Project Final2 H&F new

8/7/2019 Project Final2 H&F new

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Design and Develop VOIP Dialer and Recording Database

CHAPTER 1

1. INTRODUCTION

1.1 What is Voice Over Internet Protocol?

Voice over Internet Protocol is a general term for a family of transmission

technologies for delivery of voice communications over internet protocol networks such

as the internet or other packet-switched networks. Other terms frequently encountered

and synonymous with voice over internet protocol are internet protocol telephony,

internet telephony, voice over broadband, broadband telephony, and broadband phone.

internet telephony refers to communications services voice, facsimile, and/or voice-

messaging applications that are transported via the internet, rather than the public

switched telephone network.

Fig 1.1:- Alternative voice over internet protocol Architectures

The basic steps involved in originating an Internet telephone call are conversion of the

analog voice signal to digital format and compression/translation of the signal into

internet protocol packets for transmission over the internet; the process is reversed at the

receiving end. Voice over internet protocol systems employ session control protocols to

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control the set-up and tear-down of calls as well as audio codecs which encode speech

allowing transmission over an internet protocol network as digital audio via an audio

stream. Codec use is varied between different implementations of voice over internet

protocol (and often a range of codecs are used); some implementations rely on

narrowband and compressed speech, while others support high fidelity stereo codecs.

Voice over Internet Protocol is a technology for communicating using Internet

protocol instead of traditional analog systems. Some voice over internet protocol

services need only a regular phone connection, while others allow you to make

telephone calls using an Internet connection instead. Some voice over internet protocol

services may allow you only to call other people using the same service, but others may

allow you to call any telephone number - including local, long distance, wireless, and

international numbers. Voice over internet protocol is mainly concerned with the

realization of telephone service over internet protocol-based networks such as the

internet and intranet. Internet protocol telephony is currently breaking through to

become one of the most important service on the net. The actual breakthrough was made

possible by the high bandwidth available in an intranet and, increasingly, on the internet.

Another fundamental reason is the cost associated with the various implementations.

1.2 Phone to Phone via the Internet

Fig 1.2:- Phone to phone via internet

The public telephone network and the equipment makes it possible are taken for

granted in most parts of the world. Availability of a telephone and access to low-cost,

high quality worldwide network is considered to be essential in modern society

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(telephone are even expected to work when the power off).There is, however, a

paradigm shift beginning to occur since more and more communication is in digital

form and transported via packet networks such as internet protocol and Frame Relay

frames. Since data traffic, there has been considerable interest in transporting voice over

data networks. Support for voice communications using the internet protocol, which is

usually just called Voice over internet protocol or voice over internet protocol, has

become especially attractive given the low-cost, flat-rate pricing of the public Internet.

In fact, toll quality telephony over internet protocol has now become one of the key

steps leading to the convergence of the voice, video, and data communications

industries. The feasibility of carrying voice and signaling message over the internet has

already been demonstrated but delivering high-quality commercial products,

establishing public services, and convincing users to buy into the vision are just

beginning.

1.3 Phone to Internet to Gateway to PSTN

Fig 1.3:- Phone to internet to gateway to PSTN

1.4 Definition

Voice over internet protocol can be defined as the ability to make telephone calls

and to send facsimiles over internet protocol- based data networks with a suitable

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quality of service and a much superior cost/benefit. Equipment producers see Voice

over internet protocol as a new opportunity to innovate and copete. The challenge for

then is turning this vision into reality by quickly developing new voice over internet

protocol-enabled equipment. For Internet service providers, the possibility of

introducing usage-based pricing and increasing their traffic volumes is very attractive.

Users are seeking new types of integrated voice/data applications as well as cost

benefits. Successfully delivering voice over packet networks presents a tremendous

opportunity; however, implementing the products is not as straightforward a task as it

may first appear. This document examines the technologies, infrastructures, software,

and systems that will be necessary to realize voice over internet protocol on a large

scale. The types of applications that will both drive the market and benefit the most

from the convergence of voice and data networks will be identified.

1.5 History of Voice Over Internet Protocol

Voice over Internet Protocol owes its existence to the difference in price

between long-distance connections and the use of data networks. This technology uses

data networks such as the Internet to transmit voice information from a simple PC. A

telephone conversation is conducted via microphone and loudspeaker connected to the

sound card. Microsoft NetMeeting is the most common Internet telephony program. Its

feaures also include Internet video communication (image telephony). Or, a specially

adapter can be used to hook standard telephones up to the data network. All devices that

support the same standard can be connected over one data network. Gateways are also

available for connecting these devices to

telephones in the normal telephone network. These possibilities have led to the creation

of IP-based telephone systems using voice over internet protocol. The development of

voice over internet protocol technology is summarized and predicted in the following:

1995=> The year in which to PCs are connected using PC software

1996=> The year of the IP telephony client.

1997=> The year of the Gateway.

1998=> The year of the Gatekeeper.

1999=> The year of the Application.

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CHAPTER 2

2. Voice Over Internet Protocol Components

The components of VoIP include: end-user equipment, network components,call processors, gateways and protocols.

2.1End-user equipment

It is used to access the VoIP system to communicate with another end point.

Connection to the network may be physically cabled or may be wireless. The end-user

equipment may be a phone that sits on a desk or a softphone that is installed on a

PC.Functions include voice and possibly video communication, and may contain instant

messaging, monitoring and surveillance capabilities. 7 Though end-user equipment is

often deployed on an internal, protected network, it is usually is not individually

protected by other devices (firewalls) and may be threatened if the equipment has

vulnerabilities. The threat, of course, is also dependent on the level of security that

exists on the internal network. If the device is allowed to reach or can be reached from a

public or unprotected network, there may be threats that are not normally found on the

internal network. Softphone software may have vulnerabilities, there may be

vulnerabilities in the operating system it is running on, and there may be vulnerabilities

of other applications running on the operating system. Patching operating system, soft

phone software and those other applications can help mitigate the risk of any threats that

are present. Additionally, some end-user equipment may have firmware upgrades that

can be applied or may be able to obtain updated software during registration. For

operating system based Voice over internet protocol solutions, consideration should be

given to virus detection and host based firewalls as well as host-based intrusion

detection. Centralization of management of these security components is best, allowingthe users of the solution to focus on their duties instead of security details, increasing

productivity.

2.2 Network components

It includes cabling, routers, switches and firewalls. Usually the existing IP

network is where a new Voice over internet protocol system is installed. The impact on

the internet protocol network is greater than merely adding more traffic. The added

traffic has more of an urgency to reach its destination than most of the data traffic that is

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already supported. Switches, routers and firewalls will need to recognize and act on

Voice over internet protocol data in order to keep latency down. Additional security

measures, addressed later, will complicate this process.

Performance can be gained by separating the data traffic from the voice over

internet protocol traffic by putting them on different virtual local area networks. This

allows management of the data to be segregated so it can be handled based on data type.

Since the voice over internet protocol data must have a higher level, isolation of the data

types via virtual local area network can help increase the performance at the cost of that

on other virtual local area network. This cost may be very low to the other applications.

Although virtual local area network should not be relied on alone, they will add a layer

of security. The ability to listen to, or sniff, the network, potentially allows the hacker tomonitor calls and manipulate the voice over internet protocol system. It is generally

more difficult for a hacker to sniff or interfere with the voice traffic from the data virtual

local area network when the voice traffic is on its own virtual local area network, but it

can be done by manipulating the routing of the network. Encryption can also help

defend against sniffing. Another internet protocol network concern is network

slowdowns that might increase latency, jitter or packet loss. Slowdowns can be caused

for many reasons including configuration issues, denial of service attacks or high

bandwidth utilization by other systems on the network. Configuration issues are

probably best addressed with education and checking mechanisms, such as having a co-

worker verify configurations. Denial of service attacks are difficult to defend against,

but may be reduced by filtering the traffic that can communicate on the network to be

only that which is allowed. This may prove difficult due to the use of random ports by

voice over internet protocol. Regular network bandwidth analysis can help with tuning

of a network and helps with capacity planning. Being aware of bandwidth growth trends

helps network administrators know when bandwidth needs to be addressed.

Voice over internet protocol suffers from most of the same internet protocol

network vulnerabilities as other systems. A well secured internal network is the first step

to protecting the voice over internet protocol system as it was for the pre-existing

internet protocol network. Care must be taken to ensure security solutions keep latencies

low or the security solution itself may prove to be a denial of service.

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2.3 Call processor

These functions can include phone number to internet protocol translation, call

setup, call monitoring, user authorization, signal coordination, and may help control

bandwidth. 6 Call processors are usually software that runs on a popular OS. This leaves

it open to network attacks for the vulnerabilities of the given OS, the vulnerabilities of

the application and other applications running on the operating system.

2.4 Gateways

It can be categorized into three functional types: Signaling Gateways, Media

Gateways and Media Controllers. In general, they handle call origination and detection

and analog to digital conversion. Signaling gateways manage the signal traffic between

an internet protocol network and a switched circuit network, while media gateways

manage media signals between the two. Media Gateway Controllers manage traffic. The

most common gateway protocols are megaco. Both are composites or derivations of

previously but now less used protocols.6 Vulnerabilities can exist between the internal

internet protocol network and the gated, circuit switched network. Care should be

taken to ensure any vulnerabilities are mitigated.

Gateway communication should be secured with internet protocol Sec to prevent

interference with calls and to prevent unauthorized calls from being setup. The gateway

itself is vulnerable to internet protocol based attacks and can be mitigated by using

internet protocol Sec and by removing any unnecessary services and open ports, as

should be done with any server.

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CHAPTER 3

3.working

Voice over internet protocol converts the voice signal from your telephone into adigital signal that can travel over the internet. If you are calling a regular telephone

number, the signal is then converted back at the other end. Depending on the type of

voice over internet protocol service, you can make a voice over internet protocol call

from a computer, a special voice over internet protocol phone, or a traditional phone

with or without an adapter. In addition, new wireless "hot spots" in public locations such

as airports, parks, and cafes allow you to connect to the Internet, and may enable you to

use Voice over internet protocol service wirelessly. If your Voice over internet protocol

service provider assigns you a regular telephone number, then you can receive calls

from regular telephones that dont need special equipment, and most likely youll be

able to dial just as you always have.

Fig 3.1:-voice over internet protocol work service

The exploratory nature of this study produced focus groups as an appropriate

method for data collection. Our overarching goal was to improve our understanding of

how Latino voice over internet protocol users employ the technology and why they

select certain voice over internet protocol services and providers. In addition, we wanted

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to learn about Latinos not connected to the Internet-what they know about voice over

internet protocol and why they are not online. Moreover, we sought to learn whether the

lower cost of telephone calls associated with voice over internet protocol are enough of

an incentive for non-Internet users to get online, and, if so, under what conditions. Four

focus groups of 9 to 12 participants were held in Los Angeles in August 2008 (total

sample size, N = 43). Two of the focus groups consisted of Latinos who are Internet

users and have either heard of or used some form of voice over internet protocol

technology and service. The other two groups consisted of Latinos who reported that

they do not use the Internet.

The study participants were residents of Glendale, Cudahy, Huntington Park,

and South Gate, cities that are part of Los Angeles County, a large metropolitan areawith a significant and diverse Latino population. Glendale is the third largest city in Los

Angeles County and it is the most ethnically diverse area of the four in this study.

Twenty percent (20%) of the population is Latino, 21% is Armenian, 35% is White

(non-Armenian, non-Hispanic), and 16% is Asian from different countries of origin.

Approximately 40% of the residents are homeowners. The median household income is

$41,800 (U.S. Census, 2000). In Glendale, 70% of Latinos are connected to the Internet.

This is one of the highest connectedness rates across Latino communities in Los

Angeles County (Wilkin et al., 2007). The contiguous cities of Huntington Park, South

Gate, and Cudahy are in Southeast Los Angeles. Over 90% of the population is Latino,

and most residents are of Mexican origin. The median household income is about

$32,000, and only 24% of the population is connected to the Internet.

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CHAPTER 4

4. PROTOCOL

There are several protocols used for voice over internet protocol but two aremost common. They are H.323 and Session Initiation Protocol.

Fig 4.1:- Protocol Layers

4.1 H.323

H.323 is a protocol suite specified by the International Telecommunications

Union that lays a foundation for internet protocol based real-time communications

including audio, video and data.8 H.323 allows for different configurations of audio,

video and data. Possible configurations include audio only, audio & video, audio & data

and, audio, data and video. H.323 does not specify the packet network or transport

protocols. This standard specifies four kinds of components: Terminals, Gateways,

Gatekeepers and Multi-point Control Units .Terminals are the end-user equipment

discussed above. Gateways handle communication between unlike networks with

protocol translation and media format conversion. Gatekeepers provide services such as

addressing, authorization and authentication, accounting functions and call routing.

Multi-point Control Units handle conferencing.

The International Telecommunications Union defines the H.323 zone that consists of

terminals, gateways, Multi-point Control Units, and a gatekeeper. The gatekeeper

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manages the zone. H.323 uses different protocols to manage different needs. There are

audio codecs and video codecs that encode and decode the audio and video data. H.225

covers registrations,

Fig 4.2:- H.323 Architecture

admissions & status and call signaling. Realtime Transport Control Protocol handles

various functions between the endpoints and the gateway, including registrations and

admission control as its name implies. It also manages changes in bandwidth and

disengage procedures. A Realtime Transport Control Protocol channel is opened, prior to

opening other channels, between the gateway and endpoint whereby Realtime Transport

Control Protocol messages are passed. Call signaling channels are opened between

endpoints and between an endpoint and a gatekeeper. They are used to set up

connections. Call setup and termination uses Q.931.9 H.245 is for channel negotiations

such as flow controls and general commands and H.235 specifies security. Real-time

Transport Protocol is used to transport data, typically via user datagram protocol and

provides a timestamp, sequence number, data type and ability to monitor delivery.

Realtime Transport Control Protocol is used mainly to monitor quality and manage

synchronization. As mentioned above, the H.235 protocols of H.323 are for security

profiles. These standards address authentication, integrity, privacy, and non-repudiation

10 and are expressed as Annexes to H.23 5 Version 2. They are Annexes D, E & F as

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more security whereas, user datagram protocol allows for faster, lower latency,

connections. Usual components of an Session Internet Protocol system are the user

agent, proxy server, registrar server, and the redirect server. The usual components

software contains client and server components. The client piece makes outgoing calls

and the server is responsible for receiving incoming calls. The proxy server forwards

traffic, the registrar server authenticates requests, and the redirect server resolves

information for the usual components client. The endpoints begin by connecting with a

proxy and/or redirect server which resolves the destination number into an internet

protocol address. It then returns that information to the originating endpoint which is

responsible for transmitting the message directly to the destination. A security

advantage of session internet protocol is that it uses one port. The main concerns for

security of are confidentiality, message integrity, no repudiation, authentication and

privacy. New security mechanisms were not created for session internet protocol

instead, session internet protocol uses those provided by Hyper Text Transfer Protocol

and Simple Mail Transfer Protocol as well as Internet Protocol Security.

Fig 4.3:- Self-Provided Customer Architecture

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Signal confidentiality is best provided with full encryption, however, since some session

internet protocol message fields must be read and/or modified by some proxies, care

must be taken and possibly other methods used. If however, the proxy can be trusted,

then encryption at the transport and/or network layers may be the best solution. Security

at the transport and networking layers accomplishes full packet encryption using

internet protocol sec. TLS had been used, but has been deprecated. Full encryption

requires support of the encryption method at each end point where it is implemented.

Hyper text transfer protocol authentication uses the 401 and 407 response codes and

header fields. This provides a stateless challenge-base mechanism for authentication

whereby the challenge and user credentials are passed in the headers. When a proxy or

usual components receives a request, it may challenge to ensure the identity of thesender. Once identity has been confirmed the receiver should also verify that the

requester is authorized. Details of this digest method may be found in RFC 326112.

Secure/Multipurpose Internet Mail Extension is an enhancement to Multipurpose

Internet Mail Extension that replaces Pretty Good Privacy. Since Multipurpose Internet

Mail Extension bodies are carried by session internet protocol, session internet protocol

may use to enhance security, Multipurpose Internet Mail Extension contains

components that can provide integrity and encryption for Multipurpose Internet Mail

Extension data and as RFC 2633 states Multipurpose Internet Mail Extension can be

used for authentication, message integrity and non-repudiation of origin (using digital

signatures) and privacy and data security (using encryption). Multipurpose Internet Mail

Extension is useful when full encryption of the packet is not feasible due to the need of

network components to use data from the header fields. User identification is done via

certificate belonging to the user that is compared to the header information. Integrity of

the message is verified by matching the information in the outside header with that of

the inside header. Normally, Multipurpose Internet Mail Extension is used to encrypt

Session Description Protocol but there may be requirements to encrypt certain header

components. Session internet protocol can provide header privacy by encapsulating the

entire message using Multipurpose Internet Mail Extension type message/sip. If used for

anonymity the message will need to be decrypted before the certificate can be identified

and consequently validated. Session internet protocol Security Concerns hyper text

transfer protocol digest does not provide the best integrity. Without Multipurpose

Internet Mail Extension, spoofing of the header would not be difficult. Multipurpose

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Internet Mail Extension requires a public key infrastructure. Since certificates are

associated with users, moving from one device to another may be difficult. With

Multipurpose Internet Mail Extension there may be issues with firewalls or other proxy

devices that may require viewing and/or changing session internet protocol bodies. There

is information in session internet protocol headers that may be considered sensitive, i.e.

an unlisted phone number. Consideration may need to be given to providing per-user

options that allow protection of this information. Session internet protocol and H.323

both use protocols that may use random ports requiring that the firewall be able to open

and close ports as required. An H.323 or session internet protocol aware firewall may be

required. As with H.323, network address translation presents problems for session

internet protocol.

4.3 Network Address Translation

Network Address Translation allows one network address to be translated at a

gateway between two networks into another address so that the packet will have a valid

source address on the network it is on. Most commonly Network Address Translation is

used to change private internet protocol addresses into public, Internet routable, internet

protocol addresses. Ports may also be translated. Network Address Translation traversal

is usually only a concern if end-user devices connect directly with an external network

or if they connect to the internal network from an external network.

Fig 4.4:- Network Address Translation Architecture

Network Address Translation is a layer of security because it hides the real

addresses on the internal network from the public network. Network Address

Translation can however, be a problem, because the routing device does not know the

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actual internet protocol address of the device. The information defining the endpoint is

in the header. The routing device must be able to read the header and in some cases (i.e.

with proxy firewalls) change it. This is hampered when encryption is used. The best

solution is to not use Network Address Translation if at all possible. By removing the

issue, the problem disappears, though another problem may present itself. When

Network Address Translation is required, care must be taken to select application and

proxy firewalls that handle the implementation or, alternatively, consider a service offered

by the public networks.

4.4 Denial of Service

Denial of Service is caused by anything that prevents the service from being

delivered. A Denial of Service can be the result of unavailable bandwidth or voice over

internet protocol components being unavailable. Many things can cause a Denial of

Service including: a network getting congested to a level that it cannot provide the

bandwidth needed to support the application; servers not capable of handling the traffic;

extraneous services may be running that reduce the available resources to the server;

malicious programs such as viruses and Trojan horses; other malicious programs with

the purpose of causing Denial of Service or hacking activity.

Fig 4.5:- PSTN Architecture

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If Denial of Service is caused by bandwidth constraints, potential solutions are

increasing the bandwidth and/or isolating the voice over internet protocol traffic so that

it gets service first. Various methods of ensuring servers dont stop working, such as

failover methods like clustering, can help reduce Denial of Service from failing

components. Each component of the voice over internet protocol system offered by the

vendor, should be evaluated, removing those that are unnecessary. Server size should be

planned such that all desired vendor services and expected traffic can be supported,

adding some percentage for expected growth.

Defending against malicious programs and activity is more difficult but should begin

with applying appropriate patches in a timely manner, and installing virus protection

with frequent updates. In addition, installation designers should consider a host basedfirewall, intrusion detection and/or intrusion prevention. Defense against Denial of

Service attacks of public servers can best be done by locating the device with the public

available internet protocol addresses behind a firewall or other device that only allows

communication from trusted sources. Also, harden the operating systems in use,

removing all unnecessary services and applications from the servers and workstations,

patching, etc.

4.5 Other Concerns

Additional concerns of a VoIP system that need to be considered are databases,

web servers, additional VoIP services offered by the vendor, protocol stacks, access to

public or unknown networks, physical security and electrical power. Databases are

needed at some point of the VoIP implementation to store and retrieve information as

needed to accommodate various functions of the system. Database security principles

should be applied including changing the default administrator password, patches as

they become available, and best practices concerning access to the database, especially

from sources other than the voice over internet protocol system. A common feature of

end-user equipment is a web browser, the purpose of which is to provide additional

functionality and increased productivity. A voice over internet protocol system server

may have a web browser interface allowing management. If supported, patch the device

when the patch becomes available and use as strong authentication as can be supported.

Each vendor, having their own implementation of voice over internet protocol system,

may require any number of services to run on a server to support their product. As

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mentioned before, keep patches up to date and turn off all unneeded services. If the risk

is great enough, consider encryption and/or protection by another device such as a

firewall. The voice application and the operating system have similar vulnerabilities and

should be patched as well. If the voice over internet protocol system stays within a

secured network and only connects to the public network through a gateway, the

gateway is a vulnerability that needs addressing. Deploy the hardened gateway behind

an appropriate firewall, i.e. one that is aware of the protocols used. Voice over internet

protocol system must process the protocols that it supports so it needs to have some

implementation of a network stack. Stack implementations are written by the vendor

purchased from another vendor. With the latter, all vendors that purchased a specific

vendors stack will share the same vulnerabilities. Patch if necessary, when patches

become available. Ensure that the components are physically secure. Access to the box

allows ownership. There are many methods of compromising a device, depending on the

device and the underlying operating system, with physical access. Good security

practices include removing the a-disk and the CD- ROM from the boot list and

password protect the configuration. If a component is unavailable, then there is a denial

or service.

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CHAPTER 5

5. APPLICATIONS AND BENEFITS

Voice communication will certainly remain a basic from of interaction for all ofus. The public switched telephone network mply cannot be replaced, or even

dramatically changed, in the short term (this may not apply to provide voice networks,

however). The immediate goal for voice over internet protocol service providers is to

reproduce existing telephone capabilities at a significantly lower total cost of operation

and to offer a technically competitive alternative to the public switched telephone

network.

Fig 5.1:- Voice over internet protocol infrastructure

It is the combination of voice over internet protocol with point-of-service

applications that shows great promise for the longer term. The first measure of success for

voice over internet protocol will be cost saving for long distance calls as long as there

are no additional constraints imposed on the end user. For example, callers should not

be required to use a microphone on a pc. voice over internet protocol provides a

competitive threat to the providers of traditional telephone service that, at the very least,

will stimulate improvements in cost and function throughout the industry implemented

using an internet protocol network. This design would also apply if other types of packet

networks (such as frame relay) were being used.

Some example of voice over internet protocol applications that are likely to be useful

would be:

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5.1 Public switched telephone network gateways

Interconnection of the Internet to the public switched telephone network can be

accomplished using a gateway, either integrated into or provided as a separate device. A

PC-based telephone, for example, would have access to the public network by calling a

gateway at a point close to the destination (thereby minimizing long distance charges).

5.2 Internet-aware telephones

The goal for developers is relatively simple: add telephone calling capabilities

( both voice transfer and signaling) to internet protocol-based networks and interconnect

these to the public telephone network and to private voice networks in such as way as to

maintain current voice quality standards and preserve the features everyone expects

from the teleph Fig illustrates an overall

Fig 5.2:- overall architecture for VoIP an product developer arise

Architecture for voice over internet protocol an Suggests that the challenges forthe product developer arise in five specific areas:

1. Voice quality should be comparable to what is available using the public switched

telephone network, even over networks having variable levels of operating system.

2. The underlying internet protocol network must meet strict performance criteria

including minimizing call refusals, network latency, packet loss and disconnects. This is

required even during congestion condition or when multiple users must share network

resources.

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3. Call control (signaling) must make the telephone calling process transparent so that

the callers need not know what technology is actually implementing the service.

4. public switched telephone network service interworking (and equipment

interoperability) involves gateways between the voice and data network environments.

5. System management, security, addressing (directories, dial plans) and accounting

must be provided, preferably consolidated with the public switched telephone network

operation support systems.

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CHAPTER 6

6. Comparison of VoIP software

VoIP software is used to conduct telephone-like voice conversations acrossInternet Protocol (IP) based networks. VoIP stands for "Voice over IP". For residential

markets, VoIP phone service is often cheaper than traditional public switched telephone

network (PSTN) service and can remove geographic restrictions to telephone numbers,

e.g. have a New York PSTN phone number in Tokyo.

For businesses, VoIP obviates separate voice and data pipelines, channeling both types

of traffic through the IP network while giving the telephony user a range of advanced

capabilities.

Softphones are client devices for making and receiving voice and video calls over the IP

network with the standard functionality of most "original" telephones and usually allow

integration with IP phones and USB phones instead of utilizing a computer's

microphone and speakers (or headset). Most softphone clients run on the open Session

Initiation Protocol (SIP) supporting various codecs. Skype runs on a closed proprietary

network, though the network (but not the official Skype client software) also supports

SIP clients. Online "Chat" programs now also incorporate voice and video

communications.

Other VoIP software applications include conferencing servers, intercom systems,

virtual FXOs and adapted telephony software which concurrently support VoIP and

PSTN like IVR systems, dial in dictation, on hold and call recording servers.

6.1 General softphone clients

Program Operating systems License Open Protocols/based Encryption Max Other capabilities Latest

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Source

?

upon/compatibl

e with

conferenc

e peersrelease

AOL Instant

Messenger

Linux, Mac OS,

Windows

Freeware

/ Closed

Proprietar

y

NoSIP (Windows

ver. only), RTP

Unknown Unknown

Video, file transfer,

PC to phone, phone

to PC

BlinkLinux, Mac OS,

Windows

GPL /

Free

software

Yes

ICE, SIP,

MSRP, RFB

(VNC)

sRTP, TLS Unlimited

IM, File Transfer,

Desktop Sharing,

Multi-party

conference,

Wideband

0.23.2

(February

15, 2011)

Brosix

Linux, Mac OS,

Windows

Freeware

/ Closed

Proprietar

y

No Yes Unknown

Text chat, File

transfer, Video chat,

Screen-shot, Screen-

sharing,

Whiteboard, Co-

browse

3.0 (July

2010)

Cisco IP

Communicat

or

Windows

Closed

Proprietar

y

NoSCCP (Skinny),

SIP, TFTPsRTP Unknown

7.0.3 (Aug

2009)

Ekiga

Linux, (Beta

Windows support),

OpenSolaris

GPL /

Free

software

Yes

SIP, H.323,

H.263,

H.264/MPEG-4

AVC, STUN,

Theora,

Zeroconf

No Unknown

Video, IM, LDAP,

Call Forwarding,

Call Transfer

3.2.7 (May

31, 2010)

Empathy Linux

GPL /

Free

software

Yes

SIP, XMPP

(Jingle), ICE

(STUN/TURN),

Zeroconf

No Unknown

IM, multi-user A/V,

collaborative

applications

2.32.0.1

(2010-10-

04)

Eyeball Chat Windows

Freeware

/ Closed

Proprietar

y

NoSIP, STUN,

ICE, XMPPYes Unknown

IM, Conferencing,

Voice, Video and

SIMPLE based

presence

Windows

3.2

Gizmo5

Linux, Mac OS X,

Windows, Windows

Mobile Phone,

Blackberry, Nokia,

PDA Java

Freeware

/ Closed

Proprietar

y

No SIP, XMPP SRTP Unknown

Record Calls,

Forward Calls,

MSN IM, Windows

Live Talk, Google

Talk, Talk with

Yahoo, Messenger,

XMPP

Windows:

4.0.5.395

(23 Sep

2009), Mac

OS:

4.0.0.269

(23 Sep

2009)

Google Talk Windows Closed No XMPP zRTP Unknown Video, chat, file 1.0.0.104

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Proprietar

y (using

libjingle)

transfer, voicemail,

mail via "GMail

Integration"

iChat Mac OS X

Closed

Proprietar

y

No

SIP AIM ICQ

XMPP H263

H264

Unknown Unknown

Integrated, PBX

independent

January

2007

Jitsi

Linux, Mac OS,

Windows XP/2000

(all java supported)

LGPL /

Free

software

YesSIP/SIMPLE,

XMPP

Voice

encryption

(SRTP and

negotiation

with zRTP),

Signaling

encryption

(TLS)

Unknown

Text messaging,

audio/video

telephony, IPv6, call

recording

updated

daily

(December

26, 2010; 2

months ago)

KPhone Linux (KDE)

GPL /

Free

software

YesSIP, STUN,

NAPTR/SRVSRTP Unknown

Video, voice, IM,

external Sessions,

IPv6 support for

UDP

1.2

(November

2008)

Linphone Linux, Windows

GPL /

Free

software

Yes SIP No UnknownVideo, IM, STUN,

IPv6

3.4.1 (Feb

2011)

Lotus

Sametime

Linux, Mac OS X,

Windows, mobile

Closed

Proprietar

y

NoSIP, SIMPLE,

T.120 and H.323TLS Unknown

IM, File transfer,

Voice, Presence,

Server stored

contact list, HTTP

tunneling, plugins,

embedable in Lotus

Notes

8.5 (22.

December

2009)

Mirial

Softphone

(Mirial

s.u.r.l.)

Windows

2000/XP/2003/Vista/

7 (including 64bit

versions), Mac OS X

(x86)

Closed

Proprietar

y

No SIP, H.323,

RTSP

DTLS-SRTP Unknown H.264 Full-HD

1080p video rx/tx,

Two independent

lines supporting

Call Control and 3-

Partyvideoconference in

Continuous

Presence, G.722.1/C

wideband audio,

Call

recording/export,

DV/HDMI/Compon

ent capture,

Presentation (H.239,

RFC-4796),

Encryption, Far End

7.0.24 (May

26, 2010)

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Multiple

realms

authentificati

on

mechanism

call recording,

Multi-way

conferencing

SightSpeed Mac OS X, Windows

Freeware

/ Closed

Proprietar

y

No

SIP,

RTP,Proprietary

P2P protocol

Unknown Unknown

Video, voicemail,

phone in, phone out,

multiparty calling,

conference

recording, text

messaging, NAT

traversal, video mail

6.0

Skype Linux, Mac OS X,

Windows

2000/XP/Vista/7/Mobile (no longer

supported), BREW,

Android, iPhone, PSP

Freeware

/ Closed

Proprietary

No Proprietary P2P

protocol; SIP

users canconnect to the

Skype network

using alternate

software/hardwa

re, but the Skype

software does

not support it

directly

Yes 25

starting

withversion

3.6.0.216.

10 with

2.x

Conferencing,

video, file transfer,

voicemail, Skype tophone, phone to

Skype, additional

P2P extensions

(games, whiteboard,

etc...); depending on

platform.

5.2.60.113

(Windows)

5.0.0.7994(Mac OS X)

2.1.0.81

(Linux)

1.5.0.12

(Symbian)

(March 15,

2011; 4 days

ago

(Windows)

January 27,

2011; 51

days ago

(Mac OS X)

January 20,

2010; 13

months ago

(Linux)

December 1,

2010; 3

months ago

(Symbian).

The version

numbers are

not

synchronize

d, i.e. the

features in

Mac OS

2.0.0

version are

not the same

as those

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found in

Linux 2.0.0

version.)

Spikko

Windows

2000/XP/Vista/7/Mo

bile , iPhone,

Freeware

/ Closed

Proprietar

y

No SIP Yes 8

Conferencing,

voicemail, PC to

phone, phone to PC,

Free international

phone numbers,

address book

integration;

Dec 2010

TeamSpeakLinux, Windows,

Mac OS X

Freeware

Closed /

Proprietar

y

NoYes

(Optional)Unknown

Conferencing, File

Transfers3.0.0-beta36

Telephone Mac OS X 10.5

BSD /

Free

Software

Yes SIP, STUN, ICE No UnknownAddress Book

integration0.14.0

TokboxMac OS X, Windows

XP/2000/Vista

Freeware

/ Closed

Proprietar

y

No Unknown Unknown Unknown

Video calling, video

conferencing, chat,

IM (MSN, AIM,

Yahoo!, Google

Talk)

Unknown

TpadWindows

2000/XP/Vista

Freeware

/ Closed

Proprietar

y

No SIP, STUN Unknown U nknown

Call Forwarding,

PC to PSTN, PSTN

to PC, Voicemail to

email

3.0.1

Tru App

Windows

2000/XP/Vista/7,

Mac OS X, Linux

iOS, Android,

Symbian, BlackBerry

OS,

Freeware

/ Closed

Proprietar

y

No SIP, XMPP Unknown Unknown

Chat, file transfer,

voicemail, inbound

numbers, integration

with GTalk,

Microsoft Live,

Skype

Twinkle Linux

GPL /

Free

software

Yes SIP SRTP, ZRTP Unknown

Conferencing, chat,

file transfer, Firefox

integration, call

redirection,

voicemail, support

of VoIP-to-Phone

services

1.4.2

(2009-02-

25)

VbuzzerWindows

2000/XP/Vista

Freeware

/ Closed

Proprietar

y

No SIP TLS Unknown

IM (MSN),

voicemail,

personalized voice

greeting.

2.0.282

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Ventrilo Mac OS X, Windows

Freeware

/ Closed

Proprietar

y

No No UnknownConferencing, chat,

text-to-speech3.0.7

Voice

Operator

Panel

Windows

2000/XP/Vista

Closed

Proprietar

y

No SIP, RTP Unknown Unknown

Call forwarding,

Call transfer, Call

recording, Presence,

Outlook integration,

Windows

Messenger/MSN/Li

ve integration,

CRM, Built-in web

browser & e-mailer,

LDAP, APS.

1.3.2

X-LiteMac OS, Windows,

(Linux)

Freeware

/ Closed

Proprietar

y

No SIP, STUN, ICE Yes Unknown

IM, single loginaccount, for

Windows and Mac

also Conferencing,

Video and SIMPLE

based presence]

Windows /

Mac OS: 4.0

/ Linux:

Discontinue

d

Yahoo!

Messenger

Mac OS (8, 9, X),

Windows,

(Linux/FreeBSD

version not VoIP

capable)

Freeware

/ Closed

Proprietar

y

No

SIP (using TLS)

and RTP

(media)

Unknown Unknown

Video, file transfer,

PC to phone, phone

to PC

ZfoneLinux, Mac OS X,

Windows

Freeware

/

Viewable

source

Proprietar

y

(includes

time

bomb

provision

)

No SIP, RTP SRTP, ZRTP Unknown

Beta 2008-

09-04

(Linux

0.9.224),

(Mac OS

0.9.246),

(Windows

0.9.206)

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Fig 7.1:- Virtual Box

7.1 Why is virtualization useful?

Running multiple operating systems simultaneously. Virtual Box allows you to

run more than one operating system at a time. This way, you can run software

written for one operating system on another (for example, Windows software on

Linux or a Mac) without having to reboot to use it. Since you can configure what

kinds of virtual hardware should be presented to each such operating system, you

can install an old operating system such as DOS or OS/2 even if your real

computers hardware is no longer supported by that operating system.

7.2 Supported host operating systems

Currently, Virtual Box runs on the following host operating systems:

7.2.1 Windows hosts:

Windows XP, all service packs (32-bit)

Windows Server 2003 (32-bit)

Windows Vista (32-bit and 64-bit1).

Windows Server 2008 (32-bit and 64-bit)

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Networking This package contains extra networking drivers for your Windows

host that Virtual Box needs to support Bridged Networking (to make your VMs virtual

network cards accessible from other machines on your physical network).

Python Support This package contains Python scripting support for the Virtual

Box API.For this to work, an already working Windows Python installation on the

system is required.1

Depending on your Windows configuration, you may see warnings about unsigned

drivers or similar. Please select Continue on these warnings as otherwise Virtual

Box might not function correctly after installation.

The installer will create a Virtual Box group in the Windows Start menu which

allows you to launch the application and access its documentation.

VBoxApplication Main binaries of Virtual Box.

Note: This feature must not be absent since it contains the minimum set of files

to have working Virtual Box installation.

VBoxUSB USB support.

VBoxNetwork All networking support; includes the VBoxNetworkFlt and

VBoxNetworkAdp features (see below).

VBoxNetworkFlt Bridged networking support.

VBoxNetworkAdp Host-only networking support.

VBoxPython Python support.

7.3.3 Uninstallation

As Virtual Box uses the standard Microsoft Windows installer, Virtual Box can besafely uninstalled at any time by choosing the program entry in the Add/Remove

Programs applet in the Windows Control Panel.

7.4 Starting Virtual Box

After installation, you can start Virtual Box as follows:

On a Windows host, in the standard Programs menu, click on the item in the

Virtual Box group. On Vista or Windows 7, you can also type Virtual Box in the

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search box of the Start menu.

When you start Virtual Box for the first time, a window like the following should

come up:

Fig 7.2:- Welcome to Virtual Box

This window is called the Virtual Box Manager. On the left, you can see a pane

that will later list all your virtual machines. Since you have not created any, the list is

empty. A row of buttons above it allows you to create new VMs and work on existing

VMs, once you have some. The pane on the right displays the properties of the virtual

machine currently selected, if any. Again, since you dont have any machines yet, the

pane displays a welcome message.

To give you an idea what Virtual Box might look like later, after you have created many

machines, heres another example:

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Fig 7.3:- Virtual Box Main Menu

7.5 Creating your virtual machine

Click on the New button at the top of the Virtual Box Manager window. A

wizard will pop up to guide you through setting up a new virtual machine (VM):

Fig 7.4:- Create New Virtual Machine

On the following pages, the wizard will ask you for the bare minimum of information

that is needed to create a VM, in particular:

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Fig 7.5:- Choosing Operating System

7.5.1 Virtual Machine Name :

The VM name will later be shown in the VM list of the Virtual Box Manager

window, and it will be used for the VMs files on disk. Even though any name could

be used, keep in mind that once you have created a few VMs, you will appreciate if

you have given your VMs rather informative names; My VM would thus be less

useful than Windows XP SP2 with Open Office.

7.5.2 Operating System Type :

select the operating system that you want to install later. The supported

operating systems are grouped; if you want to install something very unusual that is

not listed, select Other. Depending on your selection, Virtual Box will enable or

disable certain VM settings that your guest operating system may require. This is partic-

ularly important for 64-bit guests. It is therefore recommended to always set it to the

correct value.

7.5.3 Virtual Machine RAM :

On the next page, select the memory (RAM) that Virtual Box should allocate

every time the virtual machine is started. The amount of memory given here will be

taken away from your host machine and presented to the guest operating system, which

will report this size as the (virtual) computers installed RAM.

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a new disk image. Hence, press the New button.

This brings up another window, the Create New Virtual Disk Wizard, which

helps you create a new disk image file in the new virtual machines folder.

Fig 7.8:- Virtual Disk Wizard

Press Next to continue.

Fig 7.9:- Type of Virtual Hard Disk

Virtual Box supports two types of image files:

A dynamically expanding file will only grow in size when the guest actually

stores data on its virtual hard disk. It will therefore initially be small on the host hard

drive and only later grow to the size specified as it is filled with data.

A fixed-size file will immediately occupy the file specified, even if only a

fraction of the virtual hard disk space is actually in use. While occupying much more

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space, a fixed-size file incurs less overhead and is therefore slightly faster than a

dynamically expanding file.

To prevent your physical hard disk from running full, VirtualBox limits the size of the

image file. Still, it needs to be large enough to hold the contents of your operating

system and the applications you want to install for a modern Windows or Linux

guest, you will probably need several gigabytes for any serious use:

Fig 7.10:- Size of Virtual Hard Disk

After having selected or created your image file, again press Next to go to the next

page.

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Fig 7.11:- Summary of Virtual Hard Disk

7.5.5 Finish:

After clicking on Finish, your new virtual machine will be created. You will

then see it in the list on the left side of the Manager window, with the name you

entered initially.

7.6 Running your virtual machine

Fig 7.12:- Running New Virtual Machine

To start a virtual machine, you have several options:

Double-click on its entry in the list within the Manager window or

select its entry in the list in the Manager window it and press the Start button at

the top or

for virtual machines created with VirtualBox 4.0 or later, navigate to the

VirtualBox VMs folder in your system users home directory, find the subdirectory of

the machine you want to start and double-click on the machine settings file (with a

. v b o x file extension).

This opens up a new window, and the virtual machine which you selected will boot up.

Every-thing which would normally be seen on the virtual systems monitor is shown in

the window.

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In general, you can use the virtual machine much like you would use a real computer.

There are couple of points worth mentioning however.

7.6.1 Starting a new VM for the first time

When a VM gets started for the first time, another wizard the First Start

Wizard will pop up to help you select an installation medium. Since the VM is

created empty, it would otherwise behave just like a real computer with no operating

system installed: it will do nothing and display an error message that no bootable

operating system was found.

Fig 7.13:- First Run Wizard

For this reason, the wizard helps you select a medium to install an operating system

from.

If you have physical CD or DVD media from which you want to install your guest

operating system (e.g. in the case of a Windows installation CD or DVD), put the

media into your hosts CD or DVD drive.

Then, in the wizards drop-down list of installation media, select Host drive with

the correct drive letter (or, in the case of a Linux host, device file). This will allow

your VM to access the media in your host drive, and you can proceed to install from

there.

If you have downloaded installation media from the Internet in the form of an ISO

image file (most probably in the case of a Linux distribution), you would normally

burn this file to an empty CD or DVD and proceed as just described. With VirtualBox

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however, you can skip this step and mount the ISO file directly. VirtualBox will then

present this file as a CD or DVD-ROM drive to the virtual machine, much like it does

with virtual hard disk images.

For this case, the wizard

s drop-down list contains a list of installation media that

were previously used with VirtualBox.

If your medium is not in the list (especially if you are using VirtualBox for the

first time), select the small folder icon next to the drop-down list to bring up a

standard file dialog, with which you can pick the image file on your host disks.

Fig 7.14:- Select Installation Media

In both cases, after making the choices in the wizard, you will be able to install your

operating system.

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Fig 7.15:- Summary First Run Wizard

Press Finish.

CHAPTER 8

8. Elastix

Elastix is an appliance software that integrates the best tools available for

Asterisk-based PBXs into a single, easy-to-use interface. It also adds its own set of

utilities and allows for the creation of third party modules to make it the best software

package available for open source telephony.

The goals of Elastix are reliability, modularity and ease-of-use. These characteristics

added to the strong reporting capabilities make it the best choice for implementing an

Asterisk-based PBX.

The features provided by Elastix are many and varied. Elastix integrates many softwarepackages, each including their own set of great features. However, Elastix adds new

interfaces for control and reporting of its own, to make it a complete package. Some of

the features provided natively by Elastix are:

VIDEO support. You can use videophones with Elastix!

Virtualization support. You can run multiple Elastix virtual machines on the

same box.

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Really friendly Web user interface.

"Fax to email" for incoming faxes. Also, you can send any digital document to a

fax number through a virtual printer.

Billing interface.

Graphical configuration of network parameters.

Resource usage reporting.

Remote restart/shutdown options.

Incoming/outgoing calls and channel usage reports.

Integrated voicemail module.

Voicemail Web interface.

Integrated operator panel module.

Extra SugarCRM and Calling Card modules included.

Download section with commonly used accessories.

Embedded help interface.

Instant messaging server (Openfire) integrated.

Multi-lingual support. Languages supported include:

o English

o Spanish

o Russian

o Korean

o Greek

o Chinese

o Polish

o German

o French

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immediately.

You will see it commence a basic start up, load a few drivers and will next stop at the

screen below

Fig 8.2:- Choosing Language

For these and all following screens, you use a combination of the up and down arrows,

the button and the bar. The space bar acts as the button,

moves between the sections (e.g. between selection of the language and the

OK button in the above screen). The bar is also used to toggle the * in

multiple selections.

Select your language using the arrow keys and then press to move to the OK

button. Once the OK is highlighted you can then press .

The following screen will appear

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Fig 8.3:- Keyboard Type

For most users, the US keyboard will suit, so press to move the highlight to the

OK button and press the bar.

The next screen may or may not come up on your installation, depending on whether

you have a clean Hard Drive with no data or you have a Hard Drive with a partition

already on it. In this case we are working with a new hard drive. The black mark out in

the diagram below may vary from system to system, so I have blanked it out to avoid

confusion.

Fig 8.4:- Warning

In this screen it is telling us that it wants to initialize the drive and erase all data. The

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YES button is already highlighted, so we proceed by pressing .

Now the next screen needs a little bit of tender care.

Fig 8.5:- Partitioning Type

The reason for this is that the default selections need to be changed, as the defaults have

been set to avoid you accidentally erasing the data on your hard drive

You need to use the arrow keys to move the selection up to REMOVE ALLPARTITIONS as shown in the previous screen. If you have multiple drives in your

system, you need to make sure that it has chosen the correct drive. Now use TAB to

move to the OK button and press the bar.

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Fig 8.6:- Warning

You need to use the key to select the YES button and press SPACE if you are

sure that there is no useable data on this drive.

Fig 8.7:- Partitioning layout

Again use the key to move the highlight, this time to the NO button. Unless

you are very familiar with Linux Partitioning, then you don't want to review andpossibly change the partitioning, so just take the easy option and select NO.

The next screen allows us to configure the network card on your machine.

Fig 8.8:- Configure Network Interface

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So just press the bar and the next screen will appear

Fig 8.9:- Ethernet Configuration

This is one of the screens where you need to use the space bar to select your options.

You definitely need to ACTIVATE ON BOOT (otherwise it will not start the Network

Card), and as a minimum select ENABLE lPv4 support. Unless you 100% know what

you are doing, I would leave lPv6 support not enabled.

Press the key to move to highlight the OK button and proceed to the next screen

Fig 8.10:- IPv4 Configuration for Ethernet

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This is where you set your Network card settings. If you want to use DHCP, then select

DHCP, and the Network card will pick up the settings from your DHCP server on your

network (if you have one). For 99% of systems however, most will be setup with a

STATIC IP (manual) address).

Now to the ok button and press

The following screen will appear

Fig 8.11:- Network Settings

Here you set the Gateway, Primary DNS and Secondary DNS IP addresses. Again you

should know these. On many systems, the Gateway is your router, your primary DNS

server would normally be a DNS Server on your Network (e.g. a Windows or Linux

Server) and as a backup a good option if your router acts as a DNS proxy (most do),

then select your router as the secondary DNS.

Press to get to the OK button and press to move to the next screen.

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Fig 8.12:- Host Name Configuration

Here you just select Manually (which is the default) and type in a name for your

server. It is not critical what the name is, just something unique to identify your

server on the network. Press to highlight the OK button and press

to move to the next screen.

Fig 8.13:- Time Zone

In this screen we set the time zone. Select the time zone you are in and press to

move to the OK button and press .

The next screen and what you place in here is critical

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Fig 8.14:- Root Password

This is ROOT password screen and what you enter here needs to be written down. The

number of people who don't write this down, or forget it is, or say that this screen did

not come up is quite bad. The reason for this is that some more password screens come

up as part of the install, and they forget which password is which. The result of losing

this password results in a complete reinstall of the Elastix product, or a lot of technical

reading and understanding of Linux to understand how to reset this password. WRITE

IT DOWN before you enter it in here.

One other word of warning, make sure of the status of your Key,

especially with the use of the key many inadvertently press the

key due their close proximity to each other.

to the OK button and press bar.

You will now witness a variety of screens pop up, which include the formatting

screen, working out dependencies, transferring image, and finally you should see the

Package Installation screen. All these screens will occur without your input. As a

guide, the Package Installation screen should be started within a few minutes of your

last press of the OK button. However, this can vary especially on the formatting

screen if you have a large hard drive.

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Fig 8.15:- Package Installation

This Package Installation screen will probably run anywhere between 5 - 18 minutes

depending on the speed of your machine / hard disks etc.

When it's finished the system will reboot, hopefully eject the CD (which you can now

and should remove. You will notice on boot up, that the various lines will have a green

OK next to each of them, except that there will probably be a red FAIL next to

WANPIPE. This is ok, don't panic. This will only ever be a OK when you use the

SANGOMA product, and have it configured properly.

The next screen that will pop up will be the password entry screen for MYSQL. Enter

a different password than what you used for the previous ROOT password. Again

WRITE IT DOWN now before you enter it. Check the status to make

sure you are entering it correctly.

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Fig 8.16:- MySQL Password

The next screen will ask you to confirm the MYSQL password you just entered. Enter it

again

Fig 8.17:- Confirm MySQL Password

It will then run off and perform some password scripts which complete and then come

up with the next screen.

This next screen will now ask you to set the password for the rest of the products

included with Elastix. These products include the Elastix Web Login, Freepbx, Vtiger,

and A2Billing. The user name is automatically admin, so you are just setting the

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default password here (don't worry they can be changed later within each application).

It is important that they have a decent password from the start. WRITE IT DOWN

before you enter it in here.

Fig 8.18:- Admin Password

The next screen will ask you to confirm it.

Fig 8.19:- Confirm Admin Password

Complete these steps and then you will be rewarded with the following screen

after it has completed its startup scripts.

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Fig 8.20:- Elastix Login

At this point, your Elastix system is installed. Now you probably want to see the Web

GUI to start exploring your Elastix system. On a separate workstation, in your Internet

Browser (Firefox is the preferred browser) enter the following address into the address

bar: htt p://{YourElastixPrimarvlPAddress} and press enter (e.g. http://172.22.22.40)

You will then see the following screen but don't panic

Fig 8.21:- Elastix Web Interface

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This has popped up as the Elastix system utilizes SSL for all configuration pages, but

your system does not have a valid SSL Certificate. Depending on your need you can

purchase your own SSL Certificate, so in the meantime, we trust the system we are

communicating with and we need to let the browser know this. Each browser/version

has a different way of handling this, so you need to work out how it works on your

browser.

On Firefox you click on I Understand the Risks and then click on Add Exception and

when the next page shows click on Confirm Security Exception.

At this point, you will now be presented with the main Elastix login page as

shown in the next diagram.

Fig 8.22:- Elastix GUI

Use the admin login and password which hopefully you wrote down for the

Elastix GUI. That's it, now all that is left is to login and start exploring and

configuring.

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CHAPTER 9

9. WEB Administration Interface

9.1. WEB Configuration

9.1.1 Network Parameters

Go to Network section.

9.1.2 Configuration of telephonic hardware

Go to Port Details.

9.1.3 Creation of new extension

This area is for handsets, softphones, paging systems, or anything else that could

be considered an 'extension' in the classical PBX context.

Defining and editing extensions is probably the most common task performed by a

PBX administrator, and as such, you'll find you'll become very familiar with this page.

There are presently four types of devices supported - SIP, IAX2, ZAP and 'Custom'.

To create a New Extension, go to the PBX menu, which by default goes to the

Configuration PBX section; in this section, choose the option Extensions on the

left panel. Now we can create a new extension.

First, choose the device from among the available options:

Fig 9.1:- Add Extension

Generic SIP Device: SIP is the Standard protocol for VoIP handsets and ATA's.

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Generic IAX2 Device: IAX is 'Inter Asterisk Protocol', a newer protocol supported by

only a few devices (eg, PA1688 based phones, and the IAXy ATA).

Generic ZAP Device: ZAP is a hardware device connected to your Asterisk machine

- E.g., a TDM400, TE110P

Other (Custom) Device: Custom is a 'catch all', for any non standard device, eg

H323. It can also be used for "mapping" an extension to an "outside" number. For

example, to route extension 211 to 1-800-555-1212, you could create a custom

extension 211 and in the "dial" text box you could enter:

Local/18005551212@outbound-allroutes.

Once the correct device has been chosen, click on Login.

Note: Now we proceed to input the necessary fields (obligatory) to create a new

extension.

Continue to enter the corresponding information:

Fig 9.2:- Add Sip Extension

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Fig 9.4:- Add Sip Account

Next, let's go to the Audio Codecs section and select all of the available codecs. We

apply the changes and click on the Register button, so that our telephone registers in

the system.

Fig 9.5:- Audio Codecs

Finally, you can make a call from one extension to another.

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9.1.5 Recording of welcome/greeting message

This section describes how to record a message or activate one that was created

in another medium.

To access this module, go to the PBX menu, where the Configuration PBX

section will appear by default. In the left panel, choose the System Recordings

option.

Fig 9.6:- System Recordings

The first option that we have is to create an announcement by recording it directly. For

this, we will need to enter the extension from which we want to make the recording,

which in this case is extension 201, then we can click on the Go button.

Next, Asterisk will be waiting for our recording at extension 201, and to continue, we

have to punch in *77. After recording our message, press the pound sign (#).

To review our recording, press *99, enter the name of the recording and click on the

Save button.

The second option that we have is to upload a recording that was created in another

medium. For that, we will need to have a file that's supported by Asterisk; click on the

Find button and locate our file. Then, continue to give the recording a name and,

finally, click on the Save button.

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9.1.6 Configuration of welcome IVR

The IVR allows us to record a welcome message and allows us to have a menu

controlled by the telephone keys (10 number keys, plus the symbols pound '#' and

asterisk '*'). With this, it is possible to send the call to another destination or to the

IVR that sent the announcement.

To access the IVR module, go to the PBX menu, which appears by default in the

Configuration PBX section. In the left panel, choose the IVR option.

Fig 9.7:- Digital Receptionist

To record a welcome or greeting message, go to the System Recordings section, for

example:

IVR: Thank you for calling Elastix. If you know the extension, please dial it now.

Otherwise, stay on the line and an operator will be with you shortly.

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Fig 9.8:- Edit Menu Digital Receptionist

To add a new IVR, it's not necessary to complete all of the fields, and in our case (a

welcome/greeting IVR), we do not need options. The necessary fields are the

following:

Change Name: To change the name, we'll put Welcome.

Timeout: Waiting time (in seconds) before the call is routed to an operator after the

welcome message is played. For this example, we will use 3.

Enable direct dial: An option that permits the caller to dial an extension directly en

case he or she knows it, without having to wait for the operator.

Announcement: This is the announcement or welcome message that was recorded

earlier. It will appear in a list with all of the available messages.

Now we can proceed to configure certain options that are frequently used. The first is

the option 0 (zero) that allows us to go directly to the operator and the second is also

to go to the operator, but the caller has to listen to the welcome message and wait for

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the time that was configured earlier to pass.

Among the available options on the menu, in the left part there is a box where you

should put the option. For the first one (zero), we'll put that in the box and assign an

extension that was previously configured; this extension will be the operator.

These extensions will appear after the option Core.

Now we'll proceed to configure the second option (to go to the operator after the

welcome message is played and the waiting time is over). In the box to the left, put the

letter t, which means timeout and we'll assign the operator's extension.

Finally, let's record the IVR.

9.1.7 Fax Configuration

Go to MENU: FAX.

9.2. Reference to available modules

9.2.1. MENU SYSTEM

9.2.1.1 System Info

The option System Info of the Menu System in Elastix lets us monitor the

servers hardware resources. Within this option, we have two sections:

System Resources

System Resource shows us the values of actual use of both the memory as well as

the processor.

Fig 9.9:- System Resource

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CPU Info Information about brand, model and processor speed

Uptime Time from the last reboot of the server

CPU usage Percentage of use of processor capacity

Memory usage Percentage of RAM memory utilized

Swap usage Percentage of SWAP memory utilized

Here is a graphic with the statistics of simultaneous calls, percentage of use of

processor and percentage of use of RAM memory.

Fig 9.10:- System Resources Graphs

Hard Drives

This section shows a summary of the utilization of storage available on the server.

Fig 9.11:- Hard Drives

9.2.1.2 Network

The option Network of the Menu System in Elastix lets us view and

configure the parameters of the network of the server.

Within this option we have two sections:

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Network Parameters

Fig 9.12:- Network Parameters

This corresponds to the general network parameters of the server:

Host Server Name, for example: pbx.subdomain.com

Default

GatewayIP Address of the Port of Connection (Default Gateway)

Primary DNS IP Address of the Primary Domain Name Server (DNS)

Secundary

DNS

IP Address of the Secondary or Alternative Domain Name

Server (DNS)

To change any of these parameters, click on the button Edit Network Parameters.

Ethernet Interfaces List

This shows the list of network interfaces available on the server, with the following

data:

Fig 9.13:- Ethernet Interfaces Links

Device Name of the Operating System that is assigned to the Interface

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Type The type of IP address that the Interface has, which could be

STATIC when the IP address is fixed or DHCP when the IP

address is obtained automatically when the equipment is booted

up. To use the second option, there should be a DHCP server in

the network.

IP IP Address assigned to the Interface

Mask The Network Mask assigned to the Interface

MAC

Address

Physical Address of the network Interface

HW Info Additional information about the network Interface

Status Shows the physical status of the Interface, if its connected or not

To change the parameters of any of the Interfaces, click on the name of the device.

The only values that can be changed are: Type, IP and Mask

Fig 9.14:- Edit Interface

9.2.1.3 User Management

Users

The option Users allows us to create and modify the users who will have

access to the Elastix Web Interface. There are three types or groups of users:

1 Administrator

2 Operator

3 Telephone User

Each of these groups represents distinct levels of access to the Elastix Web Interface.

These levels signify the group of menus to which each type of user has access. The

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distinct permissions for access to the menus are better illustrated in the following

table:

Menu Administrator

Operato

r Telephone User

Menu: System

System Info Yes Yes No

PBX Configuration Yes No No

Network Yes No No

User Management Yes No No

Shutdown Yes No No

Operator Panel

Flash Operator Panel Yes Yes No

Voicemails

Asterisk Recording

InterfaceYes Yes Yes

Fax

Virtual Fax List Yes Yes No

New Virtual Fax Yes No No

Reports

CDR Report Yes Yes No

Channels Usage Yes Yes No

Billing

Rates Yes No No

Billing Report Yes No No

Destination Distribution Yes No No

Trunk Configuration Yes No No

Extras

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SugarCRM Yes Yes Yes

Calling Cards Yes Yes Yes

Downloads

Softphones Yes Yes Yes

Fax Utilities Yes Yes Yes

Group Permission

The option Group Permission of the Menu System in Elastix lets us

determine the menus to which each group of users will have access.

The list below shows the names of the Elastix menus; you should select the ones that

each group should have permission to access, and then click the Apply button.

Fig 9.15:- Group Permission

9.2.1.4 Language

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The option Language of the Menu System in Elastix lets us configure the

language for the Elastix Web Interface.

Fig 9.16:- Language

Select the language from the list and click the Change button.

9.2.1.5 Date and Time Configuration

The option Date and Time Configuration of the Menu System Info in Elastix

lets us configure the Date, Hour and Time zone for the Elastix Web Interface.

Fig 9.17:- Date and Time Configuration

Select the new date, hour and time zone and click on the Apply changes button.

9.2.1.6 Load Module

To upload a new module, click on the Examinar button, select the file and,

finally, click on the Save button.

Fig 9.18:- Load Module

9.2.1.7 Backup

The option Backup of the Menu System Info in Elastix lets us choose the

configurations that we desire to backup Elastix.

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Fig 9.19:- Backup

To make a Backup of the Elastix configurations, select from the available options, and

click on the Process button.

9.2.1.8 Restore

The option Restore of the Menu System Info in Elastix lets us choose the

configurations to restore Elastix, apart from the aforementioned Backup.

Fig 9.20:- Restore

To restore the Elastix configurations, select from the available options, input the path

of the resto

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