Tip
If you look up the details of the Goto()
application, you’ll find that you
can actually pass either one, two, or three arguments to the
application. If you pass a single argument, Asterisk will assume
it’s the destination priority in the current extension. If you pass
two arguments, Asterisk will treat them as the extension and
priority to go to in the current context.
In this example, we’ve passed all three arguments for the sake of clarity, but passing just the extension and priority would have had the same effect.
Handling Invalid Entries and Timeouts
Now that our first voice menu is starting to come together, let’s
add some additional special extensions. First, we need an extension
for invalid entries; when a caller presses an invalid entry (e.g.,
pressing 9 in the above example), the call is sent to the i
extension. Second, we need an extension to
handle situations when the caller doesn’t give input in time (the
default timeout is 10 seconds). Calls will be sent to the t
extension if the caller takes too long to
press a digit after WaitExten()
has
been called. Here is what our dialplan will look like after we’ve
added these two extensions:
[incoming] exten => 123,1,Answer() exten => 123,n,Background(enter-ext-of-person) exten => 123,n,WaitExten() exten => 1,1,Playback(digits/1) exten => 1,n,Goto(incoming,123,1) exten => 2,1,Playback(digits/2) exten => 2,n,Goto(incoming,123,1) exten => 3,1,Playback(digits/3) exten => 3,n,Goto(incoming,123,1) exten => i,1,Playback(pbx-invalid) exten => i,n,Goto(incoming,123,1) exten => t,1,Playback(vm-goodbye)exten => t,n,Hangup()
Using the i
and t
extensions makes our dialplan a little
more robust and user-friendly. That being said, it is still quite
limited, because outside callers have no way of connecting to a live
person. To do that, we’ll need to learn about another application,
called Dial()
.
Using the Dial() Application
One of Asterisk’s most valuable features is its ability to connect
different callers to each other. This is especially useful when
callers are using different methods of communication. For example,
caller A might be communicating over the traditional analog telephone
network, while user B might be sitting in a café halfway around the
world and speaking on an IP telephone. Luckily, Asterisk takes most of
the hard work out of connecting and translating between disparate
networks. All you have to do is learn how to use the Dial()
application.
The syntax of the Dial()
application is a little more complex than that of the other
applications we’ve used so far, but don’t let that scare you off.
Dial()
takes up to four arguments.
The first is the destination you’re attempting to call, which (in its simplest form) is made up of a technology
(or transport) across which to make the call, a forward slash, and the
remote endpoint or resource. Common technology types include Zap (for
analog and T1/E1/J1 channels), SIP, and IAX2. For example, let’s
assume that we want to call a Zap endpoint identified by Zap/1
, which is an FXS channel with an
analog phone plugged into it. The technology is Zap
, and the resource is 1
. Similarly, a call to a SIP device (as
defined in sip.conf) might have a destination of
SIP/Jane
, and a call to an IAX
device (defined in iax.conf) might have a
destination of IAX2/Fred
. If we
wanted Asterisk to ring the Zap/1
channel when extension 123 is reached in the dialplan, we’d add the
following extension:
exten => 123,1,Dial(Zap/1)
We can also dial multiple channels at the same time, by
concatenating the destinations with an ampersand (&
), like this:
exten => 123,1,Dial(Zap/1&Zap/2&SIP/Jane)
The Dial()
application will
ring the specified destinations simultaneously, and bridge the inbound
call with whichever destination channel answers the call first. If the
Dial()
application can’t contact
any of the destinations, Asterisk will set a variable called DIALSTATUS
with the reason that it couldn’t dial the destinations, and
continue on with the next priority in the extension.[76]
The Dial()
application also
allows you to connect to a remote VoIP endpoint not previously defined
in one of the channel configuration files. The full syntax for this
type of connection is:
Dial(technology/user
[:password]
@remote_host
[:port]
[/remote_extension]
)
As an example, you can dial into a demonstration server at Digium using the IAX2 protocol by using the following extension:
exten => 500,1,Dial(IAX2/guest@misery.digium.com/s)
The full syntax for the Dial()
application is slightly different
when dealing with Zap channels, as shown:
Dial(Zap/[gGrR]
channel_or_group
[/remote_extension]
)
For example, here is how you would dial 1-800-555-1212
on Zap channel number
4.
exten => 501,1,Dial(Zap/4/18005551212)
The second argument to the Dial()
application is a timeout, specified
in seconds. If a timeout is given, Dial()
will attempt to call the
destination(s) for that number of seconds before giving up and moving
on to the next priority in the extension. If no timeout is specified,
Dial()
will continue to dial the
called channel(s) until someone answers or the caller hangs up. Let’s
add a timeout of 10 seconds to our extension:
exten => 123,1,Dial(Zap/1,10)
If the call is answered before the timeout, the channels are
bridged and the dialplan is done. If the destination simply does not
answer, is busy, or is otherwise unavailable, Asterisk will set a
variable called DIALSTATUS
and then
continue on with the next priority in the extension.
Let’s put what we’ve learned so far into another example:
exten => 123,1,Dial(Zap/1,10) exten => 123,n,Playback(vm-nobodyavail) exten => 123,n,Hangup()
As you can see, this example will play the vm-nobodyavail.gsm sound file if the call goes unanswered.
The third argument to Dial()
is an option string. It may contain one or more characters that modify
the behavior of the Dial()
application. While the list of possible options is too long to cover
here, one of the most popular options is the m
option. If you place the letter m
as the third argument, the calling party
will hear hold music instead of ringing while the destination channel
is being called (assuming, of course, that music on hold has been
configured correctly). To add the m
option to our last example, we simply change the first line:
exten => 123,1,Dial(Zap/1,10,m)
exten => 123,n,Playback(vm-nobodyavail)
exten => 123,n,Hangup()
Since the extensions numbered 1 and 2 in our dialplan are
somewhat useless now that we know how to use the Dial()
application, let’s replace them with
new extensions that will allow outside callers to reach John and
Jane:
[incoming] exten => 123,1,Answer() exten => 123,n,Background(enter-ext-of-person) exten => 123,n,WaitExten() exten => 1,1,Dial(Zap/1,10) exten => 1,n,Playback(vm-nobodyavail) exten => 1,n,Hangup() exten => 2,1,Dial(SIP/Jane,10) exten => 2,n,Playback(vm-nobodyavail) exten => 2,n,Hangup() exten => i,1,Playback(pbx-invalid) exten => i,n,Goto(incoming,123,1) exten => t,1,Playback(vm-goodbye) exten => t,n,Hangup()
The fourth and final argument to the Dial()
application is a URL. If the
destination channel supports receiving a URL at the time of the call,
the specified URL will be sent (for example, if you have an IP
telephone that supports receiving a URL, it will appear on the phone’s
display; likewise, if you’re using a soft phone, the URL might pop up
on your computer screen). This argument is very rarely used.
Note that the second, third, and fourth arguments may be left blank. For example, if you want to specify an option but not a timeout, simply leave the timeout argument blank, like this:
exten => 1,1,Dial(Zap/1,,m)
Adding a Context for Internal Calls
In our examples thus far, we have limited ourselves to a single
context, but it is probably fair to assume that almost all Asterisk
installations will have more than one context in their dialplans. As
we mentioned at the beginning of this chapter, one important function
of contexts is to separate privileges (such as making long-distance
calls or calling certain extensions) for different classes of callers.
In our next example, we’ll add to our dialplan by creating two
internal phone extensions, and we’ll set up the ability for these two
extensions to call each other. To accomplish this, we’ll create a new
context called [employees]
.
Tip
As in previous examples, we’ve assumed that an FXS analog
channel (Zap/1
, in this case) has
already been configured, and that your
zapata.conf file is configured so that any
calls originated by Zap/1
begin
in the [employees]
context. For a
few examples at the end of the chapter, we’ll also assume that an
FXO Zap channel has been configured as Zap/4
, with calls coming in on this
channel being sent to the [incoming]
context.
We’ve also assumed you have at least one SIP channel (named
SIP/Jane
) that is configured to
originate in the [employees]
context. We’ve done this to introduce you to using other types of
channels.
If you don’t have hardware for the channels listed above (such
as Zap/4
), or if you’re using
hardware with different channel names (e.g., not SIP/Jane
), just change the examples to
match your particular system configuration.
Our dialplan now looks like this:
[incoming] exten => 123,1,Answer() exten => 123,n,Background(enter-ext-of-person) exten => 123,n,WaitExten() exten => 1,1,Dial(Zap/1,10) exten => 1,n,Playback(vm-nobodyavail) exten => 1,n,Hangup() exten => 2,1,Dial(SIP/Jane,10) exten => 2,n,Playback(vm-nobodyavail) exten => 2,n,Hangup() exten => i,1,Playback(pbx-invalid) exten => i,n,Goto(incoming,123,1) exten => t,1,Playback(vm-goodbye) exten => t,n,Hangup() [employees] exten => 101,1,Dial(Zap/1)exten => 102,1,Dial(SIP/Jane)
In this example, we have added two new extensions to the
[employees]
context. This way, the
person using channel Zap/1
can pick
up the phone and dial the person at channel SIP/Jane
by dialing 102. By that same token,
the phone registered as SIP/Jane
can dial Zap/1
by dialing
101.
We’ve arbitrarily decided to use extensions 101 and 102 for our examples, but feel free to use whatever numbering convention you wish for your extensions. You should also be aware that you’re not limited to three-digit extensions; you can use as few or as many digits as you like. (Well, almost. Extensions must be shorter than 80 characters long, and you shouldn’t use single-character extensions for your own use, as they’re reserved.) Don’t forget that you can use names as well, like so:
[incoming] exten => 123,1,Answer() exten => 123,n,Background(enter-ext-of-person) exten => 123,n,WaitExten() exten => 1,1,Dial(Zap/1,10) exten => 1,n,Playback(vm-nobodyavail) exten => 1,n,Hangup() exten => 2,1,Dial(SIP/Jane,10) exten => 2,n,Playback(vm-nobodyavail) exten => 2,n,Hangup() exten => i,1,Playback(pbx-invalid) exten => i,n,Goto(incoming,123,1) exten => t,1,Playback(vm-goodbye) exten => t,n,Hangup() [employees] exten => 101,1,Dial(Zap/1) exten => john,1,Dial(Zap/1) exten => 102,1,Dial(SIP/Jane)exten => jane,1,Dial(SIP/Jane)
It certainly wouldn’t hurt to add named extensions if you think
your users might be dialed via a VoIP protocol such as SIP that
supports dialing by name. You can also see that it is possible to have
different extensions in the dialplan ring the same endpoint. For
example, you could have extension 200 ring SIP/George
, and then have extension 201 play
a prompt of some kind and then ring SIP/George
.
Now that our internal callers can call each other, we’re well on our way toward having a complete dialplan. Next, we’ll see how we can make our dialplan more scalable and easier to modify in the future.
Using Variables
Variables can be used in an Asterisk dialplan to help reduce typing, add clarity, or add additional logic to a dialplan. If you have some computer programming experience, you probably already understand what a variable is. If not, don’t worry; we’ll explain what variables are and how they are used.
You can think of a variable as a container that can hold one
value at a time. So, for example, we might create a variable called
JOHN
and assign it the value of
Zap/1
. This way, when we’re writing
our dialplan, we can refer to John’s channel by name, instead of
remembering that John is using the channel named Zap/1
.
There are two ways to reference a variable. To reference the
variable’s name, simply type the name of the variable, such as
JOHN
. If, on the other hand, you
want to reference its value, you must type a dollar sign, an opening
curly brace, the name of the variable, and a closing curly brace. Here’s how we’d reference the
variable inside the Dial()
application:
exten => 555,1,Dial(${JOHN})
In our dialplan, whenever we write ${JOHN}
, Asterisk will automatically replace
it with whatever value has been assigned to the variable named
JOHN
.
Tip
Note that variable names are case-sensitive. A variable named
JOHN
is different than a variable
named John
. For readability’s
sake, all the variable names in the examples will be written in
uppercase. You should also be aware that any variables set by
Asterisk will be uppercase as well. Some variables, such as CHANNEL
or EXTEN
are reserved by Asterisk. You should
not attempt to set these variables.
There are three types of variables we can use in our dialplan: global variables, channel variables, and environment variables. Let’s take a moment to look at each type.
Global variables
As their name implies, global
variables apply to all extensions in all contexts. Global
variables are useful in that they can be used anywhere within a
dialplan to increase readability and manageability. Suppose for a
moment that you had a large dialplan and several hundred references
to the Zap/1
channel. Now imagine
you had to go through your dialplan and change all of those
references to Zap/2
. It would be
a long and error-prone process, to say the least.
On the other hand, if you had defined a global variable with
the value Zap/1
at the beginning
of your dialplan and then referenced that instead, you would have to
change only one line.
Global variables should be declared in the [globals]
context at the beginning of the
extensions.conf file. They can also be defined
programmatically, using the GLOBAL()
dialplan function.[77] Here is an example of how both methods look inside of
a dialplan. The first shows the setting of a global variable named
JOHN
with a value of Zap/1
. This variable is set at the time
Asterisk parses the dialplan. The second example shows how a global
variable can be set in the dialplan. In this case, the variable
named George
is being assigned
the value of SIP/George
when
extension 124 is dialed in the [employees]
context:
[globals] JOHN=Zap/1 [employees] exten => 124,1,Set(GLOBAL(GEORGE)=SIP/George)
Channel variables
A channel variable is a variable that is associated only with a particular call. Unlike global variables, channel variables are defined only for the duration of the current call and are available only to the channels participating in that call.
There are many predefined channel variables available for use
within the dialplan, which are explained in the
channelvariables.txt file in the
doc subdirectory of the Asterisk source.
Channel variables are set via the Set()
application:
exten => 125,1,Set(MAGICNUMBER=42)
We’ll cover many uses for channel variables in Chapter 6.
Environment variables
Environment variables are a way of accessing Unix environment variables from
within Asterisk. These are referenced using the ENV()
dialplan function. The syntax looks like ${ENV(
var
)}
, where var
is the Unix environment variable you wish to reference. Environment
variables aren’t commonly used in Asterisk dialplans, but they are
available should you need them.
Adding variables to our dialplan
Now that we’ve learned about variables, let’s put them to work in our dialplan. We’ll add global variables for two people, John and Jane:
[globals] JOHN=Zap/1 JANE=SIP/Jane [incoming] exten => 123,1,Answer() exten => 123,n,Background(enter-ext-of-person) exten => 123,n,WaitExten() exten => 1,1,Dial(${JOHN},10) exten => 1,n,Playback(vm-nobodyavail) exten => 1,n,Hangup() exten => 2,1,Dial(${JANE},10) exten => 2,n,Playback(vm-nobodyavail) exten => 2,n,Hangup() exten => i,1,Playback(pbx-invalid) exten => i,n,Goto(incoming,123,1) exten => t,1,Playback(vm-goodbye) exten => t,n,Hangup() [employees] exten => 101,1,Dial(${JOHN}) exten => john,1,Dial(${JOHN}) exten => 102,1,Dial(${JANE}) exten => jane,1,Dial(${JANE})
Pattern Matching
If we want to be able to allow people to dial through Asterisk and have Asterisk connect the caller to an outside resource, we need a way to match on any possible phone number that the caller might dial. Can you imagine how tedious it would be to manually write a dialplan with an extension for every possible number you could dial? Luckily, Asterisk has just the thing for situations like this: pattern matching. Pattern matching allows you to create one extension in your dialplan that matches many different numbers.
Pattern-matching syntax
When using pattern matching, certain letters and symbols represent what we are trying to
match. Patterns always start with an underscore (_
).
This tells Asterisk that we’re matching on a pattern, and not on an
explicit extension name. (This means, of course, that you should
never start your extension names with an underscore.)
Warning
If you forget the underscore on the front of your pattern, Asterisk will think it’s just a named extension and won’t do any pattern matching. This is one of the most common mistakes people make when starting to learn Asterisk.
After the underscore, you can use one or more of the following characters.
X
Z
N
[15-7]
Matches a single digit from the range of digits specified. In this case, the pattern matches a single 1, 5, 6, or 7.
.
(period)Wildcard match; matches one or more characters, no matter what they are.
Warning
If you’re not careful, wildcard matches can make your dialplans do things you’re not expecting (like matching built-in extensions such as
i
orh
). You should use the wildcard match in a pattern only after you’ve matched as many other digits as possible. For example, the following pattern match should probably never be used:_.
In fact, Asterisk will warn you if you try to use it. Instead, use this one, if at all possible:
_X.
- ! (bang)
Wildcard match; matches zero or more characters, no matter what they are.
To use pattern matching in your dialplan, simply put the pattern in the place of the extension name (or number):
exten => _NXX,1,Playback(auth-thankyou)
In this example, the pattern matches any three-digit extension
from 200 through 999 (the N
matches any digit between 2 and 9, and each X
matches a digit between 0 and 9). That
is to say, if a caller dialed any three-digit extension between 200
and 999 in this context, he would hear the sound file
auth-thankyou.gsm.
One other important thing to know about pattern matching is
that if Asterisk finds more than one pattern that matches the dialed
extension, it will use the most specific one
(going from left to right). Say you had defined the following two
patterns, and a caller dialed 555-1212
:
exten => _555XXXX,1,Playback(digits/1) exten => _55512XX,1,Playback(digits/2)
In this case the second extension would be selected, because it is more specific.
Pattern-matching examples
Before we go on, let’s look at a few more pattern-matching examples. In each one, see if you can tell what the pattern would match before reading the explanation. We’ll start with an easy one:
_NXXXXXX
This pattern would match any seven-digit number, as long as the first digit was two or higher. This pattern would be compatible with any North American Numbering Plan local seven-digit number. In areas with 10-digit dialing, that pattern would look like this:
_NXXNXXXXXX
Note that neither of these two patterns would handle long distance calls. We’ll cover those shortly.
Let’s try another:
_1NXXNXXXXXX
This one is slightly more difficult. This would match the number 1, followed by an area code between 200 and 999, then any 7-digit number. In the NANP calling area, you would use this pattern to match any long-distance number.[78]
Now for an even trickier example:
_011.
If that one left you scratching your head, look at it again. Did you notice the period on the end? This pattern matches any number that starts with 011 and has at least one more digit. In the NANP, this indicates an international phone number. (We’ll be using these patterns in the next section to add outbound dialing capabilities to our dialplan.)
Using the ${EXTEN} channel variable
We know what you’re thinking… You’re sitting there asking yourself, “So what happens if I want to
use pattern matching, but I need to know which digits were actually
dialed?” Luckily, Asterisk has just the answer. Whenever you dial an
extension, Asterisk sets the ${EXTEN}
channel variable to the digits
that were dialed. We can use an application called SayDigits()
to test it out:
exten => _XXX,1,SayDigits(${EXTEN})
In this example, the SayDigits()
application will read back to you the three-digit extension you
dialed.
Often, it’s useful to manipulate the ${EXTEN}
by stripping a certain number of
digits off the front of the extension. This is accomplished by using
the syntax ${EXTEN:
x
}
, where x
is
where you want the returned string to start, from left to right. For
example, if the value of EXTEN
is
95551212, ${EXTEN:1}
equals
5551212. Let’s take a look at another example:
exten => _XXX,1,SayDigits(${EXTEN:1})
In this example, the SayDigits()
application would start at the
second digit, and thus read back only the last two digits of the
dialed extension.
Enabling Outbound Dialing
Now that we’ve introduced pattern matching, we can go about the process of allowing users to make
outbound calls. The first thing we’ll do is add a variable to the
[globals]
context to define which channel will be used for outbound
calls:
[globals]
JOHN=Zap/1
JANE=SIP/Jane
OUTBOUNDTRUNK=Zap/4
Next, we will add contexts to our dialplan for outbound dialing.
You may be asking yourself at this point, “Why do we need separate contexts for outbound calls?” This is so that we can regulate and control which callers have permission to make outbound calls, and which types of outbound calls they are allowed to make.
To begin, let’s create a context for local calls. To be
consistent with most traditional phone switches, we’ll put a 9
on the front of our patterns, so that
users have to dial 9 before calling an outside number:
[outbound-local] exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9NXXXXXX,n,Congestion() exten => _9NXXXXXX,n,Hangup()
Tip
Note that dialing 9 doesn’t actually give you an outside line, unlike with many traditional PBX systems. Once you dial 9 on an analog line, the dial tone will stop. If you’d like the dial tone to continue even after dialing 9, add the following line (right after your context definition):
ignorepat => 9
This directive tells Asterisk to continue to provide a dial tone on an analog line, even after the caller has dialed the indicated pattern. This will not work with VoIP telephones, as they usually don’t send digits to the system as they are input; they are sent to Asterisk all at once. Luckily, most of the popular VoIP telephones can be configured to emulate the same functionality.
Let’s review what we’ve just done. We’ve added a global variable
called OUTBOUNDTRUNK
, which simply
defines the channel we are using for outbound calls.[79] We’ve also added a context for local outbound calls. In
priority 1, we take the dialed extension, strip off the 9 with the
${EXTEN:1}
syntax, and then attempt
to dial that number on the channel signified by the variable OUTBOUNDTRUNK
. If the call is successful,
the caller is bridged with the outbound channel. If the call is
unsuccessful (because either the channel is busy or the number can’t
be dialed for some reason), the Congestion()
application is called, which
plays a “fast busy signal” (congestion tone) to let the caller know
that the call was unsuccessful.
Before we go any further, let’s make sure our dialplan allows outbound emergency numbers:
[outbound-local] exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9NXXXXXX,n,Congestion() exten => _9NXXXXXX,n,Hangup() exten => 911,1,Dial(${OUTBOUNDTRUNK}/911) exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911) ; So that folks who dial “9” ; first will also get through
Again, we’re assuming for the sake of these examples that we’re inside the United States or Canada. If you’re outside of this area, please replace 911 with the emergency services number in your particular location. This is something you never want to forget to put in your dialplan!
Next, let’s add a context for long-distance calls:
[outbound-long-distance] exten => _91NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _91NXXNXXXXXX,n,Playtones(congestion) exten => _91NXXNXXXXXX,n,Hangup()
Now that we have these two new contexts, how do we allow internal users to take advantage of them? We need a way for contexts to be able to use the functionality contained in other contexts.
Includes
Asterisk has a feature that enables us to use the extensions
from one context within another context via the include
directive. This is used to control
access to different sections of the dialplan. We’ll use the include
functionality to allow users in our [employees]
context the ability to make
outbound phone calls. But first, let’s cover the syntax.
The include
statement takes
the following form, where context
is the
name of the remote context we want to include in the current
context:
include => context
When we include other contexts within our current context, we have to be mindful of the order in which we are including them. Asterisk will first try to match the dialed extension in the current context. If unsuccessful, it will then try the first included context (including any contexts included in that context), and then continue to the other included contexts in the order in which they were included.
As it sits, our current dialplan has two contexts for outbound
calls, but there’s no way for people in the [employees]
context to use them. Let’s
remedy that by including the two outbound contexts in the [employees]
context, like this:
[globals] JOHN=Zap/1 JANE=SIP/Jane OUTBOUNDTRUNK=Zap/4 [incoming] exten => 123,1,Answer() exten => 123,n,Background(enter-ext-of-person) exten => 123,n,WaitExten() exten => 1,1,Dial(${JOHN},10) exten => 1,n,Playback(vm-nobodyavail) exten => 1,n,Hangup() exten => 2,1,Dial(${JANE},10) exten => 2,n,Playback(vm-nobodyavail) exten => 2,n,Hangup() exten => i,1,Playback(pbx-invalid) exten => i,n,Goto(incoming,123,1) exten => t,1,Playback(vm-goodbye) exten => t,n,Hangup() [employees] include => outbound-local include => outbound-long-distance exten => 101,1,Dial(${JOHN}) exten => john,1,Dial(${JOHN}) exten => 102,1,Dial(${JANE}) exten => jane,1,Dial(${JANE}) [outbound-local] exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9NXXXXXX,n,Congestion() exten => _9NXXXXXX,n,Hangup() exten => 911,1,Dial(${OUTBOUNDTRUNK}/911) exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911) [outbound-long-distance] exten => _91NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _91NXXNXXXXXX,n,Playtones(congestion) exten => _91NXXNXXXXXX,n,Hangup()
These two include
statements
make it possible for callers in the [employees]
context to make outbound calls.
We should also note that for security’s sake you should always make
sure that your [inbound]
context
never allows outbound dialing. (If by chance it did, people could dial
into your system and then make outbound toll calls that would be
charged to you!)
Conclusion
And there you have it—a basic but functional dialplan. It’s not exactly fully featured, but we’ve covered all of the fundamentals. In the following chapters, we’ll continue to add features to this foundation.
If parts of this dialplan don’t make sense, you may want to go back and re-read a section or two before continuing on to the next chapter. It’s imperative that you understand these principles and how to apply them, as the next chapters build on this information.
[70] Please note that the space is conspicuously absent from the list of allowed characters. Don’t use spaces in your context names—you won’t like the result!
[71] Asterisk permits simple arithmetic within the priority,
such as n+200
or the priority
s
(for same), but their usage
is considered to be an advanced topic. Please note that
extension s
and priority
s
are two distinct
concepts.
[72] There is nothing special about any context name. We could
have named this context [stuff_that_comes_in
], and as long as
that was the context assigned in the channel definition in
sip.conf, iax.conf,
zaptel.conf, et al., the channel would enter
the dialplan in that context. Having said that, it is strongly
recommended that you give your contexts names that help you to
understand their purpose. Some good context names might include
[incoming
], [local_calls
], [long_distance
], [sip_telephones
], [user_services
], [experimental
], [remote_locations
], and so forth. Always
remember that a context determines how a channel enters the
dialplan, so name accordingly.
[73] Asterisk selects the best file based on translation
cost―that is, it selects the file that is the least CPU-intensive
to convert to its native audio format. When you start Asterisk, it
calculates the translation costs between the different audio
formats (they often vary from system to system). You can see these
translation costs by typing show
translation
at the Asterisk command-line interface. The
numbers shown represent how many milliseconds it takes Asterisk to
transcode one second of audio. We’ll cover more about the
different audio formats (known as codecs) in
Chapter 8.
[74] In fact, if you don’t have any channels configured, now is the time to do so. There is a real satisfaction that comes from passing your first call into an Asterisk system that you built from scratch. People get this funny grin on their face as they realize that they have just created a telephone system. This pleasure can be yours as well, so please, don’t go any further until you have made this little dialplan work.
[75] It should be noted that some people expect that Background()
, due to its name, would
continue in the dialplan while the sound is being played, but its
name refers to the fact that it is playing a sound in the
background, while waiting for DTMF in the foreground.
[76] Don’t worry, we’ll cover variables (in Using Variables”) and show you how to have
your dialplan make decisions based on the value of this DIALSTATUS
variable.
[77] Don’t worry! We’ll cover dialplan functions in the Dialplan Functions” section.
[78] If you grew up in North America, you may believe that the 1 you dial before a long distance call is “the long distance code.” This is incorrect. The number 1 is in fact the international country code for all countries in NANP. Keep this in mind if you ever send your phone number to someone in another country. They may not know what your country code is, and thus be unable to call you with just your area code and phone number. Your full phone number with country code should be printed as +1 NPA NXX XXXX (where NPA is your area code)―e.g., +1 416 555 1212.
[79] The advantage of this is that if one day we decide to send
all of our calls through some other channel, we have to edit the
channel name assigned to the variable OUTBOUNDTRUNK
only in the [globals]
context, instead of having to
manually edit every reference to the channel in our
dialplan.
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