FAKULTI TEKNOLOGI KEJURUTERAAN

ELEKTRIK DAN ELEKTRONIK

UNIVERSITI TEKNIKAL MALAYSIA MELAKA

TELECOMMUNICATION SWITCHING SYSTEM

BEET3393 SEMESTER 2 SESI 2017/2018

LAB 1: LOCAL SIGNALLING AND NUMBERING ASSIGNMENT

NO. STUDENTS' NAME MATRIC. NO.

1.

2.

3.

PROGRAMME 3 BEET

SECTION /
GROUP

DATE

1.
NAME OF
INSTRUCTOR(S)
2.

EXAMINER’S COMMENT(S) TOTAL MARKS

Rev. Date Author(s) Description
No.

1.0 30 Jan 1. Fakhrullah Idris 1. Update to new UTeM logo

2019 2. Win Adiyansyah Indra 2. Update faculty's name
3. Change "course" to
"programme"
4. Remove verification stamp
5. Update Content
1.0 OBJECTIVES
1. To provide guidelines for the configuration of a basic IP telephony system.
2. To register IP Phone 7830 to eSpace U1910.
3. To register IP Phone 7810 to eSpace U1980.
4. To understand signalling between telephones and local switching centre.
5. To register numbering assignment in eSpace U1980 & eSpace U1910.

2.0 EQUIPMENT

1. eSpace U1980
2. eSpace U1910
3. IP Phone 7810
a. IP Phone 7830

3.0 SYNOPSIS & THEORY

IP telephony or Voice over Internet Protocol (VoIP) is a general term for the technologies
that use the Internet Protocol’s packet-switched connections to exchange voice, fax and other
forms of information that have traditionally been carried over the dedicated circuit-switched
connections of the public switched telephone network (PSTN).
Up to now, companies had two separate communication infrastructures. On one side,
there was the public branch exchange (PBX), featuring voice services provided by the company's
own switchboard. On the other side, we had the LAN (Local Area Network), providing data
communication. As a rule both networks had interfaces for communicating with the outside world.
This separation is very inefficient and uneconomical, because each network needs its own
technology. Two sets of know-how are needed for operation and maintenance. Futhermore it
prevents the rapid evelopment of new applications, as the basic technology of the two systems is
so different.
Classic PBXs are proprietarily based on the system architecture of the respective provider.
Only the manufacturer of a PBX can provide and market upgrades for both hardware and
applications. For example, it’s impossible to use a system-telephone from one manufacturer with
another system. Common wired telephone connections use a direct connection between the
callers. This is called connection oriented communication. The sequential dataflow always follows
the same path from the source to the recipient, and the delay always remains the same. The
connection - usually a 64 kbit/s line - is not available for other information transfer. Pauses in
speech, which make up to 60% of a standard phone call, mean that the connection is not being
used at full capacity.
The IP-PBX is a new type of PBX. Here speech is not routed over a separate
infrastructure. Instead, it’s an integrated component of a common infrastructure for speech and
data communication in a multimedia network. The packet-oriented and connectionless
communications protocol IP (Internet protocol) is fundamental for linking these infrastructures. With
the rapid development of the internet, IP has become very important in the world of
communications. Speech is compressed, wrapped in IP packets and simultaneously transported
on the same network along with other data. If speech is transmitted, the bandwidth of the
datanetwork is used only partially. As audio and video data are transferred by IP, this is often
referred to as IP telephony or Voice over IP (VoIP).

Local Signalling:
The signalling between the telephone user, and the local Switching Centre is known as
local signalling. The signals available to the user are the Switch Hook and the Keypad. The Switch
Hook operates as soon as the telephone is lifted. This is the Off Hook state of the telephone, and
is recognised by the Switching Centre. The Keypad is primarily used to send the Destination
Address to the Switching Centre; that is the number of the telephone to which the connection is
required.

Dual Tone Multi-Frequency (DTMF) signalling

The signals are in the form of a combination of two audible tones, a different combination
for each number on the Keypad. Hence it is known as Dual Tone Multi-Frequency (DTMF)
signalling. The Switching Centre can send signals to the user, by using tones and by ringing the
bell or alerter in the telephone. The audible tones are known as call progress tones, and indicate to
the user important responses of the Switching Centre. Obviously they are only useful if the user is
listening to the telephone. If the telephone is not in use, i.e. if it is on hook, then the Switching
Centre can ring it.

ITU-T Standards: Standards for the telephone industry are agreed by an international
body known as the Telecommunications Standardisation Sector of the International
Telecommunications Union (ITU-T).Up to 1994 it had a French name, Comite Consultatif
International de Telegraphique et Telephonique (CCITT). The ITU-T produced a Standard
Recommendation E.180 for the tones used in local signalling. Each telephone system is run by an
Administration, sometimes a public administration, usually running the whole telephone system in
one country, and sometimes privately owned. Historically each Administration has often used
different tones for the same purpose. The ITU-T Recommendation aims to reduce these
differences so that in international calls operators and users understand easily the meaning of the
tones.

The Recommendation includes 'acceptable' tones for each purpose, and 'recommended' tones for
new systems. The tones that are defined in the standard including Dial Tone, Ringing Tone, Busy
Tone, Caller waiting tone and Number Unobtainable Tone.

Numbering Assignment

Figure 3.1 International ITU-T E.164-number structure for geographic areas

ITU-T also recommend numbering assignment for telephony services. Recommendation

ITU-T E.164 provides the number structure and functionality for the five categories of numbers
used for international public telecommunication: geographic areas, global services, Networks,
groups of countries (GoC) and resources for trials. For each of the categories, it details the
components of the numbering structure and the digit analysis required to successfully route the
calls. Figure 3.1 explains the numbering structure for geographic areas. The country code is
assigned by ITU-T, while National Number significant to the country is assign by the respective
country.

Numbering allocation and usage in Malaysia, falls under administration of The Malaysian
Communications and Multimedia Commission (MCMC). MCMC may assign new blocks of
numbers from time to time upon request by telephony service operators. Telephony service
providers shall utilize the assigned numbers effectively and efficiently.The numbers shall be
offered to customers registered in Malaysia. This is defined as a customer having either an Internet
access, cellular telephone or fixed telephone subscription account in Malaysia.
4.0 PROCEDURE
Part A Configure IP address for IP phone eSpace 7830 using the keypad
This section shows the steps using the keypad to confitgure an IP phone with the IP address
192.168.XX.XX
1. Press Menu on the IP phone to access the main menu.
2. Press 3. The Setting Type page is displayed.
3. Press 2. The Please enter Password page is displayed. Enter the password (the initial
password is admin or admin123).
4. Press Confirm to access the Advanced Settings page
5. Press 2 to display the Network page. Press 1 to display WAN Port Option page.
6. Press 2. The Static IP Client page is displayed.
7. Press the up arrow key or down arrow key to configure the following information:
a. IP address: 192.168.XX.XX
b. Subnet Mask: 255.255.255.0
c. Default Gateway: 192.168.10.1
8. Press Save to set the configuration
9. Press Back to return to the Network page. Press Back again and the following
messages are displayed,
Networking updating
Please wait…
10. The setting take effect after the IP phone is restarted. The following messages are
displayed after it restarted.
Initializing
Please wait…
11. Configure IP phone eSpace 7810 using the keypad using similar setup as part A.

Part B Local Signalling & Dual Tone Multi-Frequency (DTMF) signalling

1. Put all 4 telephones face down (On Hook).

2. Pick up telephone 1. Dial Tone is heard.
3. Dail telephone 2. Listen to the tone while dialling. Write your observation.
4. Ring Tone and Alerting (Ring) are applied to telephones 1 and 2. Pick up telephone 2. Speak
into one of them to check the connection.
5. While the first connection is held, pick up telephone 3 and dail telephone 1. Observe and listen
to the phones. Then replace telephone 3.
6. Replace telephone 2. Is the connection broken? Replace telephone 1.
7. Using any telephone, listen to Dial Tone, and then dail any unassigned number. Listen to the
IVR.

Transfer Call and Conference Call

3. User can transfer a call to other extension numbers or mobile numbers.
a. User 0100 makes a call to user 0101. While in the conversation, user 0100 press
theTransfer key to hold user 0101’s call.
b. User 0100 dials user 0102.
c. User 0100 can performs the following transfer call option:
i. Blind transfer
 After user 0102 answers the call, user 0100 hangs up the phone to
complete the transfer. If user 0102 rejects the call, a message indicating
transfer failure is displayed on the user 0100’s phone screen. Press
Resume key, user 0100 resumes the call with user 0101.
ii. Semi-attend transfer
  When user 0102’s phone rings, user 0100 presses the Transfer key.
User 0100’s phone hangs up automatically and the transfer is complete.
iii. Attend transfer
 After user 0102 picks up the phone, user 0100 asks user 0102 whether
wants to answer user 0101’s call. If user 0102 answer yes, user 0100
presses the Transfer key to transfer the call. If user 0102 rejected, user
0100 presses the Resume key to resume the call with user 0101.

Three-Party Conference call - To establish a three-party conference call, use the conference key.
User 0100 makes a call to user 0101. User 0100 presses the Conference key. User 0101 is
held and user 0100 hears the dial tone.
User 0100 dials user 0102 and presses the soft key Send.
When user 0102 answer the call, user 0100 can only talks with user 0102. To establish a
three-party conversation, user 0100 presses the Conference key again.
When user 0100 hangs up the phone, the other two user’s conversation ends automatically.

Part C Numbering Assignment

This section describes the steps to allocate numbers for SIP users. SIP users include IP phone
users, POTS phone users under SIP IAD and soft terminal users. The SIP users can be added in
batches or add in single SIP users.

1. Choose User > SIP User.

2. Configure user information on the Add number tab page. Only add TWO (2) user information
using numbering plan provided. Refer example in Figure 4.1. Figure 4.1 shows the settings to
add SIP users 0100 to 0102 in batches with start device ID 0100.
3. Table 4.1 describes the parameters.

Figure 4.1 – Adding SIP users in batches.

 Table 4.1 – Parameter description

Parameter Description Setting

Start number User number corresponding to the Set to 0100.
start number.
Number count Range of number Set to 3
Start device ID The device ID is the registration Set to 0100.
account of a SIP device. The ID of
each SIP device must be unique.
Device type Common terminal: A SIP device Select Common terminal.
for example IP phone and soft
phone are connected to eSpace
U1910.
eSpace: eSpace U1910 connects
the calls only from the eSpace
client.
Authentication mode Mode of authenticating a SIP Select By Password with
device when it registers with password 12345678.
eSpace U1910.
User right level Level of user rights with option of Set to Default Right.
Default, Common, Advanced and
Super.

4. Click OK. Select the Manage number tab to show the configured SIP users as shown in
Figure 4.2.

Figure 4.2 – Manage number

5. Configure IP phone using new numbering that you assigned.

5.0 RESULTS
(Attached all the figures)

6.0 DISCUSSION
1. When a call is made from telephone 1 to telephone 2, what happens if telephone 2 is replaced
(Switch Hook pressed), and then picked up again?

2. When a call is made from telephone 1 to telephone 2, what happens if telephone 1 is

replaced? What is this known as?

Discuss all results.

7.0 CONCLUSION