2008 International Conference on Computer and Electrical Engineering
Asterisk Voice Exchange: An Alternative to Conventional EPBX
Mohammed A Qadeer
Department of Computer Engineering,
Aligarh Muslim University, Aligarh, India
maqadeer@zhcet.ac.in
Abstract
This paper highlights the design and
implementation aspects of a VoIP based Asterisk voice
exchange, Developing a fully functional voice
exchange requires to set up a server based on Asterisk,
connecting clients to that server with the help of soft
phones and then configuring the softphones with the
server. Here in our implementation we have connected
the clients to the server with the help of IAX protocols.
The first part of the paper contains some introductory
concepts about VoIP, followed by Asterisk’s internal
architecture. In the third part of the paper we discuss
about the codecs and protocols used by the packet
switching based PBX networks and finally we brush up
about the design and implementation aspects.
Keywords-VoIP, PBX, IAX, SIP, Asterisk, Trunk.
I. Introduction
The development of Asterisk based Voice
Exchange and its applications needs to take into
account the growing complexity with the existing PBX
(private branch exchange) networks [1]. A promising
solution is the voice exchange based on Asterisk,
which works on VoIP [2] (voice over internet
protocols) and provides a more promising and flexible
solution. VoIP technology originated in about 1995,
when hobbyists began to realize the potential of
sending voice data packets over the internet rather than
communicating through standard telephone systems.
The idea is to use the internet as a telephone network
with some additional capabilities. Instead of
communicating over a circuit switched network, this
application allows communication between two parties
over the packet switched internet and it was in 1995,
when the first internet softphone appeared.
978-0-7695-3504-3/08 $25.00 © 2008 IEEE
DOI 10.1109/ICCEE.2008.176
Ale Imran
Department of Electronics Engineering,
Aligarh Muslim University, Aligarh, India
aleimran@zhcet.ac.in
Fig1:Diagrammatic View of VoIP
This concept allowed PC users to avoid long
distance charges, hence great for international calling
and a promising solution to the corporate world. In fact
one of the main areas of its applications are the call-
centers, where the customer is able to communicate
with an operator via the internet. Recently there have
been proposals in various countries to allow internet
telephony, which is indeed a welcome step. This means
that people can use VoIP services to connect to the
traditional telephone. In this case call travels across the
IP network, but terminates at the local phone network,
whose owner than charges a fee that is typically the
price of a local call [3]. Unrestricted VoIP, which is the
basic building block of Asterisk based voice exchange
makes it possible for people to use their phones either
fixed or mobile to call others in the country and
internationally for a fraction of the prices they pay
now. all that one needs to make a PC to PC call is a
VoIP based software .
Besides this, Engg economics also plays an
important role in popularizing this new concept. A
normal telephonic call requires 64 kbps, whereas the
same can be handled an as low as 6-8kbps using
codecs such as GSM. Hence the bandwidth
requirements will be considerably reduced and one can
stuff up, more calls in the same given bandwidth,
652
providing a clear cut advantage. In thhe future people
will stop paying for voice calls just like they don’t pay
for each e-mail they sent or each sitee they view. Of
course a service provider would chharge a fee for
providing internet services, but everrything you did
beyond that point wouldn’t carry any eextra cost .But in
a future where everyone will always bbe connected the
convergence several commentators haave been talking
about will finally become a reality.
2. Asterisk as a Voice exchangee
Private Branch Exchange is sim mply a private
telephone network which is used withinn a organization.
PBX systems are large installations able to handle
many extensions. PBX systemss are highly
programmable and customizable, able to meet the
needs and demands of different businessses[1].
Fig 2: A conventional PB
BX
Asterisk is a complete PBX in softwware written in C
programming language and it runs on Linux operating
systems. Asterisk does voice overr IP in many
protocols, and can interoperate w with almost all
standards-based telephony equipment using relatively
inexpensive hardware like for ex PCI cards[4].
Asterisk in fact creates a PBX that rivals the
functionalities of traditional telephonee based systems.
Asterisk based voice exchange is cosst effective, low
maintenance and flexible enough to hhandle all voice
and data networking .Asterisk does PBX switching,
CODEC translation and various other aapplications like
voicemail, conference bridging, IVR R and various
others.
Fig 3: Overview of Asterisk ssystem
2.1. Asterisk’s Architecture
Asterisk is carefully designed for maximum
flexibility. Specific APIs are defined aaround a central
653
PBX core system. This advanceed core handles the
internal interconnection of the PBX
X, cleanly abstracted
from the specific protocols, cod decs, and hardware
interfaces from the telephony appliications. This allows
Asterisk to use any suitable hardwware and technology
available now or in the future to perform
p its essential
functions, connecting hardware andd applications.
The Asterisk core handles these items internally [5]
PBX Switching The essence of Asterisk,
A of course, is
a Private Branch Exchange Switching system,
connecting calls together between n various users and
automated tasks. The Switching Core transparently
connects callers arriving on varrious hardware and
software interfaces.
Application Launcher launchess applications which
perform services for uses, such as voicemail, file
playback, and directory listing.
Codec Translator uses codecc modules for the
encoding and decoding of variouss audio compression
formats used in the telephony ind dustry. A number of
codecs are available to suit diversee needs and arrive at
the best balance between audio qu uality and bandwidth
usage.
Scheduler and I/O Manager haandles low-level task
scheduling and system manageement for optimal
performance under all load conditio ons
Fig 4: Asterisk’s Arch
hitecture
2.2. Loadable Module API’S
Four APIs are defined for loadable modules,
facilitating hardware and protocol abstraction. Using
this loadable module system, the Asterisk core does
not have to worry about details of how a caller is
connecting, what codecs are in use etc.
Channel API The channel API handles the type of
connection a caller is arriving on, be it a VoIP
connection, ISDN, PRI, Robbed biit signaling, or some
other technology. Dynamic mod dules are loaded to
handle the lower layer details of theese connections.
Application API The applicatiion API allows for
various task modules to be run to perform various
functions. Conferencing, Paging, Directory Listing.
Voicemail, In-line data transmission, and any other
task which a PBX system might perform now or in the
future are handled by these separate modules.
Codec Translator API Loads codec modules to
support various audio encoding and decoding formats
such as GSM, Mu-Law, A-law, and even MP3.
File Format API Handles the reading and writing of
various file formats for the storage of data in the file
system.
Using these APIs Asterisk achieves a complete
abstraction between its core functions as a PBX server
system and the varied technologies existing (or in
development) in the telephony arena. The modular
form is what allows Asterisk to seamlessly integrate
both currently implemented telephony switching
hardware and the growing Packet Voice technologies
emerging today[2]. The ability to load codec modules
allows Asterisk to support both the extremely compact
codecs necessary for Packet Voice over slow
connections such as a telephone modem while still
providing high audio quality over less constricted
connections. The application API provides for flexible
use of application modules to perform any function
flexibly on demand, and allows for open development
of new applications to suit unique needs and situations.
In addition, loading all applications as modules allows
for a flexible system, allowing the administrator to
design the best suited path for callers on the PBX
system and modify call paths to suit the changing.
3. Codecs and Protocols
Asterisk is quite versatile and has been designed
such that it can use a wide range of codecs and
protocols. Codecs are the mean with the help of which
analog signals can be converted to digital and
transmitted via an IP network.
Table 1: Various Codecs Used by Asterisk [3]
VoIP generally uses two types of protocols 1).
Signaling Protocols-for setting up a conversation 2)
Media transfer protocols for actual transfer of data,
once the connection has been set.[6] To obtain good
voice quality across the network we need to minimize
the packet losses, jitter, and use echo cancellation
techniques .Generally used protocols for VoIP based
654
technologies are SIP and H.323[7]. SIP is an
application layer protocol, designed to be independent
of the underlying transport layers and is used for
establishing, managing and terminating a multimedia
session.. In fact SIP is a text based protocol just like
HTTP protocol and its lightweight approach towards
setting up a call is what makes it quite a popular choice
for VoIP based networks [2].
Fig 5: SIP call set up
IAX is the Inter Asterisk Protocol used by Asterisk
and is a alternative to SIP ,H.323..Its generally used to
enable VoIP connections between servers and between
servers and clients.IAX is a very robust and fully
featured protocol yet its quite simple.. It is agnostic to
codecs and number of streams, meaning that it can be
used as a transport for virtually any type of data.
[5]This capability will be useful as videophones
becoming common.IAX supports both trunking and
multiplexing of the channels on the same link. In fact
the basic goals for which IAX was designed were 1)
Reduction in Bandwidth used in media transmissions
2) To provide basic support for NAT transparency and
3) To be easy to use behind firewalls.
Fig 6: IAX could be easily used beyond firewalls
Both SIP and IAX generally show the same
efficiency for lower no of simultaneous calls; however
as number of calls generally goes on increasing, IAX
takes the lead in performance over SIP.
4. Implementing the Asterisk Based Voice
Exchange
We have installed Fedora 4 Linux to implement
Asterisk and begin realizing our voice exchange by
compiling the Asterisk and IAX has been used to
connect multiple clients with the server.
The following commands help us in compiling the
Asterisk.
cd/root/sam/asterisk
make
make install For setting up a client on IAX client on Asterisk we do
make samples the following:
make progdocs
;[imran]
;type=friend
;secret=2222
;auth=md5
;host=dynamic
;reinvite=no
;canreinvite=no
;qualify=1000
;dtmfmode=inband;callerid="imran"<2222>
Fig7: Snapshot after the complete installation ;disallow=all
;allow=gsm
;context=incoming
Once it has been successfully installed we can start
Asterisk on the server by running the following
commands

/root/sam/asterisk-vvvc

4.1. Asterisk’s Dial Plan

The dial plan is stored in a text file, the
configuration file extensions.conf. In this file actions
are connected to extensions. Each extension belongs to Fig 8: IAX.conf file
a context, either the default context or a specific
context you create, like incoming sip calls, long- Whereas for setting up the Extensions on the
distance outbound PSTN calls, local calls, inter-office Asterisk server we do the following:
calls, etc. Users connecting to asterisk all belong to a For editing the Extensions configuration file
specific context (specified in the channel configuration /etc/asterisk/extensions.conf
file), which is where asterisk looks for advice on how [inbound-from-iax];
to handle the calls placed by that user, checking the Our context for IAX- clients
access rights to expensive lines, with different rule sets exten => extension no, priority, application
for local users and contacts calling from an outside (argl,arg2,...)
line. exten => 1111,1,Dial(IAX2/${EXTEN})
• /etc/asterisk exten => 2222,1,Dial(IAX2/${EXTEN})
Contains all of asterisk configuration files and exten => 3333,1,Dial(IAX2/${EXTEN})
logic information. exten => 4444,1,Dial(IAX2/${EXTEN})
• /usr/lib/asterisk/modules exten => 5555,1,Dial(IAX2/${EXTEN})
Contains all of asterisk’s loadable modules, exten => 6666,1,Dial(IAX2/${EXTEN})
operating asterisk functionality.
Applications, channels and resources are located
in this directory.
• /var/lib/asterisk/sounds
Contains all of asterisk’s sound files for playback
and pre-loaded applications (eg: VoiceMail).
• /var/lib/asterisk/agi-bin
Contains all of asterisk’s AGI scripts and AGI
logic.
Fig 9: Extensions.conf file
4.2. Configuring the Files
We configure the IAX and the Extensions at the Once we are done with this, we need to concentrate
following: on the installation and registration of the soft phones
IAX: /etc/asterisk/iax.conf that we are going to use at the client end.
Extensions: /etc/asterisk/extensions.conf

655

Fig 11: IDEFISK softphones being registered on the
server
Asterisk PBX currently support 4 types of
extension: Zap (TDM), SIP (VoIP), IAX2 (VoIP) and a
custom extension. By utilizing the various options,
enclosed within each extension, it is possible to
manipulate each extension’s behavior. The custom
extension can be utilized to program special
functionality extensions, enabling the creation of
extensions binded services, such as MicroBilling,
Information gathering, etc. Once an extension had been
defined, several optional parameters can be modified,
in order to complement the default settings..Special
attention must be given to NAT traversal issues and
DTMF issues, when working with IAX based
extensions.
You use a trunk to carry a call (or any number of
calls) to a Voice Service Provider or a device that cares
about what number you send to it (eg, another
Asterisk™/FreePBX™ Machine). There are 5 types of
trunks supported:
• Zap Trunk – Zap trunks provide connectivity
to legacy TDM systems via Analog interfaces
(FXO/FXS) or Digital interfaces (E1/T1).
• IAX2 Trunk – IAX2 trunks provide
interconnecting between Asterisk™ servers,
utilizing the Inter-Asterisk Exchange
Protocol.
• SIP Trunk – SIP trunks provide
interconnecting between Asterisk™ and SIP
service providers, utilizing the Session
Initiation Protocol.
• ENUM Trunk – ENUM trunks utilize the
e164.org number lookup services, and as a
practice aren’t used in generic PBX
installations.
• Custom Trunk – Custom trunks are available
in order to configure any type of trunk which
is not covered by the previous trunks, eg.
H323, BRI ISDN, etc.
Here in our implementation, though we have not
656
yet done the connection of our server with the
conventional PSTN, though it could be done with the
help of PCI cards like for example Digium’s
TDM400P to validate the connectivity with the
existent circuit switched network
A S T E R IS K P B X
H o s t: H o m e
P h o n e L in e c o n n e c te d to F X O c a r d
PSTN
SER VER A
S IP C L IE N T S @ S c h o o l
S IP C L IE N T S @ H o m e
4001@ school
3001@ hom e
IN T E R N E T
4002@ school
3002@ hom e
2001
1001 1002
A S T E R IS K P B X
A n a l o g p h o n e s co n n e c te d t o F X S c a rd
H o s t: S c h o o l
2002
SER VER B
A n a lo g p h o n e s c o n n e c t e d
to F X S c a r d
Fig 12: Experimental set up of Asterisk based voice
exchange
5. 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 less prone to viruses,
worms and hackers. As far as future work is concerned,
we would like to connect our Asterisk PBX with the
conventional circuit switched networks with the help of
PCI cards like for example Digium’s TDM400P [7] .
6. References
[1] Andre du Toit, “Private PBX Networks: Cost Effective
Communication Solutions” in Proc. IEEE 3rd AFRICON Conference
1992.
[2] Guo Fang Mao, Alex Talevski, Elizabeth Chang, ”Voice over
Internet Protocol on mobile devices” in Proc. 6th IEEE ICIS 2007.
[3] Md. Zaidul Alam, Saugata Bose, Md. Mhafuzur Rahman,
Mohammad Abdullah Al-Mumin, “Small office PBX using Voice
over IP” in Proc. IEEE ICACT 2007, Feb 12-14 2007
[4] Ryosuke Yamamoto, Fumikazu Iseki, Moo Wan Kim,
“Validation of VoIP System for University Network” in Proc. IEEE
ICACT 2008, Feb 17-20 2008.
[5] Asterisk.org, "Features and Architecture of Asterisk PBX",
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 Proc. IEEE CCECE/CCGEI, Saskatoon, May 2005
[7] Anand Gorti ”A fault tolerant VoIP implementation based on
open standards”, in Proc. IEEE 6th EDCC’06, 2006.