IJSRD - International Journal for Scientific Research & Development| Vol.

4, Issue 02, 2016 | ISSN (online): 2321-0613

A study of Design and Implementation of IVR System using Asterisk
Samir Borkar1 Sofia Pillai2
1
GHRCE Nagpur
Abstract— now days it is very important for any
organizations to provide interactive and cooperative
customer care service using telephony with client
interaction. The interactive voice response (IVR) system
serves as a communication link between computer and
people with connection of the telephone network with
acceptable instructions. The telephone user can get access to
the information wherever wanted at any random time with
the convenience of dialing a specific registered number and
following an automated instruction when a connection
established. The Interactive voice response system makes
use of computer generated voice responses or pre-recorded
message to provide information for responding to the choice
given by a telephone caller. This choice may be the input
given by Dual Tone Multi-Frequency (DTMF) signal, which
gets generated from the telephone set as user or caller
presses a choice numbered key, and the sequence of
messages to be played according to an internal menu
structure and the user input. According to input given and
wait time voicemail should be generated. In the proposed
model, first the VoIP system has been implemented with
IVR on asterisk cloud server which works as a VoIP service
provider. The asterisk server is configured on Centos
(Linux). The noticeable significance of the proposed system
is Asterisk server itself configured as a VoIP server, IVR
provider, VPN server, in a single asterisk server internally
on cloud environment. There is one central server and many
telephony endpoints registering with central server. The
server's task is to resolve a dialed address to an IP address
and establish a call connection for IVR.
Key words: Asterisk server, IVR, Internet Protocol (VoIP),
voice over SIP
I. INTRODUCTION
Interactive Voice Response (IVR) system which is a
complete IP telephony system, IP data networks (VoIP) uses
voice calling provides the customer support and very
efficient compared to the conventional system. The
proposed system can create new users and assign numbers to
each customer. Using that number, the call is sent to the
appropriate destination. IVR system provides features like
call conferencing; call waiting, call recording, voicemail for
each user. The proposed system is free from human
intervention. The proposed system is an IP PBX server
within the network and number of users. Interactive Voice
Response System (IVRS) is a customer-facing business
processes to automate the process calls represent a powerful
way, play the voice message; Redirect calls to service agents
and provide callers with real-time data from the database.
With a properly configured IVR system when your company
has been closed for the evening or weekend, can provide
top-notch customer service. IVR systems easily available in
several languages and multilingual customer service to
international markets abroad at home. Asterisk includes a
wealth of functions that make it a powerful IVR platform.
The Asterisk software is free, and there is no need to pay for
licensing of per port or per concurrent call. Since Asterisk

uses low cost PSTN interface hardware, deploying an
Asterisk system is significantly less expensive. Another
Asterisk benefit is the open nature of the platform. In
proprietary systems, the base functionality can only change
by vendor. In Asterisk, the source code is available and can
be change according to the specific requirements. To
develop an IVR system using Asterisk requires the skills
like, Linux, script development and telephony. Asterisk is
the open nature of the gain stage. The IVR system (playing
audio) enthusiasm and the caller are able to collect points.
Touchtone interface with voice interface software requires
additional space and increases the value of the platform.
Most IVR systems accept prerecorded audio, and there are
professional voice talents that offer custom prompts. The
VoIP technology is popular because it is open source to
everyone on internet. It has established itself as the best
option to the Public Service Telephone Network (PSTN)
interface. The hard-core telephones can be replaced the IP
phones those can be installed on any platform. This
system can be connected to the PSTN via VoIP gateways.
The protocol for the implementation of this technology is
SIP. The asterisk server which acts as VoIP server is the
open source software design for establishing the VoIP
connection between the users. This server is only the session
initiation protocol server between the IP address of the
users. Asterisk can be programmed for IVRS commonly
used in customer care center or any institution for making
call from PSTN to PBX. The proposed work implemented
on cloud server. On this cloud server asterisk server will be
configured and the IVR system will be implemented on that
asterisk server. In this system the IVRS is realized by
configuring the files namely extensions.conf, sip.conf, and
voicemail.conf encompassed inside the asterisk 1.6.1.2
package on the Centos 5 Linux platform. The user can
connect and communicate through this server anywhere in
the world.
II. PREVIOUS WORK
In our previous work [10] of designing IVR on Asterisk
cloud telephony system we have discussed about the process
of designing and realization of the system. Now this system
has been configured completely and user can communicate
through this system. Customer calls in your company. Call
is answered, not by conventional receptionist but by an
electronic receptionist. It prompt user to dial the extension
number of the department or person you want to reach by a
prerecorded voice. And it keeps on giving you choices. This
receptionist directs you towards getting your query resolved.
This Electronic Receptionist is Interactive Voice Response
System. An IVR system has been developed to provide
basic customer service and solutions to communications
problems with scheduling and managing inspections,
releasing resources for more complex tasks. While studying
this system we have gone though [1] the proposed system
provides the VoIP facility on Linux Centos 5 platform. We
can understand various security issues of the VoIP
communication using asterisk server. The VoIP server

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A study of Design and Implementation of IVR System using Asterisk
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provides many facilities by configuring it once and it is
capable to operate many users which are previously
registered in the SIP proxy server. The Centos 5 platform is
open source operating system. The [2] system design is
good and gives proper response to the caller who can use it
but it require high maintenance cost. In [3] a survey of
Voice over IP security, they discussed about the existing
capabilities and to point out failure in addressing the many
possible events of attacks on VoIP systems. In [4] they
discussed about the challenges while designing the
Automatic Speech Recognition System on telephony. It also
describes the challenges in designing interactive voice
response system. In [8] [9], they discussed about SIP proxy
server performance while using VoIP services. They have
also found out most of the features of SIP.
III. CONFIGURING SIP PROXY SERVER
SIP was defined by Internet Engineering Task Force (IETF)
and is a simpler than H.323 and tailored more specifically
for session establishment and termination. SIP is a text
based application layer protocol that can operate over any
transport layer protocol. It identifies the participating
entities with SIP addresses. These addresses are similar to
mail addresses. SIP identifies SIP servers based upon the
domain portion of an address. SIP clients can register with
one or more SIP domains. SIP uses the Session Description
Protocol (SDP) in the SIP payload to exchange information
about media encoding formats and end point addresses
(including IP address, protocol, and port number). Along
with the changes in SIP, the Session Initiation Protocol used
in VoIP phones to establish connections, distribute and
allocate calls. . The modification can involve changing
addresses or ports, inviting more participants, and adding or
deleting media streams. The Session Initiation Protocol
(SIP) is control protocol for creating, maintaining, and
distributing sessions for various types of media. The SIP
clients use RTP for data transfer and RTCP for managing
data streams. In data network, session Initiation Protocol
(SIP) is the standard communications protocol for voice and
video. Many companies make use of VoIP within their PBX
on the LAN for making connection with IP phones. SIP
deals with many protocols and play its role in the signaling
portion of a communication session. SIP clients typically
use TCP or UDP on port numbers 5060 or 5061 to connect
to SIP servers and its endpoints. Port 5060 is for nonencrypted signaling traffic and port 5061 is used for traffic
encrypted with Transport Layer Security. SIP is a generally
used tool for establishing sessions, teardown sessions and
modifying sessions which can work standalone as transport
protocols with many participants and it does not need to
confirm which type of session is going to establish. Session
initiation protocol is the major signaling protocols used in
Voice over IP (VoIP). SIP (Session Initiation Protocol) is
used for configuration of the SIP proxy server. The SIP
server can be configured by configuring the sip.conf file.
This file has the details of the SIP user registration for Inter
Asterisk Communication. The SIP.config file located at
/etc/asterisk/sip.conf. In the registration process there are
many specifications of users like host, type, DTMF mode,
insecure, secret, username, context, qualify, mailbox etc.
type-friend , we are using the friend type to make things
easy. Host-dynamic, our phones will register to Asterisk.

Else we can give the IP address of the phone. Context-frominternal, When Asterisk receives a call from this phone; it'll
look for the dialed extension number inside this context
within dialplan. Allow-ulaw; only allow the ulaw codec to
be used. Secret-password, this is the authentication
password the phone needs to use when authenticating
against Asterisk. There are many other specifications we can
define for user as per need. For the audio streaming Sip file
is generally used. This file contains the registration details
of every user who is using VoIP service. If this registration
is not done then user can not avail the facility of inter user
call between parties which are registered in the sip.conf file.
There are surly multiple users in a LAN using a VoIP
service using asterisk server. Every user detail must be
entered inside the same context. The sip server is using the
IP address of the Linux system which will work for asterisk
server. For testing on local machine 192.168.0.20 IP address
has been set which is the private range address. To login on
cloud server the IP address is 67.225.254.62. Asterisk
service listens on the default port 5060.
IV. PREPARING DIAL PLAN
Asterisk is available with interface for application
programming (APIs) in its own proprietary language we can
call such API as Dialplan. The Dial Plan is very crucial part
which is written inside the extensions.conf file. The dial
plans gives interactions to the asterisk server what and how
to perform task. The dialplan is truly the heart of any
Asterisk system, as it defines how Asterisk going to handle
inbound and outbound calls. It is programmed with set of
instructions that Asterisk will follow. Dialplan provides a
basic ‘if-else’ conditional and ‘goto’ constructs for
implementing applications. Dialplan does not provide any
other programming constructs, i.e., it does not provide for
iteration, switching, and procedural or object oriented
modules. Dialplan applications can be used to develop
telecommunication utilities such as IVR, queues, and
voice/video mail. Asterisk’s dialplan is fully customizable.
The [11] extensions.conf file resides at the
/etc/asterisk/extensions.conf. When asterisk server is started
for implementation the asterisk server read the
extensions.conf file. We have to write the dialplan in the
specific context. Dialplans are divided into sections called
contexts. Contexts can be name of departments or group,
which works for various different purposes. Contexts
provide facility to various parts of the dialplan from
interacting with each other. Every context has its own
different dial plans. The asterisk server initially looks inside
the context where it reads the dial plans and identifies which
task to do. Then, the asterisk server checks the sip.conf file
and verifies the context written in the extensions.conf file.
After checking the relationship of resemblance of both the
contexts the server will execute specified task without any
error. An extension that is defined in one context is
separated from extensions in any other context, unless
interaction is specifically permitted. If it finds the illegal
context inside the sip.conf file, it will return an error and
terminate the process which was running. The dial plans
fields containing following statements.
[context-name]
exten => _5005,1,Operation();
exten => i,1,Playback(invalid);

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A study of Design and Implementation of IVR System using Asterisk
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In these statements [context-name] is the name of
the context written in the sip.conf. exten is the keyword used
to write the dial plans inside the extensions.conf file. Here
the word extension is not use for terminal equipment or
phone but it is the keyword to be used for dialplan. As the
file is extensions.conf so every dial plan must be written by
using exten = > _5005 is the extension number of the user
to whom the other user dials to call. It can be single digit for
routing the calls. i represents the invalid menu choice. The
number after the extension number is priority of the task
here it is 1. Using this priority number the task must be
performed. If the priority is set as 3 then the priority of task
is given as lower than the priority which is set as 1. The
greater numbers represent the lowest priority task. Operation
() is the function which is the task given to the asterisk
server to be done. The Operation () can be any functions of
the asterisk like Answer (), Hangup (), Playback (), Goto (),
Dial (), Read (), Voicemail () etc.
V. WORKING OF IVR SYSTEM
Setting up an IVR on Asterisk is not complex task, but we
need to have knowledge about SIP, SIP proxy, Asterisk
capability and procedure to be followed. As the proposed
system implemented on cloud server it mainly based on the
concept of cloud telephony. Cloud telephony refers
particularly to voice services and telephone equipment
replacement, such as a Private branch exchange with thirdparty VoIP service. When registered customer call on
number of the organization welcome message can be listen.
Interactive voice response activated and greeting message
played like welcome to Asterisk, Press 1 for Sales, Press 2
for support Press 3 for administrative help. As per the
customer press the number from his mobile the call is routed
to client phone of department.

Fig. 2: Levels of working according to dial plan
In the above figure the multilevel IVR is
shown.There are three levels in given IVR system, in which
when caller makes call to organization, an interactive voice
response introduces Menu1 in which 3 choices are there,
press1 for SALES, press2 for SUPPORT, press3 for
ADMIN after customer gives runtime input i.e. choice 1
then another Menu2 is activated like press1 for extension
5001, press2 for 5002, press3 for 5003 similarly Menu3 is
given. According to the choices of caller, the call will be
automatically divert to appropriate department and person.
VI. RESULT AND DISCUSSION
The interactive voice response system is implemented on
Linux Centos 5 platform on the cloud server. This system
runs on the asterisk package which is easily available and
configurable on a Linux platform. These packages are
installed on cloud asterisk server. For demonstration we
need to login to cloud sever through putty with server IP
address and password. User need to be registered on server
for making calls with specified number and authentication.
This system will provide a low cost VOIP and IVR solution
to the institutions having large number of employees. In the
proposed system the phone which are being used for inter
user communication are IP soft phones (x-lite). Depending
on the input and requested information, the IVR will visit
the resource on the server and related information system
then search the required information and queries result to the
IVR server and then play the voice files to the customers
through telephones. After that it will finish the process of
IVR request and response. We can make use of android
applications like astTECS, Zoiper as soft phone for calling
IVR. We need to register users on these applications for
calling IVR system.

Fig. 1: Simple flow of IVR process

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A study of Design and Implementation of IVR System using Asterisk
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Fig. 3: Showing SIP users
1

Fig. 4: Image showing login through putty and call
connected to user by soft phone

2

3
4
Fig 6: showing 1.user registration status, 2 registration
details of user, 3.calling screen and 4. After receiving call
from user 5001
VII. CONCLUSION

Fig. 5: Showing SIP user registered with their IP addresses

Now days, the existing IVR services need bulky hardware
and high cost maintenance to run IVR system effectively.
Traditional IVR system requires human resources to attend
and give information but by using IVR on asterisk cloud
server we can introduce flexibility and cost effectiveness.
The Asterisk software is free. The deployment of an
Asterisk system is less expensive. It provides organizations
to work comfortably through phone or computer. It will
soon replace the existing wired telephone system. The
system designed will intelligently interact and will provide
good response to the caller who can access it. This type of
system performs operations effectively and reliably without
human interventions. The asterisk system interprets only
digits. The inputs in the form of only digits can overrule this
problem. Asterisk includes many functions like call
recording, voicemail, intelligent routing, load balancing,
reports and graphs that make it a robust and effective IVR
platform. IVR applications can be developed through the
Asterisk Gateway Interface with correct dialplan and can
integrate virtually with any supporting system.

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[11] www.digium.com/asterisk_handbook/sip.conf.pdf
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[14] http://www.asterisk.org
[15] www.asteriskdocs.org

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