FINAL YEAR PROJECT REPORT

ULPHONE

AN ASTERISK BASED FREEPHONE SYSTEM FOR UL

Shane McCausland

13142976

Supervised by Dr. Conor Ryan

September 2016 – Apri1 2017

LM110

B.Sc. Computer Games Development

Project Summary

This project involves setting up a Private Branch Exchange (PBX) and the systems to

manage it. A PBX is a private telephone network used within a company. Users of the PBX

phone system share several outside lines for making external calls. Not only does a PBX

connect internal telephones within the system but it can also connect to the Public switched

telephone network (PSTN). The PSTN is the aggregate of the world’s circuit switched

telephone networks that are operated by national, regional, or local telephone operators,

providing infrastructure and services for public telecommunications (Rouse, 2005). Simply

put, it is the phone network used for the majority of phone calls made by the public.

Users will be able to apply for a new phone number through a web based front end which

will communicate with the asterisk based private branch exchange server.

Care and accountability will be taken to prevent abuse on the PBX. Security for a PBX

system is critical, this is due to the fact the system is exposed to the internet. There is a lot

of money to be made by attackers in exploiting systems to make free phone calls.

To understand this Final Year Project, it is first important to understand what Asterisk is.

Asterisk is an open source framework for developing and building communication

applications. It turns an ordinary computer into a communications server (Digium, 2015)

and it is the core element to my PBX system. Asterisk has many different applications such

as, an IP PBX, VoIP gateways and for conference servers and other custom solutions. It is

used by small business, large businesses, call centers, carriers, government agencies and

for personal use. It is sponsored by Digium but is built to be completely open-source

From the outset, I knew that there was a huge variety of ways to approach the project, this

was evident when researching the many different Asterisk applications used globally. The

motto of Asterisk is “There’s more than one way to do things” (Madsen, 2011), this

becomes more applicable with each release of the flexible toolkit. There is a massive

amount implementation possibilities with dozens of supported languages. Asterisk is also

compatible with all major Linux distributions.

I made the decision early in the project to implement my Asterisk based PBX in the cloud.

There were several options available to me in upon making this decision but I found the

must applicable option to be an AWS (Amazon Web Services) EC2 (Elastic Cloud

Compute) instance. The EC2 instance provided by Amazon is completely free and allowed

me to create an instance of an Ubuntu Linux server on which I could install the PBX. I had

many reasons for choosing a cloud based PBX, the main reason being; cloud based PBX’s

are the current state of the art of communications used by businesses today, with many

business abandoning their more expensive to maintain non-hosted solutions in favor of the

cloud based alternative. The many benefits of a cloud based PBX include dynamic

resizable compute capacity and savings on IT resources by outsourcing the maintenance of

the system, the up-keep of the system is managed by the host, so dedicated in house staff

is not necessary for maintenance. It can also be considered massively cost effective, mainly

due to the fact businesses can avoid paying for maintenance fees typically associated with

a traditional phone system. This contrasts with the cloud based solution which gives the

business a predictable monthly billing charge and eliminates the need for upfront costs for

heavy installations. Reliability is also a positive trait of a hosted PBX system, it can be

counted on to connect every call without lagging response times, delays or lapses in over

that of traditional PSTN connection. In many legacy systems, the first thing that is affected

during an unforeseen event such as a storm, fire or power outage is the phone lines. A

hosted PBX solves this issue as the infrastructure is protected in the hosted environment.

As such, flow of communications can continue.

Acknowledgements

I would like to convey my sincerest gratitude and earnest appreciation to the following

people who have supported me throughout this Final Year Project:

My project supervisor, Conor Ryan, for his extremely beneficial guidance throughout this

project and his continuous advice to keep me on track to complete the project.

I would also like to thank the University of Limerick’s Department of Computer Science

and Information Systems for their four extremely educational years of learning and for the

high-quality facilities made available for students.

I would also like to thank my friends for helping with the user testing of the project by

completing the feedback sheet. Your help was greatly appreciated.

Finally, I would like to thank my family for their constant encouragement and level of

understanding throughout this academically challenging year.

Without this support and advice, the Final Year Project would have been an unreachable

goal.

Declaration Page

I hereby declare that the development of this Final Year Project was undertaken without

the illicit assistance of any third-party source or aid other than those specified and that this

dissertation, in whole or in part, has not been submitted to any University as an exercise

for a degree. I further declare that, except where reference is made in the text, the contents

are entirely my own work. The author agrees that the library of Final Year Projects within

the department of Computer Science and Information Systems of the University of

Limerick may lend or copy this thesis upon request for study purposes, subject to normal

conditions of acknowledgement.

___________________

Shane McCausland

18th April 2017

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Contents 1. Project Introduction ........................................................................................................ 3

1.1. Overview .................................................................................................................. 3

1.2. Motivations .............................................................................................................. 4

1.3. Objectives ................................................................................................................ 4

2. Research .......................................................................................................................... 6

2.1. Existing products ..................................................................................................... 6

2.1.1. PBX in a Flash ................................................................................................................. 6

2.1.2. Free PBX ......................................................................................................................... 6

2.1.3. Asterisk Now .................................................................................................................. 6

2.2. Relevant research ..................................................................................................... 7

2.3. Implementation technologies ................................................................................... 7

2.3.1. AWS EC2 ......................................................................................................................... 7

2.3.2. LAMP .............................................................................................................................. 8

2.3.3. Fail2Ban .......................................................................................................................... 9

2.3.4. Go Trunk ......................................................................................................................... 9

2.3.5. PHPStorm ..................................................................................................................... 11

2.3.6. FileZilla & PuTTY ........................................................................................................... 11

2.4. Core Concepts & Definitions ................................................................................. 13

3. Implementation ............................................................................................................. 16

3.1. Introduction ............................................................................................................ 16

3.2. AWS EC2 Instance Configuration and Asterisk Installation ................................. 17

3.2.1 Making a call ................................................................................................................. 20

3.3. Web based front end development ......................................................................... 21

3.4. Connection between front end and server. ............................................................. 23

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3.5. Connections to PSTN using go trunk..................................................................... 26

4. Security ......................................................................................................................... 27

4.1. Issues and Complications ....................................................................................... 27

4.2. Solutions and Protection Methods ......................................................................... 28

5. System Testing & Evaluation ....................................................................................... 29

5.1. General Introduction .............................................................................................. 29

5.2. Testing methods ..................................................................................................... 29

5.3. Findings.................................................................................................................. 29

5.4. Implementation of user feedback ........................................................................... 30

6. Summary & Conclusions .............................................................................................. 31

6.1. Conclusions ............................................................................................................ 31

6.2. Recommendations .................................................................................................. 31

7. References ..................................................................................................................... 33

8. Bibliography ................................................................................................................. 35

9. Appendices .................................................................................................................... 36

9.1 Appendix 1: User testing sheet ............................................................................... 36

9.2 Appendix 2: Screenshots & Images ........................................................................ 37

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1. Project Introduction

1.1. Overview

This final year project, UL Phone, involves setting up a private branch exchange (PBX)

system and systems to manage it. Users should be able to apply for a new phone number

on a web based front end and use the number on the PBX. Care and accountability should

be taken to prevent abuse. The PBX will be able to connect to the PSTN (Public switched

telephone network) and allow users of the system to request credits to make calls on the

PSTN.

While there are many open-source PBX’s available online, such as PBXInAFlash or

OpenPBX which would have worked for this Final Year Project, I chose to create my PBX

using Asterisk. Asterisk is the core framework when it comes to creating custom PBX’s

and it is what the majority of the open source communications software is built on.

The web based front end will act a medium between the user and the Asterisk based PBX

server. Users will register with an appropriate student or staff email, name and password

and will be issued with an extension within the PBX. The web front end will also collect a

directory of registered extensions with their associated names.

Once issued with an extension the user will be able to connect to the PBX server using a

software based telephone (softphone) and make free calls to other users within the PBX

registry.

The Asterisk server is cloud based, using an Amazon Web Services (AWS) based Elastic

Cloud Computing (EC2) Instance. I made the decision to make the server cloud based to

support scalability and because having a cloud based PBX is becoming the current state

of the art in PBX technology.

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1.2. Motivations

My reasons for undertaking this final year project are motivated by the fact that I worked

in a call center for several years, I was extremely interested in learning the technology

behind such a huge industry. Learning about the power and scale of which Asterisk is used

in business is another factor that motivated me into picking this project. Today there is

more than 1 million Asterisk-based communications systems in use, in more than 150

countries. Almost the entire Fortune 1000 list of customers uses it (Digium, 2015).

The fact that Asterisk is so accessible and open source is another reason which aided me

in choosing this project, the ability to get under the hood and see how Asterisk works, along

with making any changes or enhancements I see fit is a very interesting aspect of the project

for me.

The Asterisk community is also very well supported, with a vast knowledge base on the

official website as well as many in-depth textbooks available for all skill levels it is easy

to pick up if you are a beginner.

Creating the PBX in the cloud was another motivating factor, this was because I was

looking for a topic which gave me some experience deploying software on a cloud based

system as cloud computing is taking over in the IT industry.

1.3. Objectives

My key objectives for this project were, to research and learn about Asterisk and its various

PBX based applications, to learn how to deploy an instance of an EC2 Linux server and

install Asterisk after the instance is launched and to deploy a LAMP based web front end

on the same EC2 instance and to have this web based front end communicate with the

Asterisk server.

Once I had achieved these primary objectives I hoped to give the user the ability to log into

the web based front end and request credits. These credits will enable the users to make

calls from the PBX to the PTSN. The final objectives of the project will be to polish the

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web based front end and provide security and protection on the PBX. The front end will

eventually display a directory of users which details their specific credentials on the

communications server.

As the product of the final year project is now complete, I have achieved the milestones as

intended and delivered the product in accordance to the initial requirements. I have

successfully created a functioning cloud based asterisk built PBX system wherein users

can sign up (Using the web based front end) and make free calls to other users registered

on the asterisk server. The web based front end has also been created successfully, allowing

a user to register their details to the PBX and make calls to other users as well as calls to

the PSTN. The web based front end also houses a directory of all the users within the PBX

as well as a knowledge base of information on ULPhone detailing various aspects of the

product, such as the initial phone configuration.

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2. Research

2.1. Existing products

With most businesses using a PBX for their telephony traffic, there is a vast number of

existing products and solutions available on the market. One of the very first tasks I

completed on this project was to install many different open source PBX distributions to

give me a better understanding of the technology. I focused primarily on PBX distributions

that were created with Asterisk. The three most popular distributions I researched included

PBX in a Flash, Free PBX and Asterisk Now, each with their own respective advantages

and disadvantages.

2.1.1. PBX in a Flash

This PBX comes with the latest version of the Debian Linux operating system and installs

in under 30 minutes. One of the key advantages to this distribution is the ease of

upgradability. It is also preconfigured with trunks and gateways. Trunks are the

connections shared between two entities on the PBX while the gateway is the way in which

the PBX accesses the PSTN. Asterisk 1.4 or 1.6 come preloaded with this distribution.

2.1.2. Free PBX

FreePBX is the world’s most deployed open source business communications system, with

millions of active installations and over 400 new installs per day (Sangoma Technologies,

2010). The distribution is incredibly feature rich. It also has support for other advanced

add-ons such as Zulu, which is an all in one desktop softphone system, and other phone

based applications. It can be installed a number of ways and is very versatile. I installed

the PBX on a very old laptop and it worked perfectly, allowing me to make phone calls

between two configured softphones.

2.1.3. Asterisk Now

This a complete Linux distribution with Asterisk, the DAHDI driver framework (Section

3.2), and the FreePBX administrative GUI. The installer handles all the complexity,

making this PBX very accessible in terms of ease of use.

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All three of these PBX distributions helped to give me a better understanding of the

Asterisk technologies.

2.2. Relevant research

My core piece of research material was the textbook ‘Asterisk the Definitive Guide’ by

Leif Madsen, Jim Van Meggelen and Russel Brand which I purchased for this project. The

book details the process involved in designing a complete VoIP or analogue PBX with

Asterisk. I used the book as a reference when I was installing Asterisk on the EC2 instance.

It details the different commands needed for the install of Asterisk on each major Linux

distribution.

It gave me a basic understanding of some key telephony concepts such as trunking, dial

plans and the configuration of devices (defined in section 2.6.7) which enabled me to work

better and more confidently with the PBX system. The book also explains how to connect

to the PSTN which is extremely relevant for my project.

2.3. Implementation technologies

2.3.1. AWS EC2

The core piece of technology I am using for this project is an Amazon Web Services Elastic

Cloud Compute instance running an Asterisk PBX on an Ubuntu Linux server. This allows

me to have a cloud based PBX. Figure 1 shown below shows the configuration on a hosted

PBX system, similar to what is used for ULPhone. My reason for choosing an Amazon

based cloud instance is that the AWS EC2 instance is scalable in response to popularity

spikes, so as more people use the PBX the instance will scale in response to this.

Amazon EC2 is also a highly secure method of operation, it provides secure login

information for the instances using key pairs. AWS stores the public key while I store the

private key. A firewall that enables the user to specify the protocols, ports and source IP

ranges is also available to further enhance the security of the instance (Microsoft, 2016).

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An EC2 instance is also incredibly easy to set up for a first-time user, it is as simple as

clicking a button and choosing which operating system you want your instance to have.

The final reason I had in choosing AWS is their free tier account options, which mean I

did not have to pay for any of my EC2 instances. The free tier access lasts for a year and

allows the user to have 750 hours per month of Linux EC2 usage (Amazon, 2015) which

is perfect for what I will be undertaking.

Figure 2.1: Hosted PBX Diagram.

2.3.2. LAMP

The LAMP (Linux, Apache, MySQL, and PHP) model of web service stacks is what the

web based front end will be developed with. One reason for this is that I can install all the

web services on the same EC2 instance as the Asterisk server allowing for easier

maintainability and cost effectiveness. LAMP is also a good choice for the project as it is

open-source, and has a huge community which offer support and have also developed a

wide variety of extra components and modules which give extended usefulness. It is also

widely regarded as a very secure method of web service development.

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2.3.3. Fail2Ban

Fail2Ban is an intrusion avoidance software framework that protects computer servers from

brute force attacks. Written in the Python programming language, it is able to run on

POSIX (section 2.4) systems that have an interface to a packet-control system or firewall

installed locally, for example, iptables or TCP wrapper.

While Fail2ban can reduce the rate of incorrect authentication attempts by banning the IP

address of the brute force attacker from the PBX server, it is by no means a substitute for

strong authentication procedures. Fail2ban should be used in combination with strong

authentication methods in order to achieve maximum security of the PBX server as it alone

cannot eliminate the threat weak authentication presents.

I chose to use Fail2ban for my system as it is very proficient at working with Asterisk

systems. It is very common for Asterisk based PBX to have Fail2ban scanning the log files

for breach attempts. Once the scanning has found a possible breach on the communications

server the IP address is added to an IP table and banned from accessing the server for an

administrator specified amount of time.

2.3.4. Go Trunk

GoTrunk is a SIP trunking service that provides inbound numbers for the PBX at a cost

and connects the PBX to the PSTN (public switched telephone network). A SIP trunk is

necessary for ULPhone as it acts as a tunnel from the created PBX communication server

to the PSTN. A SIP trunk utilizes the Session Initiation Protocol (section 2.4) which is an

IETF (Internet Engineering Task Force) standard that dictates how voice over IP (VoIP)

sessions are initiated, handled and terminated. While there are other proprietary protocols

available such as IAX (section 2.4), SIP has become the most commonly utilized and

implemented format.

When a phone call is made over the SIP trunk, it may begin as an IP call or eventually be

converted into one at some point. The point in question maybe an ATA adapter, which

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converts an ordinary analogue telephone signal to an IP call, the IP PBX itself or a SIP

compatible router.

While there are many SIP trunk providers available. One such provider is Blueface which

is an Irish owned SIP trunk provider, the SIP trunk for ULPhone was GoTrunk for the

following reasons:

• Flexible month’s trial for first time users: This was essential for this project as it

allowed me to test the connection to the PSTN without having to give credit card

details. This meant I could test ULPhone connectivity to the PSTN without the fear

that a breach by an attack would cause loss of money. This also allowed me to test

the security of the PBX once it was connected as any communication server which

can make calls to the PSTN is a target for hackers.

• Extremely versatile control console: The GoTrunk control console is a defining

characteristic of the SIP Trunk provider and indeed a benefit. From the dashboard,

the administrator of the PBX can view detailed analytics of calls made from their

PBX server through the SIP Trunk as well as detailed call logs of previous calls

made. The dashboard also includes a knowledge base of Country Dialing Codes

with their associated cost.

• Pay as you go system: GoTrunk works on a pay as you go system. You only spend

the money you need at that time. This means that there is no fear of exceeding

expected cost as the administrator determines the level of credit in the SIP trunk

which can be used to make calls. This is also something which enhances security

of the PBX, because there is only a set amount of credit in the PBX server at one

specific time, a hacker can never go over the amount which is present to make free

calls to international or long-distance phones.

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2.3.5. PHPStorm

PHPStorm is a commercial cross-platform interactive development environment for PHP

built on the IntelliJ IDEA platform and it is the IDE I chose to use for ULPhone’s front end

development. The main reason I was drawn to PHPStorm was the positive reputation it has

received as a development environment and the fact it is built upon the extremely robust

IntelliJ IDEA platform. Throughout the development of the front end I the most common

and extremely useful feature I used included:

• The IDE’s smart code completion: As someone who is relatively new to PHP

development the development environment’s code completion, syntax highlighting

and automatic refactoring were hugely beneficial to my development process.

• Debugging & testing: PHPStorm had many debugging tools which I frequently

made use of to inspect variable, set breakpoints and evaluate an expression in

runtime. PHPUnit tests can be developed in the editor with statistics on code

coverage to show what code is covered with the developed tests.

2.3.6. FileZilla & PuTTY

The project’s Linux instance is a virtual server in the Amazon Web Services cloud. To

connect to this instance, an SSH client was needed. I chose to use PuTTY for this function.

As well as having the SSH client to connect to the Linux instance, I needed to have a

method of file transfer to the server, I chose to use the FTP solution FileZilla. All of the

work done with Asterisk was completed using an instance created by PuTTY and all of the

front-end development files were uploaded to the EC2 server using FileZilla. In order to

connect to the instance securely I had to make use of the EC2 key pair authentication

method.

Amazon EC2 uses public-key cryptography to encrypt and decrypt login information.

Public key cryptography uses a public key to encrypt a piece of data, such as a password,

the recipient uses the private key to decrypt the data, the public and private keys are known

as a key pair. In order to login into the instance with PuTTY, I had to create a key pair,

which is done in the EC2 control console, the key pair is also named in this process and is

associated to the public key which is stored by Amazon. The private key is then

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downloaded to a secure directory. When logging into the instance the private key is given

to PuTTY and acts as a password to EC2 instance (Amazon 2017). Amazon stores the

public key only, and I, as the administrator of the EC2 instance store the private key.

Anyone who possesses the private key however, can decrypt login information so it is

important to keep it secure.

Figure 2.2: Connection to EC2 instance using PuTTY.

Figure 2.2. shows the virtual machine running on windows. It also shows the asterisk CLI

which is entered by using the ‘asterisk -r’ command.

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2.4. Core Concepts & Definitions

Trunking: Is the process of using a SIP trunk to connect the PBX to the PSTN. The sip

trunk itself acts as a gateway between the PBX and the PSTN.

Dialplans: Dialplans are essentially a scripting language specific to Asterisk and one of the

primary ways of instructing Asterisk on how to behave. It ties everything together, allowing

the PBX admin to route and manipulate calls in a programmatic way (Jordan 2016).

Contexts: The Asterisk dialplan is divided into sections, and each section is known as a

context (Wintermeyer 2010). Any dialplan must include a [general] context where

configuration entries that are global are located. Any subsequent context can have any

name. Contexts act as means by which any physical devices, in ULPhone’s case

Softphones, are bound to the dialplan. An example context specification from the sip.conf

would include the following.:

In the above example the sip device is called alice and will initiate calls in the hardcoded-

phones context. This means that if the SIP device alice calls another device by dialing an

extension, Asterisk will then search the context hardcoded-phones for a device matching

that extension and will proceed to dial it if it exists. If it does not exist Asterisk will do

nothing. The actual context itself is defined by its name within square brackets:

[alice]

type = friend

context = hardcoded-phones

secret = 1234

host = dynamic

[hardcoded-phones]

Rules, instructions etc.

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Endpoints:

1. Hardphone: A hard phone is a physical device. It looks just like an office telephone,

it has a handset, numbered buttons etc. It connects directly to the network and it is

commonly referred to as a VoIP telephone.

2. Softphone: A softphone is a software application that runs on a computer or smart

device and acts as a phone that connects to the PBX. The audio passes through the

devices audio system. This is the primary end point which is used for ULPhone as

it is a portable and cost effective solution. The softphone application Zoiper is the

softphone used, it can be downloaded for free on all major OS’s and smart devices

for free.

3. ATA: An analog terminal adapter (ATA) is designed to allow traditional analog

telephones and other machines such as fa machines to connect to a SIP network or

PBX.

Asterisk Realtime Architecture (ARA): The ARA moves all database specific code to

database specific drivers. It is a method of storing the configuration files (normally located

in the /etc/asterisk directory) and their configuration files in a database table (Madsen et

al. 2013). There are two types of Realtime;

• Static: Similar to the tradition method of reading a configuration file, except that

the data is read from the database.

• Dynamic: This is used for things such as user and peer objects (SIP, IX2) which

loads and updates the information as it is required.

POSIX (Portable Operating System Interface): POSIX is a set of standard operating system

interfaces based on the Unix operating system (Rouse 2005).

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SIP (Session Initiation Protocol): is an application level signaling protocol for setting up,

modifying and termination real-time sessions between participants over an IP data network.

(Stallings 2016). SIP is commonly used for telecommunications applications and is the

most popular protocol used in Asterisk, which is why I chose to use it for ULPhone. Figure

2.3 shows how SIP components relate to one another and the protocols that are employed.

A user agent (in this case Alice) uses SIP to set up a session with another user agent that

acts as a server (Bob).

Figure 2.3: SIP components and protocols.

IAX (Inter-Asterisk eXchange) – is another communications protocol used by Asterisk. It

differs from SIP as it is native to Asterisk while SIP has a wider variety of applications. It

is used primarily for transporting VoIP telephony sessions between servers and to terminal

devices.

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3. Implementation

3.1. Introduction

One of the most important tasks in the initial development of ULPhone was planning the

work to be done. Using various project management techniques such as work breakdown

structures and Gantt charts I set very clear milestones for myself. From the outset this made

the project more approachable as it broke the development down into sections rather than

having one huge project with no clear starting point or goals.

I also initially outlined the success criteria for the project before starting any stage of actual

development. The core goals for the project as defined by myself with advice from my

supervisor, Conor Ryan, included:

• Create a functioning PBX installed on a cloud based EC2 instance.

• Develop a functioning web based front end which users can register to with

their credentials and login to a user portal.

• The developed front end will have the ability to assign an extension to the

newly registered user and communicate the user details to the asterisk based

PBX.

• The web based front end will also have a directory of users registered of the

PBX.

• The newly registered user will be able to register to the PBX using a

softphone of their choice and make calls to other users of the PBX for free.

• The PBX users will have the ability to make calls to the PSTN if authorized

by the administrator of the communications sever.

Once the main planning element of the project’s development was completed and the

success criteria was defined the project was split in to four core milestones.

1. Asterisk setup/installation on the AWS EC2 instance.

2. Development of the web based front end.

3. Configuration of the intercommunication between the front end and Asterisk based

PBX.

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4. Configuration of the PSTN connectivity.

Once I completed these four core milestones, the project had met most of the specified

success criteria and was considered a success. The final stages of securing and testing

ULPhone were then undertaken. The way in which the core milestones were undertaken is

specified in the following sections

3.2. AWS EC2 Instance Configuration and Asterisk Installation

This aspect of the project was initially very challenging a number of reasons. The key

reason being that, even though there is a lot of support for Asterisk installations for specific

systems, there is a lack of material for installing it on a cloud based instance, more

specifically an AWS cloud based instance. I had to combine various methodologies to get

the communications framework running in the cloud.

The first step taken when installing Asterisk was to correctly initialize the AWS EC2

instance. To create the instance, I had to sign up for an Amazon AWS account which was

a straight forward process. AWS offers various tiers of accounts when signing up, some

geared towards enterprise level systems and some towards small personal systems. The tier

I chose was the free tier account. The account type entitled me to 750 hours per month of

always on micro instance usage for free. This was more than enough compute time for

ULPhone as it is only possible to exceed the plan’s constraints if more than once instance

is running, and, in the case of ULPhone only one instance is necessary as Asterisk, and the

web based front end are installed on the same single EC2 instance.

Once the account was registered, the setup of the instance was very intuitive, all that was

necessary was clicking the launch EC2 instance button from the control console home page

and specifying a few user preferences. AWS EC2 allows the user to specify which flavor

of Linux to install with various options including:

1. The Amazon Linux AMI: which is an operating system developed by amazon

specifically for EC2.

2. Red Hat Enterprise Linux.

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3. SUSE Linux.

4. Ubuntu Linux: Flavor used for ULPhone.

5. Windows Server OS

Once this process is complete the instance is launched, the EC2 control console can be used

and the virtual machine can be connected to using SSH. The EC2 control console is tailored

to act as a central hub for the instance. It presents all the instance information as well as UI

to perform various functions such as:

• Launching, terminating, or pausing an instance.

• Assigning an elastic IP address. An elastic IP address is a static IPv4 address made

specifically for cloud computing. It is associated with the AWS account and, in the

case of ULPhone it is used as the domain name for the web based front end.

• Creating security groups and forwarding ports for SSH, HTTP and SIP

Once the AWS EC2 was complete I began the Asterisk installation process. The first tasks

in this process included installing the various dependencies such as

• Libpri: Libpri is an open source library that encapsulates the protocols used to

communicate over ISDN interfaces. ISDN, the Integrated Services Digital Network

is a set of communication standards for simultaneous digital transmission of voice,

data, and other network services over the traditional circuits of the public switched

telephone network (Alfian’s Linux 2013).

• DAHDI – DAHDI is the Digium Asterisk Hardware Device Interface, which is a

collection of open source drivers for Linux that are used to interface with a variety

of telephony related hardware.

Once these dependencies were installed I was able to begin the Asterisk installation. I

chose Asterisk 13 as the version to be installed on the EC2 instance. Asterisk 13 was

released in 2014 and many user reviews says it is the best and most robust version of

Asterisk. The focus of the development of Asterisk 13 was improving the usability and

features and features developed in the previous standard release, Asterisk 12. To install

Asterisk on the EC2 I followed instructions from the textbook ‘Asterisk, the Definitive

Guide’, which details the commands necessary to install the framework. Once the

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install is complete the user is presented with the Digium logo to indicate a successful

installation (Figure 16 Appendix 2).

Once Asterisk was installed, I completed the initial configuration by modifying the

following configuration files, which are installed with asterisk into the /etc/asterisk

directory of the EC2 instance to my own specific preferences for ULPhone:

• modules.conf: This configuration file tells Asterisk what modules to load on

startup. A module provides its own specific functionality and capability to the

Asterisk PBX such as voicemail or the automated voice attendant module.

While this file is not strictly required in an Asterisk installation, without it

Asterisk won’t have any modules loaded and won’t be able to do anything.

• asterisk.conf: This file allows for tweaking of various settings that affect how

asterisk runs as a whole. I used this configuration file to edit the directory path

of where Asterisk stored the configuration files on the EC2 instance as it was

different on the initial Asterisk generated file.

• sip.conf: This file specifies the configuration of the Session Initiation Protocol

(SIP) module. Channel. It is also used for defining parameters which SIP

devices communicate with the Asterisk communications server. Initially to test

the PBX, I hardcoded to users to the sip.conf file in other to test my first call on

the PBX. Once the intercommunication between the front end and the Asterisk

server was configured, the sip.conf read from a database of registered users in

order to dynamically register these users to the PBX in real time as well as

assign them an extension using the extensions.conf file.

• extensions.conf: I used this configuration file to generate an extension number

for the SIP users specified in the sip.conf file.

After I completed the installation of the Asterisk framework on the AWS EC2

instance and the initial configuration I made my first call. To do this I hardcoded to

users to the sip.conf file, I then downloaded a softphone and registered it to the

PBX, I signed in using one of the hardcoded users, repeated these steps on another

device with the other hardcoded user and then I called the extension of the other

signed in user which proceeded to make the other softphone notify me of an

incoming call.

20

3.2.1 Making a call

Once the asterisk server has been installed and configured correctly on the EC2 instance a

test call can be made. To make a call I completed the following steps:

• Initially I configured two users in the sip.conf file and assigned

them specific extensions within the extensions.conf file.

• Using my preferred android Softphone Zoiper, I entered the

specific username and password of one user I had created. For the

host credential, which is necessary to connect to the PBX I

entered the elastic IP/domain of the front end followed by the SIP

port being used which in the case of ULPhone was:

35.160.192.57:5080. Figure 3.1 shows the account information

input page of the Zoiper softphone.

• I repeated this process on a windows softphone for the other user

with their specific username and password.

• Once both users were registered successfully, which was indicated

by the screen shown in figure 3.2. I made a call between them by

dialing one user’s extension from the other user’s softphone interface.

• All SIP user’s extensions within the PBX begin with a 6 and when

dialing out to the PSTN the extension begins with a 9 followed by the

PSTN phone number.

Figure 3.1: Zoiper

user data input.

Figure 3.2 - Zoiper

registration successful.

21

3.3. Web based front end development

This was the most important step I found in the development of the web based front end

was knowing exactly the functionality I wanted it to have in the system. The main

functional requirements and nonfunctional requirements I developed the front end to have

are as follows:

• Functional Requirements

➢ Provide a platform to register users and assign them an extension. The

extension acts a phone number within the PBX.

➢ Provide a directory of users in registered on the PBX. The directory displays

the name of each user, their extension in the PBX and other credentials such

as email and surname

➢ Provide a functional comprehensive knowledge base of the ULPhone

system. This knowledge base details the initial configuration of the user’s

softphone, information on the various softphones available as well as their

pros and cons, an about section detailing the purpose of ULPhone as well

as a contact form which users can use to request credits to make calls to the

PSTN or seek help with a problem. In the final product, all users can make

calls to the PSTN as they are assigned one credit in the system. The reason

for this being for the demonstration purposes of ULPhone, the second reader

can automatically make calls to the PSTN without needing approval initially

from the admin.

• Non-functional Requirements

➢ Usability & performance on mobile devices. I want ULPhone to act as an

all-in-one system, such that a user of the system can perform all the products

functionality on one device, be it on mobile or computer. The front end was

developed with this in mind, across all platforms the front-end scales

appropriately to appear in a user-friendly manner.

➢ Security. Every piece of user data should be safe and the PBX should be

safe from attacks as there is a lot of money to be made in exploiting a PBX

system to make free calls.

22

With these requirements in mind I decided to use the open-source LAMP web development

platform for the front end. LAMP which is the acronym for Linux-Apache-MySQL-PHP

is the most popular web development platform available. One reason I chose LAMP as the

platform for ULPhone is due to the fact it is a very well supported for platform for

development on EC2 instances. Being new to PHP development, a LAMP based platform

suited my needs perfectly as there is a vast amount of documentation and support available

on using the language and platform. Some other reasons LAMP was well suited to this

project were:

• Customization: Because LAMP components are open source, they have

accumulated a wide variety of additional components and modules that provide

additional functionality. This open source approach enabled me to do the same in

development, that is, customizing components and functionality to suit my specific

needs.

• Ease of deployment: With no licensing issues or the need to compile applications

deployment was as simple as copying the locally developed.php files to the EC2’s

/www directory using FileZilla.

• Security: Since many people use the platform and because it has been available

and used for years LAMP technology is secure and stable. Problems are fixed very

quickly, so there was no need for me to worry about the initial security of ULPhone.

• Community: There is a wide array of experienced people willing to provide help

and support for the development and deployment of LAMP based applications. This

was particularly useful in the case of ULPhone due to my initial low level of

experience with LAMP at the beginning of the project.

The base layout of every page on the front end was developed first, including a dynamic

user credentials section based on $_SESSION variables and a base container layout for

content to be slotted into with a section menu fixed on the right-hand side of the page on

computer devices or on top for mobile devices. In the diagram shown below (Figure 3) is

an example page that shows the general layout of the front end when an end user logs into

ULPhone, a simple header navigation bar on the right followed by a container for all page

content to be displayed.

23

Figure 3.3: Standard web page layout.

3.4. Connection between front end and server.

Linking the Asterisk based PBX of ULPhone to the web based front end was probably the

most difficult aspect of this project. Most of the solutions I used were created completely

from trial and error, as resources for this specific of a solution are just not available.

However, this was a major milestone for the project as once I had completed this, ULPhone

was a functional service that allows users to register and make free calls. The underlying

backbone of the intercommunication between the web based front end and back end

Asterisk communications server was a proper implementation of MySQL databases. These

databases act as a medium between the front and back end, transferring necessary data and

communicating with each. ULPhone consists of two databases, these are, in more detail:

• front_end: The front_end database contains the user table. This table and database

deal solely with the web based front end. Every piece of user data entered on the

front-end registration page is stored in this table. The data is then passed to the

asterisk database using MySQL triggers to be passed to the Asterisk

communication server.

24

• asterisk: The asterisk database deals mainly with data that is given to the Asterisk

PBX. It receives the user data from the front_end database and consists of three

tables:

1. ast_sipfriends: This table stores all the user data that is dynamically passed

to the PBX for use. When a user’s credentials are added to the ast_sipfriends

table they are sent to the PBX’s sip.conf file for registration. This is done

once Asterisk Realtime is enabled which allows asterisk to populate its

configuration files with data from MySQL databases. The table also

contains a lot of NULL values which are populated once the user has

registered their softphone to the PBX using their username and password.

These values include the host of the specific softphone device and the IP

address of the device. This data allows for checking is the specific user is

online, meaning, if the user has NULL values for their registered device’s

IP address the user is not connected via softphone, so the online status can

be set to false and passed to the directory table and vice versa if the data is

not NULL the online status can be set to true. The ast_sipfriends also allows

for specification of the SIP users context (defined in section 2.3.7.).

2. extensions: This table takes any new registered user and assigns them an

extension to be used within the PBX. Once the extension is assigned it is

passed to the Asterisk PBX and associated with the newly registered SIP

user.

3. directory: The directory table takes specific user information from all

tables and is used to display the user directory on the ULPhone front end

customer homepage. This table shows if a specific is online and eligible to

be called within the PBX.

25

Figure 3.4: Asterisk database schema.

Figure 3.4 clearly shows the schema of the backend MySQL asterisk database, with its

relations highlighted via entity relationship indicators.

Once the back-end MySQL tables were configured correctly the next step was to set up the

ODBC driver for use with ULPhone. The ODBC driver uses the Open Database

Connectivity interface that allows applications to access data in database management

systems (DBMS) using SQL. The ODBC driver is set up by creating the res_odbc.conf file.

In this file, the credential for accessing the MySQL backend databases are specified, this

allows Asterisk to pull information from them once Asterisk Realtime Architecture

(Section 2.3.7) is enabled.

The last step of the intercommunication process was to edit the sip.conf and

extensions.conf file and enable Asterisk Dynamic Realtime. Once this was enabled

Asterisk gets all its SIP information from the specified database in the ODBC

configuration, in the case of ULPhone the asterisk table. Once this final step was complete

users were dynamically registered to the PBX from the front end and ULPhone and could

make calls to other dynamically registered users.

26

3.5. Connections to PSTN using go trunk

The final step of development was to connect the PBX to the Public Switched Telephone

Network (PSTN) and allow users to make calls to PSTN mobile numbers. This step was

accomplished by making use of a SIP trunk (Section 2.3.7.), adding the trunk to the sip.conf

file and assigning it an extension within the extensions.conf file. The SIP trunk acts as

gateway from the PBX to the PSTN. There are many SIP trunks available offering a wide

variety of plans for a diverse array of implantations but for the purposes of the development

of ULPhone I chose to go with the SIP trunk provider GoTrunk. My reasoning for choosing

GoTrunk included:

• Free trial: GoTrunk offers new users a month’s free trial with credit. This was

perfect for my implementation as I needed to test the trunk within my PBX

thoroughly before committing to purchasing a plan.

• Credit System: GoTrunk is a pay as you go service which means you only pay for

talk time you need. This means that for any breach attempts the potential hacker

can only use the credit that is left on the account and not exceed any monthly plan.

• Versatile control console: GoTrunk’s admin portal is very well designed. It

provides helpful analytics on calls made, a directory of international extensions

codes and call rates, very detailed call history log and 24-hour customer support

instant messaging for any problems or queries.

Once the specific account is set up, the supplied trunk needs to be added to the PBX’s

sip.conf and extensions.conf. In the case of ULPhone the sip trunk acts as a SIP user

essentially. Any call beginning with the number nine is sent to the SIP trunk defined in

sip.conf , and then the remaining numbered dialed is sent out through the trunk to be dialed

on the PSTN.

27

4. Security

Security is an important factor in all telephony related software that must be addressed.

Care and accountability must be taken in the development of such communications servers

as there is a lot of money to be made by attackers who aim to exploit a system to make free

calls. During the development of ULPhone, security was an aspect I initially neglected as

I was unaware of the security risks posed. As the development process iterated, it was clear

that the security of the PBX needed to be addressed as I experienced several different

breach attempts.

4.1. Issues and Complications

To understand why security is important in a PBX situation it is first important to

understand the benefits of attacking a PBX. The main things hackers want is to find a

weakness that will allow them to make a long distance, international or other chargeable

calls for free. In some cases, the malicious infiltrators sell the hijacked time from the server

to end users who think they are a part of a legitimate service when, in reality, they are using

minutes stolen from a hacked PBX.

During the development of ULPhone, the PBX experienced several different breach

attempts by unknown intruders. These attempts included:

• Breach 1: The most common way intruders gain access to a PBX system is by

listening to and intercepting traffic on the common SIP port 5060. Once the hacker

finds an acceptable target the PBX is brute forced attacked with user credentials to

try and gain access to the system. This happened with ULPhone, several thousand

brute force attacks were registered from unknown IP addresses trying to breach the

system. The brute force attack was indicated in the Asterisk log files as shown in

figure 5 in appendix 2.

• Breach 2: Before I could implement the solution to Issue 1, which is detailed in

section 5.2., the PBX was unfortunately breached for a short amount of time. When

the breach occurred, the hacker made a series of calls out through the SIP trunk to

many various Asian countries trying to steal all the allocated minutes on the

GoTrunk SIP trunk, fortunately, GoTrunk’s customer care team realized there was

28

an unusual amount of foreign calls made to Vietnam, roughly 500 (Figure 5

Appendix 2), so they shut the trunk down and contacted me immediately. The task

then was to secure the PBX correctly to prevent further attacks and have the SIP

trunk relaunched.

4.2. Solutions and Protection Methods

After the breach of the system I did a lot of research into the best way to secure a PBX in

my specific scenario. The AWS EC2 instance was the most secure aspect of ULPhone as

it is secured by Amazon with encrypted keys that are required to access the instance. The

PBX however needed to be more secure, for this reason I implemented two popular PBX

security methodologies.

1. The common SIP port was changed from 5060 to 5080. Once this step was

implemented the hacker was no longer able to find the PBX on the common SIP

port. To do this, I forwarded the port 5080 in the EC2 control console and changed

the default SIP port in the Asterisk sip.conf configuration file. Once the change was

made the end user had to change their softphone credentials which are used to

connect to the PBX. The host credential changed from 35.160.192.57 to

35.160.192.57:5080. This alleviated much of the security concerns

2. Fail2Ban was installed on the system. Fail2Ban is a package which was installed

on the communications server, it scans the log files of a specific system, in

ULPhone’s case, the Asterisk log files for repeated entries of an IP and bans IPs

that exhibit behavior which would indicate malicious intent. Entries which contain

too many login attempts or too many failed password entries are flagged by the

package and added to an IP table where the admin can specify the time of the ban,

or if the IP should be banned in the first place. The configuration of Fail2Ban is

done with two files, the jail.conf file, which specifies ban times, communication

ports and protocols and the fail2ban.conf file which specifies where the log file

information should be stored as well as the path to the socket which is to be used.

29

5. System Testing & Evaluation

5.1. General Introduction

The overall testing of ULPhone was a continuous process. The nature of the PBX

development involved constant addition of features, which meant after every piece of

functionality was added to the system I had to test that all the previous functionality was

still operational.

The php, html and CSS code used to develop the web based front end was tested for errors

using the PHPStorm IDE’s built in testing techniques. I was able to write PHPUnit test

cases to test the functionality of the code and to use the IDE’s inspection tools to check for

syntax errors.

5.2. Testing methods

Once the project was completed and met the initial success criteria, I felt user testing was

extremely important for ULPhone as it is a system designed for use by people with different

areas of technical proficiency. In other words, it should not be a daunting task for potential

end users to set up and make calls using service. I developed an instruction sheet and

questionnaire for user testing (shown in appendix 1). The questionnaire contains

instructions for registering an account to the web based front end, connecting to the PBX

with the registered softphone using the registered credentials and making calls to test

number on the PBX as well as making calls to the PSTN. I distributed the document to ten

peers from various academic backgrounds and recorded the results.

5.3. Findings

From the users who completed the test document:

• 80% could follow the instructions, connect to the PBX and make calls to both the

PSTN and numbers within the PBX.

• 100% found the web based front end helpful.

• 50% found the softphone set up intuitive and user-friendly.

Most of the feedback given was positive but some of the common constructive criticism

was that the softphone setup was confusing and it was hard to understand how to connect

30

it the server. The feedback from the 20% of users who could not complete the instruction

sheet was that they didn’t really understand the concept of the PBX system and for one

user the fact that they did not possess a capable smartphone meant they could not physically

complete the test.

5.4. Implementation of user feedback

From the tests, I could target some key areas which needed to be addressed to improve the

overall functionality of ULPhone. These key areas plus my solutions to the issues include:

1. The way in which users connect their softphone to the PBX needs to be improved.

My initial concept to tackle this issue was to build my own custom android based

softphone which, once installed, was automatically connected to the ULPhone PBX

when launched. However, after some discussion with my supervisor we concluded

that was beyond the scope of the project as an FYP. Instead I opted to make the set-

up section of the web based front end more understandable to all users.

2. The web based front end needed to contain more concise information on the

ULPhone PBX system. Similar the first issue I fixed, this problem by providing

more detailed information on the front end which is easily read and comprehensible

to all ULPhone users.

31

6. Summary & Conclusions

6.1. Conclusions

In conclusion, it is fair to say that Asterisk is an extremely versatile and robust piece of

software. Not only is Asterisk extremely beneficial for small to enterprise systems but it is

also an option for personal projects that wish to reap the benefits of a corporate PBX but

lack the capital to produce one. Not only is Asterisk exceptionally cost effective when

compared to most other PBX options, it also boasts many more features and functions than

competitors.

I believe cloud based PBX’s are the future and it is clear why it is expected that the hosted

PBX market will reach a net worth of $18 Billion by 2018 (festeban, 2015). The cost

effectiveness on reduction on maintenance fees are reason enough for any PBX using

corporation to switch to a cloud- based solution.

My Final Year Project, ULPhone enabled me to set up a functional cloud based PBX with

an interconnected web based front end used to interact with the Asterisk based PBX. This

factor alone I feel shows the versatility of the framework. I also so feel that because of the

fact the initial success criteria had been met, my final year project was in fact, a success.

6.2. Recommendations

If I were to continue the development of ULPhone there are several factors of development

I would address:

• SoftPhone: I would develop a ULPhone specific softphone, my reasoning for this

is that Zoiper, as helpful as it was throughout the project, is prone to bugs and errors.

My goal would be to create a very simple UI based softphone that connects directly

to the ULPhone PBX and only prompts the user for their username and password

and does not ask them for a host name with a specific SIP port, a process which

proved confusing for some of the user tests.

• Directory Improvement: I would also enhance the directory which is located on the

user dashboard. I would divide each user into their specific education faculty within

32

the university and give them access to all the faculties’ specific information,

lecturer emails, lecturer office hours etc.

• Docker: Once all the development improvements have been completed, I would

consider possible Docker integration for the whole system. I would like to see

ULPhone as an all in one service which would include the softphone, the Asterisk

based PBX and the web based front end. With the possible Docker Container file

integration, the all in one service could be distributed to system administrators and

the whole PBX, front end and softphone service could be rolled out by simply

running a few command line prompts and customizing specific configuration files

to match their companies need.

33

7. References

1. Amazon Web Services, Inc. (2015). AWS Free Tier. [online]

Available at: https://aws.amazon.com/free/

[Accessed 1 Dec. 2016].

2. Asterisk.org. (2017). Get Started. [online]

Available at: http://www.asterisk.org/get-started

[Accessed 2 Dec. 2016].

3. Digium, (2015). Build Communications Applications, Solutions and Products

with Asterisk. [online] Digium,com.

Available at: https://www.digium.com/sites/digium/files/who-is-digium-

asterisk.pdf

[Accessed 10 Nov. 2016].

4. festeban, (2015). Market Size for Hosted PBX Growing >20% and Reaching $12

Billion in 2018. [online] SkySwitch.

Available at: http://www.skyswitch.com/market-size-hosted-pbx-growing-20-

reaching-12-billion-2018/

[Accessed 2 Jan. 2016].

5. Microsoft, (2016). Firewalls. [online] Technet.microsoft.com.

Available at: https://technet.microsoft.com/en-us/library/cc700820.aspx

[Accessed 15 Dec. 2016].

6. Samgoma Technologies, (2010). Top 6 Reasons to Choose freePBX. [online]

freepbx.org.

Available at: https://www.freepbx.org/wp-content/uploads/2016/08/Infographic-

FreePBX-Phones-web.pdf

[Accessed 15 Nov. 2016].

7. SearchNetworking. (2017). What is PSTN (public switched telephone network)? -

Definition from WhatIs.com. [online]

Available at: http://searchnetworking.techtarget.com/definition/PSTN

[Accessed 2 Jan. 2017].

8. VOIP / Asterisk – Alfian’s Linux (2017) What is LIBPRI [online],

Available at Alvienuxs.wordpress.com.

[Accessed 2 Jan. 2017].

34

9. Amazon EC2 Key Pairs - Amazon Elastic Compute Cloud (2017) Key Pairs

[online],

Available at http://docs.aws.amazon.com/AWSEC2/latest/UserGuide/ec2-key-

pairs.html

[Accessed Nov. 2016].

10. Jordan, M. (2016) Dialplan - Asterisk Project - Asterisk Project Wiki [online],

Wiki.asterisk.org,

Available at: https://wiki.asterisk.org/wiki/display/AST/Dialplan

[accessed 17 Jan 2017].

11. Wintermeyer, S. (2010) 2.1. Context [online], The-asterisk-book.com,

Available at: http://the-asterisk-book.com/1.6/der-context.html

[accessed 12 Nov 2016].

12. Rouse, M. (2005) What Is POSIX (Portable Operating System Interface)? -

Definition from Whatis.Com [online], SearchEnterpriseLinux,

Available at: http://searchenterpriselinux.techtarget.com/definition/POSIX

[accessed 28 Nov 2016].

13. Stallings, W. (2016) The Session Initiation Protocol - The Internet Protocol

Journal - Volume 6, Number 1 [online], Cisco,

Available at: http://www.cisco.com/c/en/us/about/press/internet-protocol-

journal/back-issues/table-contents-23/sip.html

[accessed 9 Dec 2016].

35

8. Bibliography

1. Meggelen, J., Madsen, L. and Smith, J. (2007). Asterisk. 1st ed. Sebastopol,

Calif.: O'Reilly.

2. Wintermeyer, S., Bosch, S. (2010) Practical Asterisk 1.4 And 1.6, 1st ed,

Addison-Wesley: Upper Saddle River, NJ.

3. Lewis. T.R. (2016) Building an Asterisk GUI in C# Proven Method.1st ed.

Create Space.

4. Gotrunk: Ultimate SIP Trunking Solution [online] (2017) [online], GoTrunk.

5. Amazon Web Services (AWS) - Cloud Computing Services [online] (2017)

[online], Amazon Web Services, Inc.

6. Meloni, J. (2012) Sams Teach Yourself PHP, Mysql And Apache All In One, 1st

ed, Sams Pub.: Indianapolis, Ind.

7. Business Phone Systems | Unified Communications | Digium [online] (2017)

[online], Digium.com.

36

9. Appendices

9.1 Appendix 1: User testing sheet

ULPHONE USER TESTING INSTRUCTIONS

Step 1. Download the Zoiper softphone application to your smartphone via the app store or google play

store.

Step 2. Visit http://35.160.192.57/ and fill out the registration form with your own unique credentials.

Upon successful registration, a confirmation email will be sent to the email address you registered.

Step 3. Log into ULPhone using the username and password entered during registration and make note of

your extension in the PBX which is located at the top of the page in the ‘Your Details’ section.

Step 4. Read carefully the ‘Set Up’ section which details how to connect your Zoiper softphone to the

PBX.

Step 5. Connect your softphone to the PBX using the instructions.

Step 6. Once connected successfully dial the test extension 6001 from the Zoiper dial pad. You can also

dial your own extension which you made note of to test you can receive calls.

Step 7. Dial a number on the PSTN by entering 9 + the number you want to dial. For example, an Irish

mobile would be reached by dialing 9353851234567

Questionnaire (Please circle the answers which applies)

Q1. Did you find the web based front end helpful?

YES NO

Q2. Did you find the softphone set up intuitive and user friendly?

YES NO

Q3. Were you successfully complete step 6 and 7?

YES NO

Please provide any feedback of your experience with ULPhone:

______________________________________________________________________________

______________________________________________________________________________

______________________________________________________________________________

37

9.2 Appendix 2: Screenshots & Images

Figure 1: ast_sipfriends table.

Figure 2: Notification of successful registration in Asterisk CLI.

Figure 3: Information displayed when call is made.

38

Figure 4: Brute force attack.

Figure 5: SIP trunk breach.

39

Figure 6: EC2 free tier.

Figure 7: EC2 control console.

40

Figure 8: Front-end home page.

Figure 9: Home page on mobile 1. Figure 10: Home page on mobile 2.

41

Figure 11: User dashboard with directory and user credentials.

Figure 12: Set Up instructions page.

42

Figure 13: Initial breakdown of work.

Figure 14: GoTrunk admin portal.

43

Figure 15: Asterisk make menuselect.

Figure 16: Successful Asterisk install notification.