 Asterisk: The Open Source PBX Solution
Adam Olson
Systems and network administrators typically deal with
data and functionality such as email communications, Web and database
applications, network infrastructure, and network performance. Analog and
voice may be part of the picture, but these tasks are typically handled by
concentrated DSP cards that, in the end, hand the call off to internal data
networks in the form of dial-up users or the like.
Today, more systems and network administrators are
managing voice communications traffic in addition to their other duties.
These tasks include managing private branch exchange (PBX) extensions,
Session Initiation Protocol (SIP) phones, Interactive Voice Response (IVR)
prompts, voicemail-to-email functionality, call forwarding, etc. Many of us have been waiting for a production grade
solution to handle voice traffic inside the corporate network that offers
an alternative to hardwired jacks, fixed desk locations, and remote staff
members.
This is where Asterisk comes in, and it really does
the job well. Asterisk is an open source software package that will turn
almost any Linux, BSD, or Mac OSX box into a full-featured PBX. Public
Switched Telephone Network (PSTN) calls are terminated via quad RJ-11 or
PRI PCI cards, allowing administrators to choose between Voice over IP
(VoIP) providers or maintaining their existing PSTN provider contracts.
Asterisk
Mark Spencer created Asterisk, and he is also known
for creating a few other packages, such as the popular chat client GAIM.
Asterisk is capable of using only Internet Telephony Service Providers
(ITSPs), also known as VoIP providers, in addition to the more traditional
PSTN for handling incoming and outgoing calls.
Because Asterisk is an open source solution, a number
of additional packages and feature add-ons are available online. These
include a complete set of high-quality voice recordings for IVR prompts, a
Web-based management interface, back-end processing engines, etc.
Requirements
You don't need any additional hardware if you
are looking solely for a solution with SIP phones and an ITSP provider for
inbound and outbound calling and you have at least one SIP phone and a
server. If you want to utilize your existing PRIs or analog phone circuits,
visit:
http://www.digium.com
Check out the PCI PRI or 4 Port FXO/FXS cards to meet your exact needs.
The examples in this article are based on Red Hat
Fedora Core and Asterisk version 1.2.7.1 built from source. Make sure to
download the latest version from:
http://www.asterisk.org/download
when you attempt the build process. Please consult the
Asterisk Web site for more details regarding platform support if you are
using a different operating system. I will be using an ITSP for outbound
phone service and will be terminating DIDs over a local PRI, using a PCI
card supplied by Digium.
Regarding phones, X-Lite makes a nice, free, soft SIP
phone that can be downloaded from their site. This is a good way to get
started without purchasing a hard SIP phone right away. You can also order
network adapters online that will allow your existing phones to function.
However, using hard SIP phones is nice when possible, because they are
typically Web managed and provide administrators much appreciated
flexibility when supporting them remotely.
The Build Process
Asterisk compiles quickly and easily. You should also
know that there is a version of Asterisk, called Asterisk@Home, which is
basically a preconfigured version of Asterisk with the Asterisk Management
Portal (AMP) included. After putting the ISO image on a CD and booting it,
your hardware will be rebuilt as an Asterisk@Home PBX. Again, this article
focuses on building Asterisk from source, which allows you to keep
preexisting services running.
Download the most updated version of the packages listed below from:
http://www.asterisk.org/download
Include the following:
asterisk-1.2.7.1.tar.gz
asterisk-perl-0.08.tar.gz
asterisk-addons-1.2.2.tar.gz
asterisk-sounds-1.2.1.tar.gz
Uncompress all the files with:
# gzip -d *
To compile Asterisk, type:
# tar xf asterisk-1.2.7.1.tar
# cd asterisk-1.2.7.1
# make ; make install ; make samples
To compile Asterisk add-ons, type:
# tar xf asterisk-addons-1.2.2.tar
# cd asterisk-addons-1.2.2
# make ; make install
To compile Asterisk Perl, type:
# tar xf asterisk-perl-0.08.tar
# cd asterisk-perl-0.08
# perl Makefile.PL ; make install
To add the additional audio files, type:
# tar xf asterisk-sounds-1.2.1.tar
# cd asterisk-sounds-1.2.1
# cd sounds
# cp -pr * /var/lib/asterisk/sounds/
If you run into any errors, please consult the
Asterisk documentation for assistance. I haven't seen Asterisk fail
to compile on any supported platform that I've tested so far.
In addition to the Asterisk package and a few add-ons,
which should now be installed, you will want to make sure you have mpg123
installed. This program is used to play mp3s, for example, when callers are
placed on hold.
If you do not have mpg123 installed, download it,
uncompress it, and type:
# tar xf mpg123-version.tar
# cd mpg123-version
# make linux ; make install
Configuration
By default, the primary configuration directory is
/etc/asterisk. Asterisk looks here for a number
of configuration files like sip.conf, extensions.conf, asterisk.conf, etc.
Let's take a look at each of the more important configuration files
now. I will provide trimmed down examples that you can use to get started.
Please note that Asterisk frequently deals within a "context",
and contexts start when a name is read between brackets.
Asterisk requires a reload when configuration changes
are made. There are ways to selectively restart
portions of Asterisk from the command line, but in this article, we will be
restarting Asterisk completely. To fire up
Asterisk with verbose output for testing, run asterisk
-cvvvv from the command line. When you want to
restart Asterisk after a change, hit Control-C and run asterisk -cvvvv again. You will
encounter configuration problems, and when these occur, the console output
is extremely helpful. You can run Asterisk in the background when you are
ready by simply typing asterisk on the command line.
sip.conf
The sip.conf file is where you define your SIP phones.
For example, below you will see a general, default context and an entry for
extension 1000:
[general]
context=default ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default
; is yes, this can also be set to 'osp'
realm=yourdomain.com ; Realm for digest authentication
; Realms MUST be globally unique
; according to RFC 3261
; Set this to your host name or domain
; name
bindport=5060 ; UDP Port to bind to (SIP standard
; port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
; to all)
disallow=all ; Disallow all codecs
allow=ulaw ; Use ulaw codec
callerid = Unknown
[1000]
type=friend ; either "friend" (peer+user), "peer" or
; "user"
context=from-sip
callerid=Joe Schmoe <1000>
secret=petname
host=dynamic ; we have a static but private IP address
nat=yes ; there is not NAT between phone and
; Asterisk
dtmfmode=inband ; either RFC2833 or INFO for the
; BudgeTone
mailbox=1000@default ; mailbox 1234 in voicemail context
; "default"
disallow=all ; need to disallow=all before we can use
; allow=
allow=ulaw ; Note: In user sections the order of
; codecs
; listed with allow= does NOT matter!
For a full list of available values, please see:
http://www.digium.com/en/docs/asterisk_handbook/sip.conf.html
The following excerpt from the reference documentation
is helpful:
The sip.conf file is read from the top down. The first
section is for general server options, such as the IP address and port
number to bind to. The following sections define client parameters such as
the username, password, and default IP address for unregistered clients.
Sections are delineated by a name in brackets. The first section is called
general (which cannot be used as a client name.) The following sections
begin with the client name in brackets, followed by the client options.
We'll focus more on the context value later when
we begin defining the dial plan in extensions.conf. The most important
parameters are:
type=friend -- Allows each SIP phone to receive and initiate phone calls.
dtmfmode=inband -- Relates to how tones are registered for each SIP phone.
mailbox=1000@default -- States where to send calls to voicemail.
disallow=all -- Reject all codec types.
allow=ulaw -- Force the ulaw codec; this can be set to a number of codecs.
The other options should be fairly self-explanatory,
and the reference documentation listed above addresses each option as well.
iax.conf
The iax.conf file is where you define ITSP accounts
for inbound and outbound voice service, among other things. I will focus on
outbound only, as a PRI is going to handle incoming calls. If you are
interested in using an ITSP for inbound, check out Junction Networks as an
option. They provide easy-to-implement documentation about using their
service with Asterisk.
We don't need to add much to iax.conf for
outbound service; we will add more to extensions.conf that really makes it
work. For this example, set up an account with VoipJet and follow their
instructions. VoipJet provides a chunk of free minutes, which is helpful
when in test mode. In the end, you will add something like this to iax.conf:
[voipjet]
type=peer
host=64.34.45.100
secret=nkd38sne89d78fc
auth=md5
context=default
For a full list of available values, please reference:
http://www.digium.com/en/docs/asterisk_handbook/iax.conf.html
The following excerpts from the reference documentation and my own additions describe the example values above:
- peer -- A peer receives calls from
the Asterisk server, but does not place them. (This is what we want as we
are only making outbound calls via VoipJet.)
- host -- The IP address of your ITSP.
- secret -- Your md5 value.
- auth -- The type of secret being used.
- context -- How the context is referenced in extensions.conf, if desired.
extensions.conf
Over time, you will likely become most familiar with
the extensions.conf file. Most of the logic behind handling call flow is
managed here. Because of this, let's look a little bit more at how
the configuration is parsed. I'm not going to cover all the syntax as
there are quite a few options.
Recalling sip.conf, you'll remember we added a
SIP phone at extension 1000. Until it is referenced in extensions.conf, the
phone will be nonfunctional, as Asterisk doesn't know when to send a
call to it. Extension to extension dialing will also not work. Add the
following to extensions.conf:
[extensions]
exten => 1000,1,Dial(Sip/${EXTEN}|15|t)
exten => 1000,2,Voicemail(u1000@default)
exten => 1000,3,Hangup
exten => 1000,hint,Sip/1000
We've now defined extension 1000 and told
Asterisk how to dial it. Pay special attention to the order value, the
field after the first comma. This field tells Asterisk in which order to
process the configuration block for 1000. So, the first line tells Asterisk
to dial the SIP phone registered at extension 1000 for 15 seconds and to
play back a traditional ring tone while dialing (Asterisk can be configured
to play music while dialing extensions as well). If there is no answer
after 15 seconds, the second line sends the call to the voicemail box of
extension 1000. The third line then simply hangs up the line.
This is nice so far, but how exactly do calls end up
referencing the extensions context? Look at the sip.conf file again, and
you'll see that we placed extension 1000 inside the context of
"from-sip". Make sure the following exists in extensions.conf:
[from-sip]
include => extensions
include => eleven-digit
This context definition tells Asterisk that when a SIP
phones dials a number, it should first check the extensions context and
then check the 11-digit context. You can add as many contexts as you like;
these are only examples. If the dialed number matches a rule within a
context, that action is then taken. Our 11-digit context looks like this:
[eleven-digit] ;11 digit dialing
exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=7078222005)
exten => _1NXXNXXXXXX,n,Dial(IAX2/9142@voipjet/1${EXTEN}||T)
These rules tell Asterisk to match any 11-digit
pattern starting with a 1. The caller ID is set to 7078222005 and VoipJet
is then used to place the outgoing call.
You'll want to define another SIP phone and test
extension-to-extension dialing after adding a configuration for both phones
to sip.conf, extensions.conf, and restarting Asterisk. You can also try
dialing 11-digit numbers now from a registered SIP phone.
Inbound calls over the PRI are also defined in
extensions.conf. Regarding setting up the hardware and
installing/configuring the card, please reference Digium's
documentation. We will place the configuration information for calls coming
in over the PRI under the context default in extensions.conf. The
configuration will look like:
[default] ; For the PRI
exten => 8222005,1,Answer
exten => 8222005,n,Wait(1)
;exten => 8222005,n,GotoIfTime(9:00-18:00|mon-fri|*|*?,,CLOSED)
exten => 8222005,n,Goto(ivr,s,1)
exten => 8222005,n(CLOSED),Macro(closed)
These rules tell Asterisk that when a call comes in on
8222005, which is the DID of the PRI, it should make sure it is within
business hours and then send the call over to the IVR menu for further call
routing.
IVR Configuration
IVR menus can be complicated and spiffy, but I want
to briefly cover some of the basics. The IVR configuration, like many other
functions, is handled inside of extensions.conf. The context
"ivr" was referenced above for calls coming in on the PRI, so
we need the following in extensions.conf:
[ivr]
exten => s,1,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=5)
exten => s,n,Background(custom/aa) ;thanks for calling
;press 1, 2, etc
exten => s,n,WaitExten(5)
exten => s,n,Background(silence/4)
exten => s,n,Goto(ivr,0,1)
exten => 0,1,Background(silence/1)
exten => 0,n,Dial(SIP/1000|10|t)
exten => 0,n,Dial(SIP/1001|10|t)
exten => 0,n,VoiceMail(u1000) ; Send the call to voicemail
; if nobody answers
exten => 0,n,Hangup
These rules comprise a basic example of how to get
started with IVR prompts. For more information on how to actually record
the audio for playback on the menu, please check out the online
documentation. In this example, we have defined a general operator
extension that dials two SIP extensions consecutively. Extension 1000 is
dialed for 10 seconds and then extension 1001, if 1000 does not answer the
call. If both extensions ring for 10 seconds without an answer, the call is
sent to voicemail. The voicemail box is defined within the parenthesis, and
in this example the call is sent to the voicemail box of extension 1000.
Voicemail Configuration
We've looked at how to set up a SIP phone, how
to handle basic call management and IVR prompts, but we still haven't
told Asterisk how to handle calls when we attempt to forward them to the
voicemail system. Voicemail configuration is handled in voicemail.conf, and
the file needs to include:
[default]
1000 => 1000,Joe Schmoe,joeschmoe@yourdomain.com
Asterisk will now be able to associate a name and
email address to the voicemail box for extension 1000. If you have properly
defined the mailer in voicemail.conf, all voicemail messages will be
recorded and sent via email to the recipient's email address.
Conclusion
Asterisk is a great open source PBX solution that is
easy to install and straightforward to maintain once you get the hang of
the configuration files. With an ever-increasing number of staff members
working remotely, it also helps to be able to get inside the PBX and
configure it to your own specifications, instead of relying on a vendor
with a closed PBX system.
Resources
Asterisk Web site -- http://www.asterisk.org
Great VoIP Resource -- http://www.voip-info.org
Adam Olson lives in Northern California. He's
been active in network design, systems administration, and systems
programming for more than 10 years with various companies like MCI
WorldCom, small Bay Area startups, and a California-based ISP. He recently
co-founded a company, called Office Appliance (http://officeappliance.com), which serves the needs
of small- and medium-sized businesses.
|