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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

DESIGN AND DEVELOPMENT OF ASTERISK BASED
COMPUTER SERVICES ACCESSING FRAMEWORK FOR
THE VISUALLY IMPAIRED
Vaskar Deka1, Neelakshi Sarma2 and Mayashree Gogoi 3
Department of Information Technology, Gauhati University

ABSTRACT
We are today living in age where technology continues to bring about one magical transformation after
another into our lives. Especially computers and internet that constitute the cardinal components of the
present day technology have so penetrated into our live that they today make up an indispensable part of
life. In this paper, we present an Asterisk based framework that is designed particularly to the benefit of the
visually challenged people. These people find it difficult to use the conventional computer access devices
like the keyboard, the mouse or the monitor as these devices require a good hand-eye co-ordination. Our
framework is designed to take DTMF input from the user that serve as commands for execution of different
operations on the computer and provide output in the form of voice using speech synthesis. Additionally,
our framework provides the benefit of customizing it as per user needs as it is based on the open source
platform Asterisk.

KEYWORDS
Asterisk, DTMF, IVR, text2wave, dialplan, context, phpagi

1. INTRODUCTION
Living in this era of constantly evolving technology, it is very difficult for us to imagine of lives
when the elements of the present day technology, existed, if at all, only in the imaginations of the
demiurgic minds. We are today living in an era of information technology with computers
constituting an inevitable force of this era. From education to entertainment, computer has a role
everywhere to play. Whether its reading the encyclopedia or reading newspapers from across the
world, doing some shopping or checking movie times, paying the bills or accessing bank accounts
or turning the lights and the AC on, the computer is always there at our service. However, what
the general user takes for granted is infact an issue for the physically challenged people. Visually
challenged people for example find it troublesome to use conventional computer access devices
like the mouse or keyboard. Limitations as these tend to produce a gap between these people and
the technological marvels of the day. Hence, there is an increased need to address the computing
requirements of those who have physical limitations. This, infact is an important aspect of
information technology.
To address this issue, several efforts have already been made and successfully implemented.
Users with limited hand mobility who want to use a keyboard have several options. Keyboards
with larger keys are available.[1] Hearing-impaired users can instruct programs to display words
instead of sounds. On the contrary, blind users can work with voice output i.e, the computer reads
DOI : 10.5121/ijcsity.2013.1402

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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

the information that appears on the screen. In our approach, we focus on making use of the
Asterisk platform to send DTMF input to the computer that would serve as commands for
execution of different programs on the computer and relay voice the output in the form of
narrated voice to the user.

2. ASTERISK AND ITS OVERVIEW
2.1. What is Asterisk?
Asterisk is a computerized telephony integration platform.[1] It is a software that can turn an
ordinary computer into a voice communications server.[2]. Asterisk is today one of the most
popular open source telephony software and provides most features of a traditional PBXs such as
voicemail, call conferencing, ACD(Automatic Call Distribution) and IVR(Interactive Voice
Response) as well as it can handle call queuing, bridging and conference calling with the use of
virtual conference rooms. [3] It can serve both as a tool-kit for telephony applications as well as a
full featured call-processing server in itself.[4]Asterisk can interoperate with almost all standards
based telephony equipment using relatively inexpensive hardware.[4]Digium and a number of
firms sell PCI cards to attach telephones, telephone lines, T1 and E1 lines and other analog and
digital phone services to a Asterisk server.[5]. By supporting a mix of traditional and VoIP
telephony services, Asterisk allows developers to build new telephone systems or gradually
migrate existing systems to new technologies.[5]. Though originally designed for Linux, Asterisk
also runs on a wide variety of operating systems including NetBSD, OpenBSD, FreeBSD, Mac
OS X and Solaris.[6]

2.2. Asterisk architecture
Asterisk acts as a middleware connecting telephony technologies on the bottom to telephony
applications on the top, creating a consistent environment for deploying a mixed telephony
environment.[7]Telephony technologies can include VoIP services like SIP, H.323, IAX, and
MGCP(both gateways and phones) as well as more traditional TDM technologies like T1, ISDN
PRI, analog POTS and PSTN services , Basic Rate ISDN(BRI) and more.[7] Telephony
applications include things such as call bridging, conferencing, voicemail, auto-attendant, custom
IVR scripting, call parking, intercom and more.[7]

Figure 1. Asterisk architecture
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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

2.3. Asterisk Applications
The modular architecture of Asterisk makes it highly extensible and flexible thus facilitating its
application in a wide variety of uses some of which are briefly stated below:
 Asterisk as a PBX : Asterisk can be configured as the core of an IP or hybrid PBX, switching
calls, managing routes enabling features and connecting callers with the outside world over IP
analog(POTS) and digital(T1/E1) connection.[8]
 Asterisk as a Gateway: Asterisk in connection with telephony interface cards can be used to
connect legacy PBX systems with VoIP services or conversely to link IP-PBX systems to legacy
PBX systems.[9]
 Asterisk in the call center: With support for call queues, IVRs, outbound dialing, recording,
live monitoring and reporting, Asterisk includes virtually everything needed to create a working
call center.[10]
 Asterisk as a conference bridge: Asterisk includes a standard application called confBridge
which is a high definition capable conference bridge component that makes it easy to build
standalone conferencing services or to integrate conferencing into other solutions including IPPBX systems.[11]
 Asterisk as a voicemail server: Asterisk‟s standard voicemail components make it trivial to
assemble a world class voicemail server.[12]
 Asterisk as an IVR: Asterisk acts strongly as an IVR platform because of it‟s wide support to
various functions such as audio playback, and recording, digit collection, database and webservice access, calendar integration and optional speech recognition and synthesis.[13]

3. RELATED WORKS
 Lwaji: The Lwaji project was commissioned by the Department of Arts and Culture, South
Africa. The project aims to develop a telephone-based speech-driven information system. The
project aimed at providing South Africans with an opportunity to access government information
and services in any of South Africa‟s eleven official languages using either landline telephones or
mobile telephones, free of charge. Since not all South Africans have access to the internet or to
printed media, telephones therefore serve as an important alternative for accessing government
information. [14]The underlying telephony platform used is Asterisk with focus on integration of
Test-To-Speech and Automatic Speech Recognition technologies with Asterisk[15]
 Zahir Koradia and Aaditeshwar Seth [16] deployed PhonePeti, an answering machine system
at a Community Radio station in north India called Gurgaon Ki Aawaz(GKA). PhonePeti is
implemented as an application over Asterisk. A GSM gateway is connected to the computer
running Asterisk to terminate GSM calls and redirect them to Asterisk for further processing.
Thus when a listener calls the mobile number associated with PhonePeti, the call lands on
Asterisk via the GSM gateway. At this point, Asterisk begins execution of the telephony
application which prompts the user to leave a message after a beep. The application records the
message from the caller and pushes it to a web server to make it available to the station staff over
the internet.
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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

 Miroslav Voznak, Lukas Kapicak, Jaroslav Zdralek, Pavel Nevlud and Jan Rozhan[17]
implemented a SW PBX Asterisk with capabilities for recording and playback of video files.
Native Asterisk supports audio and video calls but do not allow recording and playback of video
files. In this system, the video files were directly saved into a directory in the server and a list
generated and read out to the user by means of speech synthesis. DTMF key input from the user
was used to select the desired file.
 Raphael Frank, Benjamin Hourte, Dan Jungels and Thomas Engel[18] developed a
communication tool to enhance the co-ordination and response time during emergency situations.
The application is supposed to enable group calls among different rescue groups in such
situations. Additionally they also presented a video application that provides real time access to
surveillance cameras. The group call system was based on a centralized client-server model to
allow concentrating all relevant information in one central point to obtain a better view of the
crisis situation. The central entity used in the system was a Voice-over-IP server based on
Asterisk. The server was equipped with a additional PRI cards1 to connect to PSTN/ISDN and
GSM networks so as to interconnect the VoIP communications with traditional circuit switched
communications.

 Iasonas Polakis, Georgios Kontaxis and Sotiris Ioannidis[19] implemented a system to protect
landlines from DIAL attacks.The core component of the platform is asterisk PBX. They crafted a
CAPTCHA called Phone CAPTCHA. When an incoming call is received, Asterisk places the call
in the call queue. The caller then receives a Phone CAPTCHA and has a limited amount of time
to answer the CAPTCHA test using the phone dialpad. The Phone CAPTCHA test requires the
caller to press a sequence of keys based on the instructions presented to him by recorded
message. If the caller provides the correct answer, Asterisk forwards the call to the destination as
determined by the dialplan. Otherwise the call is dropped. This mechanism provides automated
calls from binding to the end device and consuming resources which could prevent legitimate
callers from reaching the destination number.
 Munir B.Sayyad, Nilesh Nikam, S.L Nalbalwar[20] examined the security issues in IP-PBX
and suggested mechanisms to resolve them. Security issues that arises in an IP-PBX are
dummy/blank/fake CLI(Caller Line Identification) configuration at IP-PBX or dummy IP
addresses or IP phones connection to IP-PBX. For blocking of calls with dummy CLI or IP
address, they suggested to maintain a mapping table of the IP addresses with assigned subscriber
numbers(CLI ) at MSC end. If a call comes from IP-PBX with IP address and CLI that is not
matching with the entries in the mapping table of MSC, such calls would be rejected. This is
authentication of both IP address and assigned CLI number. Only authenticated calls will be
allowed.
 Ndaohialy Manda-Vy Ravonimanantosa, Andry Auguste Randriamitantosa and Malalatiana
Hajasoa [21] defined a relationship between the number of calls made in an Asterisk server and
use of the processor in this server as the processor is one of the major hardware devices. They
calculated the correlation and found that there was a close relation between the number of calls
made and percentage of CPU usage. The relationship led to the inference that if the number of
calls continues to increase, it could lead to a limit so that this limit represents the physical
capacity the Asterisk server.

 Kranti Kumar Appari [22] developed a customized IVR implementation using VoiceXML on
VoIP platform. VoiceXML delivers its output in form of audio which will be very much useful to
all kinds of people. When the user places a call to the designated VoiceXML extension, the
particular code for call flow gets activated. After the session is established successfully, Asterisk
Gateway Interface call the initial VoiceXML browser and calls the initial VoiceXMl script hosted
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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

on the same server. VoiceGlue starts interpreting the VoiceXML code for audio output and the
user input by way of DTMF signals or voice commands. Once the user is finished with his
operation on VoiceXML the control is handed over to Asterisk(IPPBX) .

 Miroslav Voznak and Jan Rozhan [23] developed a speech quality monitoring tool in
accordance with PESQ (Perceptual Evaluation of Speech Quality ) that automatically runs and
calculates the MOS(Mean Opinion Score). As the heart of the system, they configured the
Asterisk PBX to make a call regularly every 5 minutes to the SIP/GSM gateway, which is
connected to Asterisk PBX with the SIP trunk over the LAN. The call is handled by the gateway
and sent to the GSM network over a GSM interface equipped with a SIM card. The call is
destined to another GSM interface of the SIP/GSM gateway from where it is routed back to the
Asterisk PBX. Thos way they created a loop which starts and ends on the Asterisk PBX. This
approach was mainly beneficial because of the huge configuration capabilities of the Asterisk
PBX.

 Sadaf Nawaj, Saman Cheema, Sana Akram, K. Shehzad and S. Muzahir Abbas [24]
introduced a new procedure for Vertical Handoff(VHO). It is a mechanism in which multiple
users having access to different networks can execute the VHO in such a manner that when
handoff takes place, it do not interrupt any ongoing call or data session. A server and clients were
created. By employing a technique called IP Assignment VHO was carried out successfully.
Asterisk was chosen as the server taking into consideration the facts that Asterisk system controls
all routing information as well as it supports SIP which is the most important component to
establish, maintain and terminate a connection. Also Asterisk supports different soft-phones that
were used as clients in the system.
 Lerato Lerato, Molise Mokhomo, Thato Pebane and Lerato Mohapi[25] introduces a
VoiceXML gateway that comprises the integration of OpenVXI and an Asterisk telephony
platform; focusing on the addition of the HTK-based speech recognition capability of the system.
Finally the paper reports the successful implementation of a recognition interface that conforms
to the SRGS specification. SRGS is a standard that defines the syntax representing grammars to
be used in speech recognition.

4. PROPOSED SYSTEM
Our system will maintain a client-server architecture wherein a PC installed with Asterisk will act
as a server. A Bluetooth enabled mobile phone will remain connected to the server via Bluetooth.
Any phone that makes a call to this particular mobile phone is a client that accesses the services
offered by the server. The server will have a preprocessed rule set on what task to accomplish
when a certain command is issued by the client. The commands along with their corresponding
usage are made available to the client on demand. The client would accordingly issue a
command to be executed on the server for the desired action. Prior to execution, the server will
check the command for appropriateness. A command appropriately issued would execute the
corresponding program on the server while any inappropriate command or any unsuccessful
execution of a program would send a negative feedback to the user.

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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

Figure 2. Block diagram of proposed system

The core of the system is the Bluetooth connection between the PC and the mobile phone. Once
the Bluetooth address of the mobile phone is specified in the chan_mobile.conf file of Asterisk
which is the Bluetooth configuration file for Asterisk, any calls made to this mobile phone are
redirected to the Asterisk server. The server can then be made to play around with the call by
making appropriate configurations in the Asterisk dialplan which is specified in the file
extensions.conf. The dialplan is the heart of any Asterisk system that is built of a multitude of
configuration contexts, each configuration context containing the logic to be performed in
accordance o a dialed number or a dialed pattern. In our case, we invoke a phpagi script that
defines the sequence of actions to be performed once a call is received by the server.
Once a call is received, the phpagi script reads out a menu of available options to the user by
means of speech synthesis. This is accomplished using the text2wave program of the Festival
framework that takes as input plain text and converts it into audible speech output. The menu
provided to the user here is a list of operations to be performed on the server computer along with
the corresponding key to be pressed by the user in the dialpad. The user accordingly feeds in the
DTMF key input. This enables DTMF based control of a computer system

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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

5. IMPLEMENTATION DETAILS
5.1. Installing Asterisk and configuring it for Bluetooth
Asterisk 1.8 is installed on a Ubuntu10.04 machine and appropriate configurations made on the
chan_mobile.conf and extensions.conf file in the /etc/asterisk directory. As already mentioned
earlier, the chan_mobile.conf file is the blue-tooth configuration file for Asterisk, therefore in
order for the Asterisk server to be able to listen to other Bluetooth devices, we first of all need to
find out the Bluetooth adapter MAC address of the server and add it to the chan_mobile.conf file.
Once this is done, the next step is to pair the Bluetooth enabled mobile phone to the server. This
is done by first scanning the phone using the mobile search command from the asterisk console
that provides Bluetooth details of the phone. All that is to be done next is to provide these details
in the chan_mobile.conf file under a particular adapter and direct it to a particular context defined
in the extensions.conf file. Depending on the context we put it in, if a call comes in to the mobile
phone and the phone is within the proximity of the Asterisk server, it should route the call to the
server. The context contains the logic o be performed in accordance to a dialled number or
pattern. As have already been mentioned earlier, in this particular case, we invoke a phpagi script
within the context for the sequence of actions to be performed once a call is received.

5.2. Implementation of different services
5.2.1. Execution of different system commands on the server computer through DTMF
The first step towards the implementation of the framework involved the execution of different
system commands on the computer. Our framework being based on Linux, the implementation
involved the execution of different Linux system commands. Though there are a multitude of
commands to be executed on a Linux system, we have concentrated only on a handful of them;
primarily those that would enable a user to navigate through the file-system and perform some of
the very often required operations on it‟s contents. The list of commands along with the key to be
pressed in the dial-pad is read out to the user. The user accordingly chooses the desired command
by pressing the appropriate key in the dial-pad

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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

Figure 3. Execution of system commands part-I

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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

Figure 4. Execution of system commands part-II

After having successfully controlled the in-house resources of the computer, we next
concentrated on exercising the capabilities of the server for accessing the resources external to the
computer system and relaying the retrieved information to the user through narrated voice in the
same way as we have done for the in-house resources.

5.2.2. Voice based retrieval of weather information through DTMF tone
For the retrieval of weather information, we have used the WorldWeatherOnline free API. This
API provides the current weather condition and next 5 days of accurate and reliable weather
forecast. The API returns weather data in XML, JSON or CSV format and contains weather
elements like temperature, precipitation, weather description, weather icon and wind speed. We
have initially retrieved the weather forecast of four cities in XML format and have used the
XMLDOM parser to parse the XML response and retrieve the important weather elements. The
user is provided a menu listing the cities and the user is supposed to select one among these for
which he wishes to know the weather condition which the user does by a DTMF key input. As
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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

already mentioned, the response is received in XML format which is parsed and is then integrated
with our phpagi solution to convey the message to the user via voice over the telephone.

Figure 5. Retrieval of weather information

5.2.3. Voice based retrieval of current currency exchange rate through DTMF tone
We have used the „THE EASY API‟ free API for the current currency exchange rate and
conversion from a source currency format to the destination currency format. The user is
provided with a list of source and destination currency formats for the user to select the desired
one through DTMF. The user is then prompted to enter the amount for conversion which is done
again through DTMF. The API sends response in XML format which is parsed using the
XMLDOM parser and the output conveyed back to the user via voice using speech synthesis over
the telephone.

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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

Figure 6. Retrieval of current currency exchange rate

5.2.4. Voice based retrieval of e-news through DTMF tone
This module is aimed at retrieving an e-newspaper and conveying the contents back to the user
via voice over the phone using speech synthesis. We have chosen the „The Hindu‟ newspaper for
the purpose. The response is received in HTML format which is parsed using the SIMPLE HTML
DOM parser. The headlines are read out to the user with a query after each one of them regarding
whether or not the user wants to go through the news in detail to which if the user response in an
affirmative, the detailed news is read out; otherwise it is skipped and moved to the next headline .

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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

Figure 7. Retrieval of e-newspaper

5.2.5. Integrating the different services into a DTMF tone based Welcome Module
All the services discussed above are integrated into a single module resulting in a compact
computer control framework. When a connection is first set-up between the server and the user,
the Welcome Module presents a list of the services to the user. Each item in the list is again
divided into a menu and submenu allowing the user to effectively and efficiently control the
services on the computer.

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International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

Figure 8. Integration of different services

5.2.6. Introducing the security services
To prevent misuse of the system by unauthorized users, user authentication using username and
password is provided. When the user first connects to the system, the user is prompted to enter
the username and password. On successful verification, the user is authorized to access the server.
Authentication is particularly important because Asterisk always runs on administrator mode.
Therefore anyone who gains access to the server can execute any operation on the computer
system.

6. CONCLUSION AND FUTURE SCOPE
In this paper, we have demonstrated a framework that makes use of Asterisk to control a
computer system using DTMF and speech synthesis. The system can further be enhanced to
accept voice input from the user. This will also ensure better security of the system. It can still
further be augmented to offer services in local languages that would particularly be beneficial to
the common people.

REFERENCES
[1]
[2]
[3]

i6net* (2013) “What is Asterisk?” [Online] Available: http://www.i6net.com/what-is-asterisk/
Asterisk(2013) Asterisk Home Page [Online] Available: http://www.asterisk.org/get-started/
PCMAG.COM(2013) “Online Extra: Asterisk :An Open Source PBX Gateway” [Online] Available:
http://www.pcmag.com/article2/0,2817,1607848,00.asp
33
International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013

[4]
[5]
[6]
[7]
[8]
[9]
[10]
[11]
[12]
[13]
[14]
[15]
[16]
[17]

[18]

[19]
[20]

[21]

[22]

[23]
[24]

[25]

M-indya(2013) “What is Asterisk? Advantages and disadvantages of Asterisk” [Online] Available:
http://www.m-indya.com/shownews.php?newsid=3063
GNUAcademy(2013) “Asterisk(PBX)” [Online] Available:
http://www.gnuacademy.org/ippbx/asterisk-pbx
Wikipedia (2013) “Asterisk (PBX)” [Online] Available:
http://en.wikipedia.org/wiki/Asterisk_%28PBX%29
Mark Spencer, Mack Allison and Christopher Rhodes, “The Asterisk Handbook Version 2” [Online]
Available: http://bananabread.net/AsteriskHandbk_w_Bkmarks.pdf
TRTechServices(2013) “About Asterisk(VoIP PBX)” [Online] Available:
http://www.itrtechservices.com/index.php?option=com_content&view= article&id=107&Itemid=136
Slideshare(2013) “Asteriskl as a VoIP Gateway” [Online] Available:
http://www.slideshare.net/Catharine24/asterisk-as-voip-gateway
Asterisk(2013) “Call-center” [Online] Available: http://www.asterisk.org/getstarted/applications/call-center
Asterisk(2013)”Conference Bridge” [Online] Available: http://www.asterisk.org/getstarted/applications/conference
Asterisk(2013)”Voicemail Server” [Online] Available: http://www.asterisk.org/getstarted/applications/voicemail
Asterisk(2013)”IVR Server”[Online] Available: http://www.asterisk.org/get-started/applications/ivr
Lwaji(2013) “About LWAJI” [Online] Available: http://www.meraka.org.za/lwazi/about.php
Lwaji(2013) “Scientific and technical outputs development” [Online] Available:
http://www.meraka.org.za/lwazi/stop.php
Zahir Koradia and Aaditeshwar seth, “PhonePeti: Exploring the Role of an Answering Machine
system in a Community Radio Station in India”
Miroslav Voznak, Lukas Kapicak, Jaroslav Zdralrk, Pavel Nevlud, Jan Rozhon, “Video Files
Recording and Playback with VoiceXML,” Recent Researches in Communications, Automation,
Signal Processing, Nanotechnology, astronomy and Nuclear Physics
Raphael Frank, Benjamin Hourte, Dan Jungels and Thomas Engel, “Interoperable Networking
Applications for Emergency Services”, International Journal of Multimedia and Ubiquious
Engineering Vol $, No. 1,January, 2009
Iasonas Polakis, Georgios Kontaxis and Sotiris Ioannidis, ”CAPTCHuring Automated (Smart) Phone
Attacks”
Munir B. Dayyad, Nikesh Nikam, S.L Nalbatwar, “Novel Approach to Resolve Network Security
Issues in IP PBX in Converged Architecture”, International Journal of Engineering Science and
Technology Vol.2(10), 2010, 5765-5769
Ndaohialy Manda-Vy RAvonimanantosa, Andry Auguste Randriamitantsoa, Malalatiana Hajasoa
Ramafiarisona, “The Impact of Calls to the Processor on a Server PABX”, International Journal of
Future Computer and Communication, Vol. 1, No. 1, June 2012
Kranti Kumar Appari, “Customized IVR Implementation Using VoiceXML on SIP(VoIP)
Communication Platform ”, International Journal of Modern Engineering Research(IJMER) Vol.2,
Issue. 6, Nov-Dec. 2012 pp-4239-4243 ISSN: 2249-6645
Miroslav Voznak and Jan Rozhan , “Automated Speech Quality Monitoring Tool based on
Perceptual Evaluation”, Recent Researches in Communications, Information Science and Education
Sadaf Nawaj, Saman Cheema, Sana Akram, K. Shehzad , S. Muzahir Abbas, “Implementation of
Vertical Handoff in Heterogenous Networks”, International Journal of Future Generation
Communication and Networking Vol. 5, No. 3, September 2012
Lerato Lerato, Molise Mokhimo, Thato Pebane, Lerato Mohapi,” Integration of an HTK Speech
Recognizer with Open VXI and an Asterisk based VoiceXML Gateway”

34

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Design and development of asterisk based computer services accessing framework for the visually impaired

  • 1. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 DESIGN AND DEVELOPMENT OF ASTERISK BASED COMPUTER SERVICES ACCESSING FRAMEWORK FOR THE VISUALLY IMPAIRED Vaskar Deka1, Neelakshi Sarma2 and Mayashree Gogoi 3 Department of Information Technology, Gauhati University ABSTRACT We are today living in age where technology continues to bring about one magical transformation after another into our lives. Especially computers and internet that constitute the cardinal components of the present day technology have so penetrated into our live that they today make up an indispensable part of life. In this paper, we present an Asterisk based framework that is designed particularly to the benefit of the visually challenged people. These people find it difficult to use the conventional computer access devices like the keyboard, the mouse or the monitor as these devices require a good hand-eye co-ordination. Our framework is designed to take DTMF input from the user that serve as commands for execution of different operations on the computer and provide output in the form of voice using speech synthesis. Additionally, our framework provides the benefit of customizing it as per user needs as it is based on the open source platform Asterisk. KEYWORDS Asterisk, DTMF, IVR, text2wave, dialplan, context, phpagi 1. INTRODUCTION Living in this era of constantly evolving technology, it is very difficult for us to imagine of lives when the elements of the present day technology, existed, if at all, only in the imaginations of the demiurgic minds. We are today living in an era of information technology with computers constituting an inevitable force of this era. From education to entertainment, computer has a role everywhere to play. Whether its reading the encyclopedia or reading newspapers from across the world, doing some shopping or checking movie times, paying the bills or accessing bank accounts or turning the lights and the AC on, the computer is always there at our service. However, what the general user takes for granted is infact an issue for the physically challenged people. Visually challenged people for example find it troublesome to use conventional computer access devices like the mouse or keyboard. Limitations as these tend to produce a gap between these people and the technological marvels of the day. Hence, there is an increased need to address the computing requirements of those who have physical limitations. This, infact is an important aspect of information technology. To address this issue, several efforts have already been made and successfully implemented. Users with limited hand mobility who want to use a keyboard have several options. Keyboards with larger keys are available.[1] Hearing-impaired users can instruct programs to display words instead of sounds. On the contrary, blind users can work with voice output i.e, the computer reads DOI : 10.5121/ijcsity.2013.1402 21
  • 2. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 the information that appears on the screen. In our approach, we focus on making use of the Asterisk platform to send DTMF input to the computer that would serve as commands for execution of different programs on the computer and relay voice the output in the form of narrated voice to the user. 2. ASTERISK AND ITS OVERVIEW 2.1. What is Asterisk? Asterisk is a computerized telephony integration platform.[1] It is a software that can turn an ordinary computer into a voice communications server.[2]. Asterisk is today one of the most popular open source telephony software and provides most features of a traditional PBXs such as voicemail, call conferencing, ACD(Automatic Call Distribution) and IVR(Interactive Voice Response) as well as it can handle call queuing, bridging and conference calling with the use of virtual conference rooms. [3] It can serve both as a tool-kit for telephony applications as well as a full featured call-processing server in itself.[4]Asterisk can interoperate with almost all standards based telephony equipment using relatively inexpensive hardware.[4]Digium and a number of firms sell PCI cards to attach telephones, telephone lines, T1 and E1 lines and other analog and digital phone services to a Asterisk server.[5]. By supporting a mix of traditional and VoIP telephony services, Asterisk allows developers to build new telephone systems or gradually migrate existing systems to new technologies.[5]. Though originally designed for Linux, Asterisk also runs on a wide variety of operating systems including NetBSD, OpenBSD, FreeBSD, Mac OS X and Solaris.[6] 2.2. Asterisk architecture Asterisk acts as a middleware connecting telephony technologies on the bottom to telephony applications on the top, creating a consistent environment for deploying a mixed telephony environment.[7]Telephony technologies can include VoIP services like SIP, H.323, IAX, and MGCP(both gateways and phones) as well as more traditional TDM technologies like T1, ISDN PRI, analog POTS and PSTN services , Basic Rate ISDN(BRI) and more.[7] Telephony applications include things such as call bridging, conferencing, voicemail, auto-attendant, custom IVR scripting, call parking, intercom and more.[7] Figure 1. Asterisk architecture 22
  • 3. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 2.3. Asterisk Applications The modular architecture of Asterisk makes it highly extensible and flexible thus facilitating its application in a wide variety of uses some of which are briefly stated below:  Asterisk as a PBX : Asterisk can be configured as the core of an IP or hybrid PBX, switching calls, managing routes enabling features and connecting callers with the outside world over IP analog(POTS) and digital(T1/E1) connection.[8]  Asterisk as a Gateway: Asterisk in connection with telephony interface cards can be used to connect legacy PBX systems with VoIP services or conversely to link IP-PBX systems to legacy PBX systems.[9]  Asterisk in the call center: With support for call queues, IVRs, outbound dialing, recording, live monitoring and reporting, Asterisk includes virtually everything needed to create a working call center.[10]  Asterisk as a conference bridge: Asterisk includes a standard application called confBridge which is a high definition capable conference bridge component that makes it easy to build standalone conferencing services or to integrate conferencing into other solutions including IPPBX systems.[11]  Asterisk as a voicemail server: Asterisk‟s standard voicemail components make it trivial to assemble a world class voicemail server.[12]  Asterisk as an IVR: Asterisk acts strongly as an IVR platform because of it‟s wide support to various functions such as audio playback, and recording, digit collection, database and webservice access, calendar integration and optional speech recognition and synthesis.[13] 3. RELATED WORKS  Lwaji: The Lwaji project was commissioned by the Department of Arts and Culture, South Africa. The project aims to develop a telephone-based speech-driven information system. The project aimed at providing South Africans with an opportunity to access government information and services in any of South Africa‟s eleven official languages using either landline telephones or mobile telephones, free of charge. Since not all South Africans have access to the internet or to printed media, telephones therefore serve as an important alternative for accessing government information. [14]The underlying telephony platform used is Asterisk with focus on integration of Test-To-Speech and Automatic Speech Recognition technologies with Asterisk[15]  Zahir Koradia and Aaditeshwar Seth [16] deployed PhonePeti, an answering machine system at a Community Radio station in north India called Gurgaon Ki Aawaz(GKA). PhonePeti is implemented as an application over Asterisk. A GSM gateway is connected to the computer running Asterisk to terminate GSM calls and redirect them to Asterisk for further processing. Thus when a listener calls the mobile number associated with PhonePeti, the call lands on Asterisk via the GSM gateway. At this point, Asterisk begins execution of the telephony application which prompts the user to leave a message after a beep. The application records the message from the caller and pushes it to a web server to make it available to the station staff over the internet. 23
  • 4. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013  Miroslav Voznak, Lukas Kapicak, Jaroslav Zdralek, Pavel Nevlud and Jan Rozhan[17] implemented a SW PBX Asterisk with capabilities for recording and playback of video files. Native Asterisk supports audio and video calls but do not allow recording and playback of video files. In this system, the video files were directly saved into a directory in the server and a list generated and read out to the user by means of speech synthesis. DTMF key input from the user was used to select the desired file.  Raphael Frank, Benjamin Hourte, Dan Jungels and Thomas Engel[18] developed a communication tool to enhance the co-ordination and response time during emergency situations. The application is supposed to enable group calls among different rescue groups in such situations. Additionally they also presented a video application that provides real time access to surveillance cameras. The group call system was based on a centralized client-server model to allow concentrating all relevant information in one central point to obtain a better view of the crisis situation. The central entity used in the system was a Voice-over-IP server based on Asterisk. The server was equipped with a additional PRI cards1 to connect to PSTN/ISDN and GSM networks so as to interconnect the VoIP communications with traditional circuit switched communications.  Iasonas Polakis, Georgios Kontaxis and Sotiris Ioannidis[19] implemented a system to protect landlines from DIAL attacks.The core component of the platform is asterisk PBX. They crafted a CAPTCHA called Phone CAPTCHA. When an incoming call is received, Asterisk places the call in the call queue. The caller then receives a Phone CAPTCHA and has a limited amount of time to answer the CAPTCHA test using the phone dialpad. The Phone CAPTCHA test requires the caller to press a sequence of keys based on the instructions presented to him by recorded message. If the caller provides the correct answer, Asterisk forwards the call to the destination as determined by the dialplan. Otherwise the call is dropped. This mechanism provides automated calls from binding to the end device and consuming resources which could prevent legitimate callers from reaching the destination number.  Munir B.Sayyad, Nilesh Nikam, S.L Nalbalwar[20] examined the security issues in IP-PBX and suggested mechanisms to resolve them. Security issues that arises in an IP-PBX are dummy/blank/fake CLI(Caller Line Identification) configuration at IP-PBX or dummy IP addresses or IP phones connection to IP-PBX. For blocking of calls with dummy CLI or IP address, they suggested to maintain a mapping table of the IP addresses with assigned subscriber numbers(CLI ) at MSC end. If a call comes from IP-PBX with IP address and CLI that is not matching with the entries in the mapping table of MSC, such calls would be rejected. This is authentication of both IP address and assigned CLI number. Only authenticated calls will be allowed.  Ndaohialy Manda-Vy Ravonimanantosa, Andry Auguste Randriamitantosa and Malalatiana Hajasoa [21] defined a relationship between the number of calls made in an Asterisk server and use of the processor in this server as the processor is one of the major hardware devices. They calculated the correlation and found that there was a close relation between the number of calls made and percentage of CPU usage. The relationship led to the inference that if the number of calls continues to increase, it could lead to a limit so that this limit represents the physical capacity the Asterisk server.  Kranti Kumar Appari [22] developed a customized IVR implementation using VoiceXML on VoIP platform. VoiceXML delivers its output in form of audio which will be very much useful to all kinds of people. When the user places a call to the designated VoiceXML extension, the particular code for call flow gets activated. After the session is established successfully, Asterisk Gateway Interface call the initial VoiceXML browser and calls the initial VoiceXMl script hosted 24
  • 5. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 on the same server. VoiceGlue starts interpreting the VoiceXML code for audio output and the user input by way of DTMF signals or voice commands. Once the user is finished with his operation on VoiceXML the control is handed over to Asterisk(IPPBX) .  Miroslav Voznak and Jan Rozhan [23] developed a speech quality monitoring tool in accordance with PESQ (Perceptual Evaluation of Speech Quality ) that automatically runs and calculates the MOS(Mean Opinion Score). As the heart of the system, they configured the Asterisk PBX to make a call regularly every 5 minutes to the SIP/GSM gateway, which is connected to Asterisk PBX with the SIP trunk over the LAN. The call is handled by the gateway and sent to the GSM network over a GSM interface equipped with a SIM card. The call is destined to another GSM interface of the SIP/GSM gateway from where it is routed back to the Asterisk PBX. Thos way they created a loop which starts and ends on the Asterisk PBX. This approach was mainly beneficial because of the huge configuration capabilities of the Asterisk PBX.  Sadaf Nawaj, Saman Cheema, Sana Akram, K. Shehzad and S. Muzahir Abbas [24] introduced a new procedure for Vertical Handoff(VHO). It is a mechanism in which multiple users having access to different networks can execute the VHO in such a manner that when handoff takes place, it do not interrupt any ongoing call or data session. A server and clients were created. By employing a technique called IP Assignment VHO was carried out successfully. Asterisk was chosen as the server taking into consideration the facts that Asterisk system controls all routing information as well as it supports SIP which is the most important component to establish, maintain and terminate a connection. Also Asterisk supports different soft-phones that were used as clients in the system.  Lerato Lerato, Molise Mokhomo, Thato Pebane and Lerato Mohapi[25] introduces a VoiceXML gateway that comprises the integration of OpenVXI and an Asterisk telephony platform; focusing on the addition of the HTK-based speech recognition capability of the system. Finally the paper reports the successful implementation of a recognition interface that conforms to the SRGS specification. SRGS is a standard that defines the syntax representing grammars to be used in speech recognition. 4. PROPOSED SYSTEM Our system will maintain a client-server architecture wherein a PC installed with Asterisk will act as a server. A Bluetooth enabled mobile phone will remain connected to the server via Bluetooth. Any phone that makes a call to this particular mobile phone is a client that accesses the services offered by the server. The server will have a preprocessed rule set on what task to accomplish when a certain command is issued by the client. The commands along with their corresponding usage are made available to the client on demand. The client would accordingly issue a command to be executed on the server for the desired action. Prior to execution, the server will check the command for appropriateness. A command appropriately issued would execute the corresponding program on the server while any inappropriate command or any unsuccessful execution of a program would send a negative feedback to the user. 25
  • 6. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 Figure 2. Block diagram of proposed system The core of the system is the Bluetooth connection between the PC and the mobile phone. Once the Bluetooth address of the mobile phone is specified in the chan_mobile.conf file of Asterisk which is the Bluetooth configuration file for Asterisk, any calls made to this mobile phone are redirected to the Asterisk server. The server can then be made to play around with the call by making appropriate configurations in the Asterisk dialplan which is specified in the file extensions.conf. The dialplan is the heart of any Asterisk system that is built of a multitude of configuration contexts, each configuration context containing the logic to be performed in accordance o a dialed number or a dialed pattern. In our case, we invoke a phpagi script that defines the sequence of actions to be performed once a call is received by the server. Once a call is received, the phpagi script reads out a menu of available options to the user by means of speech synthesis. This is accomplished using the text2wave program of the Festival framework that takes as input plain text and converts it into audible speech output. The menu provided to the user here is a list of operations to be performed on the server computer along with the corresponding key to be pressed by the user in the dialpad. The user accordingly feeds in the DTMF key input. This enables DTMF based control of a computer system 26
  • 7. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 5. IMPLEMENTATION DETAILS 5.1. Installing Asterisk and configuring it for Bluetooth Asterisk 1.8 is installed on a Ubuntu10.04 machine and appropriate configurations made on the chan_mobile.conf and extensions.conf file in the /etc/asterisk directory. As already mentioned earlier, the chan_mobile.conf file is the blue-tooth configuration file for Asterisk, therefore in order for the Asterisk server to be able to listen to other Bluetooth devices, we first of all need to find out the Bluetooth adapter MAC address of the server and add it to the chan_mobile.conf file. Once this is done, the next step is to pair the Bluetooth enabled mobile phone to the server. This is done by first scanning the phone using the mobile search command from the asterisk console that provides Bluetooth details of the phone. All that is to be done next is to provide these details in the chan_mobile.conf file under a particular adapter and direct it to a particular context defined in the extensions.conf file. Depending on the context we put it in, if a call comes in to the mobile phone and the phone is within the proximity of the Asterisk server, it should route the call to the server. The context contains the logic o be performed in accordance to a dialled number or pattern. As have already been mentioned earlier, in this particular case, we invoke a phpagi script within the context for the sequence of actions to be performed once a call is received. 5.2. Implementation of different services 5.2.1. Execution of different system commands on the server computer through DTMF The first step towards the implementation of the framework involved the execution of different system commands on the computer. Our framework being based on Linux, the implementation involved the execution of different Linux system commands. Though there are a multitude of commands to be executed on a Linux system, we have concentrated only on a handful of them; primarily those that would enable a user to navigate through the file-system and perform some of the very often required operations on it‟s contents. The list of commands along with the key to be pressed in the dial-pad is read out to the user. The user accordingly chooses the desired command by pressing the appropriate key in the dial-pad 27
  • 8. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 Figure 3. Execution of system commands part-I 28
  • 9. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 Figure 4. Execution of system commands part-II After having successfully controlled the in-house resources of the computer, we next concentrated on exercising the capabilities of the server for accessing the resources external to the computer system and relaying the retrieved information to the user through narrated voice in the same way as we have done for the in-house resources. 5.2.2. Voice based retrieval of weather information through DTMF tone For the retrieval of weather information, we have used the WorldWeatherOnline free API. This API provides the current weather condition and next 5 days of accurate and reliable weather forecast. The API returns weather data in XML, JSON or CSV format and contains weather elements like temperature, precipitation, weather description, weather icon and wind speed. We have initially retrieved the weather forecast of four cities in XML format and have used the XMLDOM parser to parse the XML response and retrieve the important weather elements. The user is provided a menu listing the cities and the user is supposed to select one among these for which he wishes to know the weather condition which the user does by a DTMF key input. As 29
  • 10. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 already mentioned, the response is received in XML format which is parsed and is then integrated with our phpagi solution to convey the message to the user via voice over the telephone. Figure 5. Retrieval of weather information 5.2.3. Voice based retrieval of current currency exchange rate through DTMF tone We have used the „THE EASY API‟ free API for the current currency exchange rate and conversion from a source currency format to the destination currency format. The user is provided with a list of source and destination currency formats for the user to select the desired one through DTMF. The user is then prompted to enter the amount for conversion which is done again through DTMF. The API sends response in XML format which is parsed using the XMLDOM parser and the output conveyed back to the user via voice using speech synthesis over the telephone. 30
  • 11. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 Figure 6. Retrieval of current currency exchange rate 5.2.4. Voice based retrieval of e-news through DTMF tone This module is aimed at retrieving an e-newspaper and conveying the contents back to the user via voice over the phone using speech synthesis. We have chosen the „The Hindu‟ newspaper for the purpose. The response is received in HTML format which is parsed using the SIMPLE HTML DOM parser. The headlines are read out to the user with a query after each one of them regarding whether or not the user wants to go through the news in detail to which if the user response in an affirmative, the detailed news is read out; otherwise it is skipped and moved to the next headline . 31
  • 12. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 Figure 7. Retrieval of e-newspaper 5.2.5. Integrating the different services into a DTMF tone based Welcome Module All the services discussed above are integrated into a single module resulting in a compact computer control framework. When a connection is first set-up between the server and the user, the Welcome Module presents a list of the services to the user. Each item in the list is again divided into a menu and submenu allowing the user to effectively and efficiently control the services on the computer. 32
  • 13. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 Figure 8. Integration of different services 5.2.6. Introducing the security services To prevent misuse of the system by unauthorized users, user authentication using username and password is provided. When the user first connects to the system, the user is prompted to enter the username and password. On successful verification, the user is authorized to access the server. Authentication is particularly important because Asterisk always runs on administrator mode. Therefore anyone who gains access to the server can execute any operation on the computer system. 6. CONCLUSION AND FUTURE SCOPE In this paper, we have demonstrated a framework that makes use of Asterisk to control a computer system using DTMF and speech synthesis. The system can further be enhanced to accept voice input from the user. This will also ensure better security of the system. It can still further be augmented to offer services in local languages that would particularly be beneficial to the common people. REFERENCES [1] [2] [3] i6net* (2013) “What is Asterisk?” [Online] Available: http://www.i6net.com/what-is-asterisk/ Asterisk(2013) Asterisk Home Page [Online] Available: http://www.asterisk.org/get-started/ PCMAG.COM(2013) “Online Extra: Asterisk :An Open Source PBX Gateway” [Online] Available: http://www.pcmag.com/article2/0,2817,1607848,00.asp 33
  • 14. International Journal of Computational Science and Information Technology (IJCSITY) Vol.1, No.4, November 2013 [4] [5] [6] [7] [8] [9] [10] [11] [12] [13] [14] [15] [16] [17] [18] [19] [20] [21] [22] [23] [24] [25] M-indya(2013) “What is Asterisk? Advantages and disadvantages of Asterisk” [Online] Available: http://www.m-indya.com/shownews.php?newsid=3063 GNUAcademy(2013) “Asterisk(PBX)” [Online] Available: http://www.gnuacademy.org/ippbx/asterisk-pbx Wikipedia (2013) “Asterisk (PBX)” [Online] Available: http://en.wikipedia.org/wiki/Asterisk_%28PBX%29 Mark Spencer, Mack Allison and Christopher Rhodes, “The Asterisk Handbook Version 2” [Online] Available: http://bananabread.net/AsteriskHandbk_w_Bkmarks.pdf TRTechServices(2013) “About Asterisk(VoIP PBX)” [Online] Available: http://www.itrtechservices.com/index.php?option=com_content&view= article&id=107&Itemid=136 Slideshare(2013) “Asteriskl as a VoIP Gateway” [Online] Available: http://www.slideshare.net/Catharine24/asterisk-as-voip-gateway Asterisk(2013) “Call-center” [Online] Available: http://www.asterisk.org/getstarted/applications/call-center Asterisk(2013)”Conference Bridge” [Online] Available: http://www.asterisk.org/getstarted/applications/conference Asterisk(2013)”Voicemail Server” [Online] Available: http://www.asterisk.org/getstarted/applications/voicemail Asterisk(2013)”IVR Server”[Online] Available: http://www.asterisk.org/get-started/applications/ivr Lwaji(2013) “About LWAJI” [Online] Available: http://www.meraka.org.za/lwazi/about.php Lwaji(2013) “Scientific and technical outputs development” [Online] Available: http://www.meraka.org.za/lwazi/stop.php Zahir Koradia and Aaditeshwar seth, “PhonePeti: Exploring the Role of an Answering Machine system in a Community Radio Station in India” Miroslav Voznak, Lukas Kapicak, Jaroslav Zdralrk, Pavel Nevlud, Jan Rozhon, “Video Files Recording and Playback with VoiceXML,” Recent Researches in Communications, Automation, Signal Processing, Nanotechnology, astronomy and Nuclear Physics Raphael Frank, Benjamin Hourte, Dan Jungels and Thomas Engel, “Interoperable Networking Applications for Emergency Services”, International Journal of Multimedia and Ubiquious Engineering Vol $, No. 1,January, 2009 Iasonas Polakis, Georgios Kontaxis and Sotiris Ioannidis, ”CAPTCHuring Automated (Smart) Phone Attacks” Munir B. Dayyad, Nikesh Nikam, S.L Nalbatwar, “Novel Approach to Resolve Network Security Issues in IP PBX in Converged Architecture”, International Journal of Engineering Science and Technology Vol.2(10), 2010, 5765-5769 Ndaohialy Manda-Vy RAvonimanantosa, Andry Auguste Randriamitantsoa, Malalatiana Hajasoa Ramafiarisona, “The Impact of Calls to the Processor on a Server PABX”, International Journal of Future Computer and Communication, Vol. 1, No. 1, June 2012 Kranti Kumar Appari, “Customized IVR Implementation Using VoiceXML on SIP(VoIP) Communication Platform ”, International Journal of Modern Engineering Research(IJMER) Vol.2, Issue. 6, Nov-Dec. 2012 pp-4239-4243 ISSN: 2249-6645 Miroslav Voznak and Jan Rozhan , “Automated Speech Quality Monitoring Tool based on Perceptual Evaluation”, Recent Researches in Communications, Information Science and Education Sadaf Nawaj, Saman Cheema, Sana Akram, K. Shehzad , S. Muzahir Abbas, “Implementation of Vertical Handoff in Heterogenous Networks”, International Journal of Future Generation Communication and Networking Vol. 5, No. 3, September 2012 Lerato Lerato, Molise Mokhimo, Thato Pebane, Lerato Mohapi,” Integration of an HTK Speech Recognizer with Open VXI and an Asterisk based VoiceXML Gateway” 34