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Asterisk VoIP Private Branch Exchange

Conference Paper · April 2009

DOI: 10.1109/MSPCT.2009.5164214 · Source: IEEE Xplore

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IMPACT-2009

Asterisk VoIP Private Branch Exchange

Ale Imran1, Mohammed A Qadeer2, M J R Khan1
1
Department of Electronics Engineering, Aligarh Muslim University, Aligarh, India
2
Department of Computer Engineering, Aligarh Muslim University, Aligarh, India
{aleimran, maqadeer, mjrkhan}@zhcet.ac.in

Abstract-This paper intends to present some important As we know that a PBX (Private Branch Exchange) is a
theoretical and practical results that we faced during setting private telephone network used within a company and it handles
up a VoIP (Voice over Internet Protocol) server with the an organization’s voice and data communications [1]. In our
well known open source VoIP server Asterisk. For a fully experimental setup we tried to establish a PBX system that
functional voice exchange we require to set up a server works with incoming and outgoing voice calls. We used PC to
based on Asterisk, connecting clients to the server with the PC communication for simulating the whole task, however IP
help of soft/hard phones and then comes the configuration phones could also be used in the place of PC’s. Here in our
aspects of the soft phones with the server. Here in our implementation, though we have not yet done the connection of
implementation we have connected the clients to the server our server with the conventional PSTN, though it could be done
with the help of SIP protocols. with the help of PCI cards like for example Digium’s
TDM400P to validate the connectivity with the existent circuit
Keywords-VoIP, Asterisk, PBX, IAX, SIP, Trunks. switched network.

1. INTRODUCTION ASTERISKPBX
Host: Home
Phone Line connected to FXOcard
PSTN
SERVERA
The term VoIP stands for Voice over Internet Protocol.
VoIP originated in mid 90’s, when hobbyists began to realize SIP CLIENTS@School

the potential of sending voice data packets over the internet SIPCLIENTS@Home
4001@school

rather than communicating through standard telephone systems. 3001@home

The idea is to use the internet as a telephone network with some INTERNET
additional capabilities. VoIP converts the voice signal from a
telephone into a digital signal, sends it through the internet, and 3002@home
4002@school

then converts it back at the other end.

2001
1001 1002

ASTERISKPBX
Analog phones connected to FXScard
Host: School
2002
SERVERB

Fig 1: Setting up of a VoIP call Analog phones connected

to FXS card

When we are using a PSTN line, we typically pay the Fig 2: Experimental setup
charges according to the time usages. In addition we couldn’t
VoIP uses the Internet Protocol to transmit voice as packets
talk to more than one person at a time. However with VoIP
over an IP network. So VoIP can be achieved on any data
mechanism we can talk all the time, with every person we want,
network that uses IP like the Internet, Intranets and Local area
as far as we want and in addition we can talk with many people
networks (LAN).Here the voice signal is digitized, compressed
at the same time. If we are still not convinced we can consider
and converted to IP networks and transmitted over IP networks
that while we re talking we can exchange data with the people
[2].
we are talking with.

978-1-4244-3604-0/09/$25.00 ©2009 IEEE 217

IMPACT-2009

2. ASTERISK AS A VOICE EXCHANGE between various users and automated tasks. The Switching Core
transparently connects callers arriving on various hardware and
Asterisk is a complete PBX in software written in C software interfaces.
programming language and it runs on Linux operating systems. Application Launcher launches applications which perform
Asterisk does voice over IP in many protocols, and can services for uses, such as voicemail, file playback, and directory
interoperate with almost all standards-based telephony listing.
equipment using relatively inexpensive hardware like for ex Codec Translator uses codec modules for the encoding and
PCI cards [3]. Asterisk in fact creates a PBX that rivals the decoding of various audio compression formats used in the
functionalities of traditional telephone based systems. telephony industry. A number of codecs are available to suit
diverse needs and arrive at the best balance between audio
quality and bandwidth usage.
Scheduler and I/O Manager handles low-level task
scheduling and system management for optimal performance
under all load conditions.

Fig 3: Model of Asterisk Based Exchange

The benefits associated with an Asterisk based voice exchange

could be summarized as:

• Low implementation cost

• Working on TCP/IP protocol
• PBX with enhanced features
• Low Maintenance required
Fig 4: Asterisk’s Architecture
• Convergence of Voice ,Video, Data on a single
connection
2.2. Asterisk’s Services
• Easy to add or remove additional extensions.
VoIP generally uses two types of protocols :1)Signaling
Asterisk does PBX switching, CODEC translation and
Protocols-for setting up a conversation 2) Media transfer
various other applications like voicemail, conference bridging,
protocols for actual transfer of data, once the connection has
IVR and various others.
been set. Session Initiation Protocol (SIP) is an application-
layer control (signaling) protocol for creating, modifying, and
2.1. Architecture of Asterisk Based PBX
terminating sessions with one or more participants. These
sessions include Internet telephone calls, multimedia
Asterisk is carefully designed for maximum flexibility.
distribution, and multimedia [5]. SIP has the following features:
Specific APIs are defined around a central PBX core system.
This advanced core handles the internal interconnection of the
PBX, cleanly abstracted from the specific protocols, codecs, • Lightweight, in that SIP has only six methods,
and hardware interfaces from the telephony applications. This reducing complexity.
allows Asterisk to use any suitable hardware and technology • Transport-independent, because SIP can be used with
available now or in the future to perform its essential functions, UDP, TCP, ATM & so on.
connecting hardware and applications. • Text-based, allowing for humans to read SIP
messages.
The Asterisk core handles these items internally [4]
PBX Switching The essence of Asterisk, of course, is a Private Firewalls typically block media packet types such as UDP,
Branch Exchange Switching system, connecting calls together though one way around this is to use TCP tunneling and relays

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IMPACT-2009

for media in order to provide NAT and firewall traversal. One Contains all of asterisk’s sound files for playback and pre-
solution involves tunneling the media packets within TCP or loaded applications (eg: VoiceMail).
HTTP packets to a relay. This solution uses additional • /var/lib/asterisk/agi-bin
functionality in conjunction with SIP, and packages the media Contains all of asterisk’s AGI scripts and AGI logic.
packets into a TCP stream which is then sent to the relay. The
relay then extracts the packets and sends them on to the other For our experimental setup we configured the SIP and the
endpoint. If the other endpoint is behind a symmetrical NAT or Extensions at the following:
corporate firewall that does not allow VOIP traffic, the relay SIP: /etc/asterisk/sip.conf
would transfer the packets to another tunnel. Extensions: /etc/asterisk/extensions.conf.
For setting up a client on SIP client on Asterisk we do the
following:

;[phone1(ale)]
;type=friend
;secret=2222
;auth=md5
;host=dynamic
;reinvite=no
;canreinvite=no
;qualify=1000
;dtmfmode=inband;callerid="ale"<2222>
;disallow=all
;allow=gsm
Fig 5: SIP allowing
;context=incoming.

3. IMPLEMENTATION OF THE VOICE EXCHANGE

For configuring Asterisk as a voice exchange, the administrator

must create [6]:

• Dial Plan to make Asterisk respond to users through

their devices.
• Devices that allow Asterisk to communicate through a
voice path that uses that channel.

Asterisk is controlled by editing a series of configuration Fig 6: The sip.conf file

files. Users connecting to asterisk all belong to a specific
context (specified in the channel configuration file), which is
where asterisk looks for advice on how to handle the calls The other being extensions.conf, where the administrator
placed by that user, checking the access rights to expensive defines what actions Asterisk will take when calls are answered.
lines, with different rule sets for local users and contacts calling A native language is used to define contexts, extensions, and
from an outside line. actions. Each context defines where a device starts its dial plan,
• /etc/asterisk and therefore restricts what extensions the device may access.
Contains all of asterisk configuration files and logic Extensions are written within contexts, and consist of numbered
information. lines, each line performing either logic on known variables to
• /usr/lib/asterisk/modules the dial plan, or executing one of many applications available in
Contains all of asterisk’s loadable modules, operating Asterisk.
asterisk functionality. For editing the Extensions configuration file
Applications, channels and resources are located in this /etc/asterisk/extensions.conf
directory. [inbound-from-sip];
• /var/lib/asterisk/sounds Our context for SIP clients

219
IMPACT-2009

exten => extension no, priority, application (argl,arg2,...) less prone to viruses, worms and hackers. As far as future work
exten => 1111,1,Dial(SIP/${EXTEN}) is concerned, we would like to work on connecting our Asterisk
exten => 2222,1,Dial(SIP/${EXTEN}) PBX with the conventional circuit switched networks with the
exten => 3333,1,Dial(SIP/${EXTEN}) help of PCI cards like for example Digium’s TDM400P[7] .
exten => 4444,1,Dial(SIP/${EXTEN})

REFRENCES

[1]Andre du Toit “Private PBX networks-Cost effective communication

solutions” in IEEE,1992.

[2]Guo Fang Mao,Alex Talevski,Elizabeth Chang, “Voice over Internet

Protocol on mobile devices” in ICIS 2007.

[3]Md. Zaidul Alam, Saugata Bose, Md. Mhafuzur Rahman, Mohammad

Abdullah Al-Mumin, “Small office PBX using Voice over IP” in ICACT I2-14
Fig 7 : Extensions.conf file FEB,2007

[4]Ryosuke Yamamoto,,Fumikazu Iseki,Moo Wan Kim, “Validation of Voip

Once we are done with this, we need to concentrate on the system for University Network” in ICACT,feb 2008.
installation and registration of the soft phones that we are going
[5] Asterisk.org, "Features and Architecture of Asterisk PBX",
to use at the client end. http://www.asterisk.org/features, accessed in March, 2006.

[6]Taemoor Abbasi, Shekhar Prasad , Nabil Seddigh, Ioannis Lambadaris, “A

comparative study of the SIP & IAX voice protocols” in CCECE/CCGEI,
Saskatoon, May 2005

[7]Anand Gorti, “A fault tolerant VoIP implementation based on open

standards” in EDCC in 2006.

Fig 8 : X-lite softphone at the client’s end,along with its configuration

4. CONCLUSION AND FUTURE WORK

We expect that design and implementation presented in this

paper will be a valuable developing guide for similar kind of
operations. Asterisk based voice exchange provides us with a
much better alternative solution. Its not only cost effective but
also provides us with various features which we generally don’t
get with the conventional circuit switched based PBX
.Moreover the system also provides for unlimited expansion and
since it runs on a secure operating system like LINUX, its much

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