SUBMITTED BY:

Lokesh Sharma

ELECTRONICS & COMMUNICATION

A.I.E.T, Jaipur

AFFILIATED TO:

(Rajasthan Technical University)

CERTIFICATE OF APPROVAL

The foregoing project entitled, “VOICE Infra And Maintenance” is hereby approved

as a creditable study of research topic and has been presented in a satisfactory

manner to warrant its acceptance as prerequisite to the degree for which it was

submitted.

It is understood that by this approval, the undersigned do not necessarily endorse

any conclusion drawn or opinion expressed therein, but approve the thesis for the

purpose for which it is submitted.

DECLARATION CERTIFICATE

This is to certify that the work presented in the project entitled”VOICE Infra And

Maintenance” in partial fulfillment of the requirement for the award of degree of

Electronics & Communication , A.I.E.T, Jaipur is an authentic work carried out

under my supervision.

Mr. Anand Vidhate

ACKNOWLEDGEMENT

It gives me immense pleasure to present a project report on topic “HCL VOICE Infra and Maintenance”.

I am highly indebted to Mr. Anand VidhateMr. Anand Vidhate (Assistant Manager HCL Technology). He has been a source of inspiration. He always encouraged me to do something innovative.

I would like to express my deep sense of gratitude to our guide his supervision, encouragement and affection never allowed me to deviate from our objective.

I also express our sincere thanks to all others who helped with their best efforts from time to time during the project.

Special thanks to Ms. Laxmi Bhardwaj (project co-coordinator) (project co-coordinator) to support and guiding role.

OVERVIEW

HCL is a leading global Technology and IT enterprises with annual revenues of US$ 4 billion. The HCL Enterprise comprises two companies listed in India, HCL Technologies and HCL Infosystems.

The 30-year-old enterprise, founded in 1976, is one of India's original IT garage start-ups. Its range of offerings span R&D and Technology Services, Enterprise and Applications Consulting, Remote Infrastructure Management, BPO services, IT Hardware, Systems Integration and Distribution of Technology and Telecom products in India. The HCL team comprises 45,000 professionals of diverse nationalities, operating across 17 countries including 360 points of presence in India. HCL has global partnerships with several leading Fortune 1000 firms, including several IT and Technology majors

HISTORYBorn in 1976, HCL has a 3-decade rich history of inventions and innovations. In 1978, HCL developed the first indigenous microcomputer at the same time as Apple and 3 years before IBM's PC. During this period, India was a black box to the world and the world was a black box to India. This microcomputer virtually gave birth to the Indian computer industry. The 80's saw HCL

developing expertise in many other technologies. HCL's in-depth knowledge of UNIX led to the development of a fine grained multi-processor UNIX in 1988, three years ahead of Sun and HP.

HCL's R&D was spun off as HCL Technologies in 1997 to mark their advent into the software services arena. During the last eight years, HCL has strengthened its processes and applied its expertise, developed over 30 years into multiple practices - semi-conductor, operating systems, automobile, avionics, bio-medical engineering, wireless, telecom technologies, and many more.

Today, HCL sells more PCs in India than any other brand, runs Northern Ireland's largest BPO operation, and manages the network for Asia's largest stock exchange network apart from designing zero visibility landing systems to land the world's most popular airplane.

Shiv Nadar is the founder of HCL. He founded HCL in 1976 in a Delhi "barsaati". In 1978, HCL developed the first indigenous micro-computer at the same time as Apple and 3 years before IBM's PC. In 1980, HCL introduced bit sliced, 16-bit processor based micro-computer. In 1983, HCL Indigenously developed an RDBMS, a Networking OS and a Client Server architecture, at the same time as global IT peers. In 1986, HCL became the largest IT company in India. In 1988, HCL introduced fine grained multi-processor Unix-3 years ahead of "Sun" and "HP". In 1991, HCL entered into a joint venture Hewlett Packard and HCL-Hewlett Packard Ltd. was formed. The joint developed multi-processor Unix for HP and heralded HCL's entry into contract R&D. In 1997, HCL Infosystems was formed. In the same year HCL ventured into software services. In 1999, HCL Technologies Ltd issued an IPO and became a public listed company. In 2001, HCL BPO was incorporated and HCL Infosystems became the largest hardware company. In 2002, software businesses of HCL Infosystems and HCL Technologies were merged. In 2005, HCL set up first Power PC architecture design centre outside of IBM. In the same year HCL Infosystems launched sub Rs.10,000 PC. In 2006, HCL Infosystems became the first company in India to launch the New Generation of High Performance Server Platforms Powered by.

Major Achievements of HCL

Developed the first indigenous micro-computer in 1978. Indigenously developed an RDBMS, a Networking OS and a Client Server architecture in 1983. In 1986, HCL becomes the largest IT company in India.

Snapshot of HCL

Business Model

The HCL Enterprise comprises two companies listed in India,

1. HCL Technologies2. HCL Info systems

HCL Technologies is the IT and BPO services arm focused on global markets, while

HCL Info systems is the IT, Communication, Office Automation Products & System Integration arm focused on the Indian market.

Together, these entities have uniquely positioned HCL as an enterprise with service offerings spanning the IT Services and Product spectrum.

Business Process Outsourcing

HCL's Business Processing Division (BPO) offers a comprehensive service range – Order to Cash, Procure to Pay, Technical Help Desk, Knowledge Services, Supply Chain Management, Finance and Accounting Services and Customer Life cycle Management.

The BPO focus industry verticals are Telecom, Retail and Media Publishing, Banking and Financial Services, Hi-tech and Manufacturing, Insurance (Life & Non-Life) and Knowledge Process Outsourcing. These verticals have very large number of specific processes that would need specific capability acquisition. The division provides a high quality, economical solution for a wide range of BPO requirements using a tested transition methodology supported by a strong transition team.

OVERVIEW OF TECHNOLOGY DEPARTMENT

Technology has two departments:-

(1) Service support.(2) Service delivery.

(1)Service support-

(a)Networks

(b)Voice

(c)Dialers

(d)Systems

(e)Security

(2)Service Delivery:-Provides solution to the problems.

Ensure IT:- If there is any problem, ensure IT provides solution to it. Call logging is done by the user & solution is provides by technology department.

1. What is Telecommunication?

Telecommunication is the transmission of messages, over significant distances, for the purpose of communication. In modern times, this process almost always involves the sending of electromagnetic waves by electronic transmitters but in earlier years it may have involved the use of smoke signals, drums or semaphore.

Basic Elements

A basic telecommunication system consists of three primary units that are always present in some form:

A transmitter that takes information and converts it to a signal.

A transmission media, also called the "physical channel" that carries the signal. An example of this is the "free space channel".

Receivers that take the signal from the channel and convert it back into usable information.

Making a phone call

What do you do to make a phone call? You pick up the phone, dial some digits, and wait for the person you called to pick up their ringing phone, and then you begin your conversation. To most phone users the details and intricacies of how the phone call occurs are transparent. We have a simplistic view of our phone connection as:

Figure a. Simplistic View of a Phone Call

But typically, our phone system looks more intricate than this, and really looks like this:

A complex network of local, national, and international phone companies and carriers may provide the intermediate devices that handle our phone connection.

2. Primary Rate Interface

The primary Rate Interface (PRI) is a standardized telecommunication service level within the Integrated Services Digital Network (ISDN) specification for carrying multiple DS0voice and data transmissions between a network and a user.

PRI is the standard for providing telecommunication services to offices. It is based on the T carrier (T1) line in the US, and the E-carrier (E1) line in Europe. The T1 line consists of 24 channels, while an E1 has 32 channels.

PRI provides a varying number of channels depending on the standards in the country of implementation. In North America and Japan it consists of 23xB (B channel) and 1xD (D channel) (23 64-kbit/s digital channels + 1 64-kbit/s signaling/control channel) on a T1 (1.544 Mbit/s). In Europe and Australia it is 30xB + 1xD on an E1 2.048 Mbps. One timeslot on the E1 is used for synchronization purposes and is not considered to be a B or D channel.

The Primary Rate Interface consists of 23 B-channels and one 64-kbit/s D-channel using a T1 line or 30 B-channels and one D-channel using an E1 line.

Larger connections are possible using PRI pairing. A dual PRI could have 24+23= 47 B-channels and 1 D-channel but more commonly has 46 B-channels and 2 D-channels thus providing a backup signaling channel. The concept applies to E1s as well and both can include more than 2 PRIs. Normally, no more than 2 D-channels are provisioned as additional PRIs are added to the group.

Application

The Primary Rate Interface channels are typically used by medium to large enterprises with digital PBXs to provide them digital access to the Public Switched Telephone Networks (PSTN). The 23 (or 30) B-channels can be used flexibly and reassigned when necessary to meet special needs such as video conferences. The Primary Rate user is hooked up directly to the telephone company central office

T-carrier systems

The T-carrier system, introduced by the Bell System in the U.S. in the 1960s, was the first successful system that supported digitized voice transmission.

A T1 line in which each channel serves a different application is known as integrated T1 or channelized T1. Another commonly installed service is a fractional T1, which is the rental of some portion of the 24 channels in a T1 line, with the other channels going unused.

In the T1 system, voice or other analog signals are sampled 8,000 times a second and each sample is digitized into an 8-bit word. With 24 channels being digitized at the same time, a 192-bit frame (24 channels each with an 8-bit word) is thus being transmitted 8,000 times a second. Each frame is separated from the next by a single bit, making a 193-bit block. The 192 bit frame multiplied by 8,000 and the additional 8,000 framing bits make up the T1's 1.544 Mbps data rate. The signaling bits are the least significant bits in each frame.

Chart 1 - T1 Hierarchy

The fundamental frame of T1 is shown in Figure b.

DS0 64Kbps 1/24 of T-1 1 Channel

DS1 1.544Mbps 1 T-1 24 Channels

DS1C 3.152 Mbps 2 T-1 48 Channels

DS2 6.312 Mbps 4 T-1 96 Channels

DS3 44.736 Mbps 28 T-1 672 Channels

DS3C 89.472 Mbps 56 T-1 1344 Channels

DS4 274.176 Mbps 168 T-1 4032 Channels

Figure b.

E-Carrier Systems

E-carrier system, which is revised and improved version of the earlier American T-carrier technology. Now it is widely used in almost all countries outside USA, Canada and Japan. The line data rate for E1 is 2.048 Mbit/s which is split into 32 time slots, each being allocated 8 bits in turn. It is an ideal for voice traffic because voice is sampled at the same 8 kHz rate so E1 line can carry 32 simultaneous voice conversions.

CalculationWe often hear of T1 speed as 1.544 Mbits/second or 1,544,000 bits/second. This is determined

by:

Sampling at 8000 times/second x each sample is 8 bits =

64000 bits/second x 24 timeslots per frame

___________________________________

1,536,000 bits/second

+ 8000 bits/second of framing bits

____________________________

1,544,000 bits/second (1.54 Mbps)

D Channel D channel is a telecommunication term which refers to the ISDN channel in which the control and signaling information is carried.

The bit rate of the D channel of a basic rate interface is 16 Kbit/s, whereas it amounts to 64 Kbit/s on a primary rate interface.

B Channel

B channel is a telecommunication term which refers to the ISDN channel in which the primary data or voice communication is carried. It has a bit rate of 64 Kbit/s in full duplex.

The term is applied primarily in relation to the ISDN access interfaces, since deeper in the PSTN network an ISDN bearer channel is essentially indistinguishable from any other bearer channel

Signaling

In telecommunication, signaling has the following meanings:

a. The use of signals for controlling communications.b. The information exchange concerning the establishment and control of a

telecommunication circuit and the management of the network, in contrast to user information transfer.

c. The sending of a signal from the transmitting end of a telecommunication circuit to inform a user at the receiving end that a messages to be sent.

The signaling is divided into two main parts named as:

I. Channel Associated Signaling(CAS)II. Common Channel Signaling(CCS)

Channel Associated Signaling

Channel Associated Signaling (CAS), also known as per-trunk signaling (PTS), is a form of digital communication signaling. As with most telecommunication signaling methods, it uses routing information to direct the payload of voice or data to its destination. With CAS signaling, this routing information is encoded and transmitted in the same channel as the payload itself. This information can be transmitted in the same band (in-band signaling) or a separate band (out-of-band signaling) to the payload.

CAS potentially results in lower available bandwidth for the payload. For example, in the PSTN the use of out-of-band signaling within a fixed bandwidth reduces a 64 Kbit/s DS0 to 56 Kbit/s.

Various types of CAS signaling are available in the T1 world. The most common forms of CAS signaling are loop start, ground start, and E&M signaling. The biggest disadvantage of CAS signaling is that the network uses bits from information IP packets, such as voice packets, to perform signaling functions.

The most common implementation of CAS is robbed bit signaling.

Common Channel Signaling

Common channel signaling (CCS) is signaling (for example, in a T-carrier system line) in which a group of voice-and-data channels share a separate channel that is used only for control signals.

CCS offers the following advantages over CAS:-

a. Faster call setup. b. No interference between signaling tones by network and frequency of human speech pattern.

c. Greater trunking efficiency due to the quicker set up and tears down, thereby reducing traffic on the network.

d. No security issues related to the use of in-band signaling with CAS.

e. CCS allows the transfer of additional information along with the signaling traffic providing features such as caller ID.

f. The most common CCS signaling methods in use today are ISDN and SS7.

3. Anatomy of an Office Phone Call

When a phone call is made between two corporate phone users Mr. Naveen (Ext-3605) and Mr. Nakul (Ext-3690), for example the call is made by Mr. Nakul to Mr. Naveen, and then the phone connects directly to the PBX with a dedicated pair of wires. This pair of wires is called the subscriber or local loop. One wire is called the Tip and the other wire, the Ring. The figure is shown as below:

Let us see what happens when Mr. Nakul picks up his phone to call Mr. Naveen.

Onhook

Before Mr. Nakul or Mr. Naveen picks up their phone, both phones are onhook, which means that the phone handset is not lifted off the phone. The PBX provides power and potential across each subscriber loop to monitor the activity and power of each phone. When the phone is onhook, there is no current flow through the subscriber loop. See the figure:-

Offhook

When Mr. Nakul lifts the phone handset, the phone is now offhook. A switch hook within the phone closes and current flows through the subscriber loop. The current flow tells the PBX that Mr. Nakul wishes to place a call. See the figure:-

Dial Tone

At this point, the PBX:

• Searches for an unused dial register to store the dialed phone number digits.

• Sends a dial tone through the subscriber loop to Mr. Nakul’ phone. Once Mr. Nakul hears the dial tone he can begin to dials the digits, 3605. These digits are sent over the subscriber loop to the PBX dial register. (See the Figure below)

Ringing Voltage

The PBX looks in its routing table and recognizes that extensions 3605 is a local number and exist on another subscriber loop. The PBX sends a ringing voltage across the subscriber loop to ring the bell within 3605 phone. (See the figure below)

Call Completion

Once Mr. Naveen lifts up the phone handset, current flows through the subscriber loop and the circuit is complete for the call. Mr. Nakul and Mrs. Naveen can begin talking and their speech is carried over the subscriber loop as an electric signal. (See the figure below)

Components of the Phone System

The Main Components of the phone System are:-

• The phone

• The PBX

• The subscriber loop

• The trunk lines

The Phone

The phone is an analog device that carries our speech as an electric signal. To make this occur, all phones have the basic components shown in Figure

HandsetThe handset contains a receiver, transmitter, and hybrid.

Receiver and TransmitterThe handset houses the earpiece or receiver and the mouthpiece or transmitter.We speak into the mouthpiece and our speech is transmitted over a pair of wires.

We listen through the earpiece or receiver and receive sound over another pair of wires. In total, four wires make up the handset.

HybridThe handset also contains a device known as the hybrid. As we learned earlier, the phone connects to the PBX using a dedicated pair of wires, the subscriber loop. The phone receiver/transmitter, however, has four wires. In order to interface between the receiver/transmitter that has four wires and the PBX which uses two-wire, a hybrid is needed. Figure illustrates an example of a hybrid.

Switch Hook

The switch hook is located directly below the handset. When you lift the handset, the switch hook closes and current flows through the phone. The phone is offhook. The PBX supplies power to operate the phone. When you replace the handset, the switch hook opens and current ceases to flow through the phone. The phone is onhook.

Ringer

When a PBX wants to alert a remote phone of an inbound call, it rings the remote phone by sending ringing voltage down the subscriber loop. The ringing voltage causes an armature within the phone to pivot. The armature in turn drives a hammer against a bell, which causes ringing.

The PBX

A private branch exchange (PBX) is a telephone exchange that serves a particular business or office, as opposed to one that a common carrier or telephone company operates for many businesses or for the general public. PBXs are also referred to as:

PABX - private automatic branch exchange

EPABX - electronic private automatic branch exchange

PBX functions

Functionally, the PBX performs four main call processing duties:

Establishing connections (circuits) between the telephone sets of two users (e.g. mapping a dialed number to a physical phone, ensuring the phone isn't already busy)

Maintaining such connections as long as the users require them (i.e. channeling voice signals between the users)

disconnecting those connections as per the user's requirement Providing information for accounting purposes (e.g. metering calls)

In addition to these basic functions, PBXs offer many other calling features and capabilities, with different manufacturers providing different features in an effort to differentiate their products. Common capabilities include (manufacturers may have a different name for each capability):

Auto attendant Auto Dialing Automatic call distributor Automated directory services (where callers can be routed to a given employee by keying or

speaking the letters of the employee's name) Automatic ring back

Call accounting Call Blocking Call forwarding on busy or absence Call park Call pick-up Call transfer Call waiting Camp-on Conference call Custom greetings Customized Abbreviated dialing Busy Override Direct Inward Dialing Direct Inward System Access (DISA) (the ability to access internal features from an outside

telephone line) Do not disturb (DND)

Follow-me, also known as find-me: Determines the routing of incoming calls. The exchange is configured with a list of numbers for a person. When a call is received for that person, the exchange routes it to each number on the list in turn until either the call is answered or the list is exhausted (at which point the call may be routed to a voice mail system).

Interactive voice response Music on hold Night service Shared message boxes (where a department can have a shared voicemail box) Voice mail Voice message broadcasting Voice paging (PA system) Welcome Message

Subscriber Loop

The subscriber loop is a dedicated pair of wires that connect the phone to the PBX. Each phone connected to the PBX has its own subscriber loop.

The subscriber loop consists of two wires known as, Tip and Ring. As shown in

Figure, the Ring lead connects to the negative side of the battery; the Tip lead connects to ground. When the circuit is complete, current flows. The current flow is shown by the dashed loop.

Trunk Line

Trunk lines connect one PBX to another PBX, or a trunk line can connect the PBX to the outside world, the public phone network. There can be many trunk lines between two PBXs and these trunk lines are usually shared. The number of trunk lines available will depend on the number of phones connected to each PBX. Trunk lines are shared because it is assumed that the phone users will not all simultaneously try to make a call at the same time. This means that when a call is made, the PBX seizes and uses one trunk line. Upon completion of the call, the PBX releases the trunk line and it is available for use by another call.

Types of trunk lines

There are two types of trunk lines, two-wire and four-wire.

Two-wire Circuits

Two-wire trunk lines are usually used to connect PBXs at distances of up to several thousand feet. The exact distance between two PBXs will depend on the thickness or gauge of the wire used. The thicker the gauge, the longer the distance. However, as distance increases the signal quality decreases until the receiver cannot recognize the signal. In this situation, amplifiers are needed to amplify the signal. Since amplifiers work in only one direction, the voice is separated into different paths: one for transmit and one for receive.

Four-Wire Circuits

Four wire trunk lines are used for most high-volume, long distance lines. A four-wire phone line circuit uses two wires for the transmit path and two wires for the receive path. Voice and signals are transmitted on the Tip and Ring and received on the Tip1 and Ring1.

If a PBX connects to a four-wire circuit, a hybrid is needed. The phone connects to

The PBX over the two-wire subscriber loop and in order to interface with the four wire trunk line, the PBX uses a hybrid to provide the conversion as shown in Figure.

Some of the common trunk lines are:

Tie TrunksTie trunks connect one PBX to another PBX. Tie trunk are either two-wire or four-wire.

Central Office (CO) TrunkA CO trunk connects the PBX to a local Central Office (CO). The local CO is part of the local phone company. This type of trunk line allows calls to go from a private network to a public network.

Foreign Exchange Station (FXS) TrunkA typical FXS trunk consists of a subscriber trunk connected directly to a distant CO or a PBX. This service can be leased to avoid long distance charges to a distant CO. The subscriber dials a local exchange number. FXS trunks provide the convenience of a seven-digit number plus service to a distant location at a reduced cost.

Direct Inward Dialing (DID)

Direct Inward Dialing (DID) trunks allow a caller to dial into a PBX directly to a phone or a group of phones without operator intervention. DID typically use two-wire trunks. In DID, an outside caller dials the number of the desired phone extension, which the connecting CO passes to the PBX.

Direct Outward Dialing (DOD)With DOD, the extension phone user automatically accesses the local CO without operator intervention. You typically dial “9” and then the outside number.

Wide Area Phone Services (WATS)

Inward WATS and Outward WATS let users either receive or originate long distance calls and have them billed at a bulk rate rather than individually. Inward WATS calls are billed to the called number; outward WATS is billed to the calling party.

Signaling

Signaling is important because it is how phone system components communicate and exchange information.This is a signal to the phone and the user that the PBX is ready to receivedialed numbers. Dialing numbers on a phone keypad is another example ofsignaling.There are three signaling types and each provides particular information about a voice call:

Methods of Address SignalingThere are two types of address signaling methods:• Dial pulse. This is associated with rotary dial phones.• Dual Tone Multifrequency (DTMF) signaling, most often supported on a pushbutton phone.

Informational SignalingInformational signals are generated by the PBX or switch to tell the user about thecall’s progress. There are different types of informational signals; the common typesinclude:• busy signal• fast busy• dial tone• ring back

Supervisory SignalingSupervisory signaling monitors the status of a line or trunk, which is either idle

Signaling Types Function

Supervisory Supervisory signaling provides on the subscriber loop and trunk status.

Address Address signaling is how the phone system directs or routes a call to the call destination.

Informational Informational signaling tells the phone user or subscribers about call progress.

(onhook) or active (offhook). Supervisory signaling types include the following:• Loopstart• Ground Start• E&M

4. The Telephone Set

TDM phones

VoIP phones

TDM Phones

Time-division multiplexing (TDM) is a type of digital or (rarely) analog multiplexing in which two or more signals or bit streams are transferred apparently simultaneously as sub-channels in one communication channel, but are physically taking turns on the channel. The time domain is divided into several recurrent timeslots of fixed length, one for each sub-channel. A sample byte or data block of sub-channel 1 is transmitted during timeslot 1, sub-channel 2 during timeslot 2, etc. One TDM frame consists of one timeslot per sub-channel. After the last sub-channel the cycle starts all over again with a new frame, starting with the second sample, byte or data block from sub-channel 1, etc.

VoIP phones

Voice over IP (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), 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 (PSTN). 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 (IP) packets for transmission over the Internet; the process is reversed at the receiving end.[1]

VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. Codec use is varied between different implementations of VoIP (and often a range of codecs are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs.

Benefits

Operational cost

VoIP can be a benefit for reducing communication and infrastructure costs. Examples include:

Routing phone calls over existing data networks to avoid the need for separate voice and data networks.[24]

Conference calling, IVR, call forwarding, automatic redial, and caller ID features that traditional telecommunication companies (telcos) normally charge extra for are available free of charge from open source VoIP implementations.

Costs are lower, mainly because of the way Internet access is billed compared to regular telephone calls. While regular telephone calls are billed by the minute or second, VoIP calls are billed per megabyte (MB). In other words, VoIP calls are billed per amount of information (data) sent over the Internet and not according to the time connected to the telephone network. In practice the amount charged for the data transferred in a given period is far less than that charged for the amount of time connected on a regular telephone line.

Flexibility

VoIP can facilitate tasks and provide services that may be more difficult to implement using the PSTN. Examples include:

The ability to transmit more than one telephone call over a single broadband connection without the need to add extra lines.

Secure calls using standardized protocols (such as Secure Real-time Transport Protocol). Most of the difficulties of creating a secure telephone connection over traditional phone lines, such as digitizing and digital transmission, are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream.[26]

Location independence. Only a sufficiently fast and stable Internet connection is needed to get a connection from anywhere to a VoIP provider.

Integration with other services available over the Internet, including video conversation, message or data file exchange during the conversation, audio conferencing, managing address books, and passing information about whether other people are available to interested parties.

5. System architecture

Meridian

Nortel Meridian is a private branch exchange. It provides advanced voice features, data connectivity, LAN communications, computer telephony integration (CTI), and information services for communication applications ranging from 60 to 80,000 lines.

The Meridian has 43 Million installed users worldwide, making it the most widely used PBX.

The Meridian is one of the few PBX's still available from a major communications supplier that can be configured as non-VOIP PBX.

Hardware architecture

A Meridian 1 is a circuit-switched digital system that provides voice and data transmission. The internal hardware is divided into the following functional areas (see Figure 11):

• Common equipment circuit cards provide the processor control, software execution, and memory functions of the system.

• Network interface circuit cards perform switching functions between the processor and peripheral equipment cards.

• Peripheral equipment circuit cards provide the interface between the network and connected devices, including terminal equipment and trunks.

• Terminal equipment includes telephones and attendant consoles (and may include equipment such as data terminals, printers, and modems).

• Power equipment provides the electrical voltages required for system operation, and cooling and sensor equipment for system protection.

The Meridian 1 product line consists of system types referred to as system Options. A system option is made up of Universal Equipment Modules (UEMs) stacked one on top of another to form a column. Each column contains a pedestal, a top cap, and up to four modules. A system can have one column or multiple columns.

Each UEM is a self-contained unit that, when equipped, houses a card cage and backplane, power and ground cabling, power units, I/O panels, circuit cards, and cables. When the card cage is installed, the function of the UEM is established and the module is no longer “universal.” Meridian 1 module is as follows:

• NT5K11 Enhanced Existing Peripheral Equipment Module for Options51C, 61C, and 81C.

• NT4N41DA PCI® Core/Network Module for Option 81C

• NT5D21 Core/Network Module for Options 51C, and 61C

• NT8D35 Network Module required for Options 51C, 61C, and 81C

• NT8D37 Intelligent Peripheral Equipment (IPE) Module required for Options 51C, 1C, and 81C

• NT8D47 Remote Peripheral Equipment (RPE) Module optional for Options 51C, 61C, and 81C

MERIDIAN 1 PBX SYSTEM OPTIONS

This document includes information on the following Meridian 1 PBX system types :

• Option 51C: enhanced common control complex, single CPU, and half-network group

• Option 61C: enhanced common control complex, dual CPU, and one full-network group

• Option 81C: enhanced common control complex, dual CPU, and multiple-network groups.

All system Options are available in AC- and DC-powered versions.

System Option 51C

Option 51C is a single-CPU system with a half-network group. One

Core/Network Module and one IPE Module are required. Additional IPE

Modules, PE Modules, RPE Modules, and application modules can be used.

System Option 61C

Option 61C is a dual-CPU system with standby processing capability, fully redundant memory, and a full-network group. Two Core/Network Modules and one IPE Module are required. Additional IPE Modules, PE Modules, RPE Modules, and application modules can be used.

System Option 81C

Option 81C is a dual-CPU system with standby processing capabilities, fully redundant memory, and up to eight full-network groups. Option 81C is equipped with two redundant input/output processor and disk drive unit combination packs.

The following modules are required:

• Two PCI Core/Network Modules (provides one network group)

• A minimum of two Network Modules (provides one network group)

• A minimum of one IPE Module

Additional Network and IPE Modules are required for additional network groups. PE Modules, RPE Modules, or application modules can also be used.

Table 3 lists the specifications for Option 81C. Figure 1 shows a typical configuration for eight full network groups. Additional columns can be added, and there can be more than one row of columns.

Figure Meridian 1 PBX 81C CP PIV or Option 81C

FigureCS 1000M MG

6. System modules

Each type of module is available in AC-powered and DC-powered versions (except theNT8D36 InterGroup module that does not require power). AC-power modules generally require a module power distribution unit (MPDU) to provide circuit breakers for the power supplies. DC-powered modules do not require an MPDU because a switch on each power supply performs the same function as the MPDU circuit breakers..NT4N41 Core/Network moduleThis module provides common control and network interface functions. With the CS 1000M MG and the Meridian 1 PBX 81C CP PIV, two Core/Net modules are installed side-by-side. With the CS 1000M SG and the Meridian 1 PBX 61C CP PIV, the modules are stacked or mounted side-by-side.

One section of this module houses the common control complex (CPU, memory, up to four cCNI cards, and mass storage functions). The other section supports a Conference card, one Peripheral Signaling card, one 3-Port Extender card, and optional network cards.

Each Core/Network module houses up to four NT8D04 Superloop Network Cards for a total of 16 network loops. Superloop Network cards are cabled to the backplane of an IPE module. In a typical configuration, one conference/TDS card is configured in the module, leaving 14 voice/data loops available. Core sideThe Core side of the module contains the circuit cards that process calls, manage network resources, store system memory, maintain the user database, and monitor the system. These circuit cards also provide administration interfaces through a terminal, modem, or enterprise IP network.

The Core side runs in redundant mode: one Core operates the system while the other runs diagnostic checks and remains ready to take over if the active Core fails. Both Cores are connected to each Network group depending on hardware configuration. If one Core fails, the second Core immediately takes over call processing. The Core shelf backplane is a compact PCI data bus. Network sideThe Network side of this module contains the cards for half of the Network group 0. The other half of Network group 0 resides in the second core network module.

The CS 1000M MG and Meridian 1 PBX 81C CP PIV support a Fiber Network Fabric network system with a FIJI card in slots 8 and 9 on the Net side of the Core/Net module.

FigureNT4N41 cPCI Core/Network module

NT8D35 Network moduleThis module provides the network switching functions in the Meridian 1 Option 81C, Meridian 1 PBX 81C CP PIV, and CS 1000M MG. Two Network modules are required to make a full network group of 32 loops. A maximum of 16 Network modules (eight network groups) can be configured in the Meridian 1 Option 81C, CS 1000M MG, and Meridian 1 PBX 81C CP PIV.

The Network module houses up to four NT8D04 Superloop Network Cards, for a total of 16 network loops. Superloop network cards are cabled to the backplane of an IPE module. In a typical configuration, one Conference/TDS card is configured in the module, leaving 14 voice/data loops available. In CS 1000M MG and Meridian 1 PBX 81C CP PIV, the Conference/TDS cards are located in the Core/Network module. The Clock Controller must be installed in slot 13.

FigureNT8D35 Network module

The Network module can be used as a PRI/DTI expansion module. The number of PRI/DTI expansion modules that can be used is determined by traffic considerations.

Figure NT8D35 Network module configured for PRI/DTI expansion

NT8D37 Intelligent Peripheral Equipment moduleThe Intelligent Peripheral Equipment (IPE) module provides the interface between network switching and IPE cards, such as intelligent line and trunk cards, in all Large Systems.

The IPE module houses one NT8D01 Controller Card, which is the peripheral equipment controller, and up to 16 IPE cards, supporting up to 512 terminal numbers (256 voice and 256 data). The controller card is cabled to the NT8D04 Superloop Network Card.

Figure NT8D37 IPE module

Fiber Remote IPE moduleThis module provides fiber-optic links between the network functions in a Large System and the peripheral controller functions in the Fiber Remote IPE. A floor-standing column or wall-mounted cabinet is installed at the remote site and is connected to the Large System using fiber-optic links. The Fiber Remote IPE provides Large Systems functionality with the installation of only IPE modules and IPE cards at a distant site. Since the remote IPE system uses the common equipment and network equipment of the associated local Large System, it can deliver the same features and functionality as the local system. Large Systems can be configured in a distributed system to support remote subscribers, using Remote IPE modules or small cabinets. Fiber-optic links are used to connect the Remote IPE modules and small cabinets to the PBXs.

FigureLarge System to Remote IPE site

Carrier Remote IPEThe Carrier Remote IPE provides functionality by installing only IPE modules and IPE cards at a distant site. The Remote IPE shares the system common and network equipment to provide the same functions and features to remote subscribers that are available to local system subscribers.FigureMeridian 1 Large System to Carrier Remote IPE links

Terminal equipment

Large Systems support a wide range of telephones, including multiple-line and single-line telephones, as well as digital telephones with key and display functions and data transmission capabilities. A range of options for attendant call processing and message center applications is also available. In addition, a number of add-on devices are available to extend and enhancethe features of telephones and consoles. Add-on devices include key/lamp modules, lamp field arrays, handsets, and hands free units.

Signaling ServerCS 1000M systems use a Signaling Server. The Signaling Server is an PC-based server that provides a central processor to drive H.323 and Session Initiation Protocol (SIP) signaling for IP Phones and IP Peer Networking. It provides signaling interfaces to the IP network using software components that operate on the VxWorksª real-time operating system. The legacy Nortel ISP1100 Signaling Server can still be used. CS 1000 Release 5.5 introduces three new servers that can host a CS 1000 Release 5.5 Signaling Server:• "Nortel Common Processor Pentium Mobile server"• "International Business Machines X306m server"• "Hewlett Packard DL320-G4 server"

The Signaling Server has both an ELAN and TLAN network interface. The Signaling Server communicates with the Call Server through an ELAN subnet.

The Signaling Server is mounted in a 19-inch rack. The Signaling Server can be installed in a load-sharing redundant configuration for higher scalability and reliability.

The following software components operate on the Signaling Server:• "Terminal Proxy Server” (TPS)• "SIP/H.323 Signaling Gateways"• "Network Routing Service" (NRS)• "Element Manager"• Application Server

All the software elements can coexist on one Signaling Server or reside individually on separate Signaling Servers, depending on traffic and redundancy requirements for each element.

To check no. of ip phones registerd on Signaling server

oam> isetShow*It will show the total no. of phone registered

Set Information--------------- IP Address NAT Model Name Type RegType State Up Time Set-TN Regd-TN HWID FWVsn UNIStimVsn SrcPort DstPort RFC2833PTTx------------------ ---- -------------------------------- ---------- ------- ------------ -------------- ------------ ------------ -------------------- ------- ---------- ------- ------- ------------

To check no. of VGMC(Voice Gateway Media Card) cards in Use

oam> electShowNode ID : 151Node Master : Yes

Up Time : 1 days, 2 hours, 15 mins, 46 secs TN : 000 00 00 00Server Type : Signaling ServerPlatform Type : ISP1100TLAN IP Addr : ELAN IP Addr : Election Duration : 15Wait for Result time : 31Master Broadcast period : 30===== Node Master =====Server Type Platform TN TLAN IP AddrSignaling Server ISP1100 000 00 00 00 Next timeout : 25 secAutoAnnounce : 1Timer duration : 60 (Next timeout in 0 sec)====== all tps ======Num Server Type Platform ELAN MAC TLAN IP ELAN IP001 Signaling Server ISP1100 TN = 000 00 00 00 UpTime = 001 02:15:46 NumOfSets = 381 NumOfCensusTimeout = 0

====== Cards in node configuration that are not registered ======Num TN Server Type ELAN MAC TLAN IP Addr ELAN IP Addr

To check the signaling server status and uptime

oam> pbxLinkShowActive Call Server type = CS 1000M HG/SG/MGActive Call Server S/W Release = 500WSupported Features: CorpDir UserKeyLabel VirtualOffice UseCSPwd 2001P2 2004P2 2002P2 PD/RL/CL QoS Monitoring NAT Traversal ACF IP ACD 1150 NextGen Phones Call Server Main: ip =X, ConnectID = 0x2f0c5678, BroadcastID = 0x2f0c5578, Link is upCall Server Redundant: ip =X, ConnectID = 0x2f0c5778, BroadcastID = 0x0, Link is in TBD stateCall Server Signaling Port = 15000Call Server Broadcast Port = 15001Broadcast PortID = 0x2e8eeae0RUDP portID = 0x2e8eeb40Tcp Link state = upTcp Signaling Port: 15000Tcp socket fd: 27Tcp msgs sent: 471810Tcp msgs recd: 3574341oam> uptime03:39 PM up 1 day(s), 02:15oam> tpsShowNode ID : 151Is master : 1

Up time : 1 days, 2 hours, 16 mins, 12 secs (94572 secs)Server Type : Signaling ServerPlatform : ISP1100TPS Service : YesIP TLAN : IP ELAN : ELAN Link : UpSets Connected: 381Sets Reserved : 0

oam> cslogin SEC054 A device has connected to, or disconnected from, a pseudo tty without authenticatingloii admin1

PASS?SEC0029 SECURITY WARNING: THIS SYSTEM CONTAINS INSECURE PASSWORDS, NOTIFY YOUR SYSTEM ADMINISTRATORTTY #15 LOGGED IN ADMIN1 15:39 13/6/2010 >****OVL000 >OVL000 >****OVL000 >

Terminal Proxy ServerThe Terminal Proxy Server (TPS) acts as a signaling gateway between theIP Phones and the Call Server using the UNIStim protocol. It performs the following functions:• converts the IP Phone UNIStim messages into messages the Call Server can interpret.• allows IP Phones to access telephony features provided by the CallServer.

SIP/H.323 Signaling GatewaysSIP/H.323 Signaling Gateways are software components configured on virtual loops, similar to IP Phones. They bridge existing call processing features and the IP network. They also enable access to the routing and features in the MCDN feature set.To support IP Peer Networking, dual Call Servers in a CS 1000M must be associated with Signaling Servers that run SIP/H.323 Signaling Gateway software. The number of Signaling Servers required depends on thecapacity and level of redundancy required.

Network Routing ServiceNRS for CS 1000 Release 5.5 software is offered in two versions: a SIP Redirect Server NRS and a SIP Proxy NRS. The SIP Redirect Server NRS is hosted either co-resident with Signaling Server applications, or in a stand-alone mode on a dedicated Common Processor Pentium Mobile (CP PM) server running the VxWorks™ real-time operating system. There are no changes to the SIP Redirect Server NRS in CS 1000 Release 5.5.

The SIP Proxy NRS is hosted in a stand-alone mode on a dedicated commercial off the shelf server running the Linux™ real-time operating system. The SIP Proxy NRS is referred to as the Linux-based NRS.

The NRS application provides network-based routing, combining the following into a single application:

• H.323 Gatekeeper — The H.323 Gatekeeper provides central dialing plan management and routing for H.323-based endpoints and gateways.

• SIP Redirect Server—The SIP Redirect Server provides central dialing plan management and routing for SIP-based endpoints and gateways.

• NRS Database — The NRS database stores the central dialing plan in XML format for both the SIP Redirect Server and the H.323 Gatekeeper. The SIP Redirect Server and H.323 Gatekeeper both access this common endpoint and gateway database.

• Network Connect Server (NCS) — The NCS is used only for Virtual Office, Branch Office, and Geographic Redundancy solutions.

• NRS Manager web interface — The NRS provides its own web interface to configure the SIP Redirect Server, the H.323 Gatekeeper, and the NCS.

The NRS application provides routing services to both H.323 and SIP-compliant devices. The H.323 Gatekeeper can be configured to support H.323 routing services, while the SIP Redirect Server can be configured to support SIP routing services.

The H.323 Gatekeeper and the SIP Redirect Server can reside on the same Signaling Server. Examples of H.323 and SIP-compatible endpoints needing the services of the NRS are CS 1000E. The NRS also supports endpoints that do not support H.323 Registration, Admission, and Status (RAS) or SIP registration with the NRS. Gatekeeper procedures are referred to as non-RAS or static endpoints. Each CS 1000E in an IP Peer network must register to the NRS. The NRS software identifies the IP addresses of PBXs based on the network-wide numbering plan. NRS registration eliminates the need for manual configuration of IP addresses and numbering plan information at every site.

7. Load directory

What are Overlays?

•Prompt – Response systems used to configure M1 switch.

•Used to Create, Modify & Delete configurations

•Are identified by “LD xxx” where “xxx” are numbers.

CLASSIFICATION OF OVERLAYS

•Administration Overlays

•Maintenance Overlays

•Print Overlays

ADMINISTRATION OVERLAYS

Configuration Record (LD 17, LD 97)

Hardware of the System

Loops

To peripheral equipment (ENET) Conference and Tone loops (XCT) T1/E1 (DTI, PRI)

Super loops to IPE shelf – IPE segmentation

TTY’s, Number of CPUs

Background and Midnight routines

Call Processing Data structure allocation

Digitone Receivers (LD 13)

CDB (CUSTOMER DATA BLOCK) – (LD 15)

Characteristics applying to all customer resources are defined in this data block

ROUTES (LD 16)

Logically groups multiple trunks coming\going to a common destination Each route is assigned an Access code (ACOD) which determine which calls it will handle

Trunks (LD 14)

Links trunks with routes Has a physical location called a Terminal Number (TN)

- TN consists of loop, shelf, card, unit (lscu)

Digital Sets (LD 11)

M39XX sets

Analog Sets (LD 10)

500, 2500 sets

Attendant Console (LD 12)

2250 sets

MAINTENANCE OVERLAYS

LD 30 – Network and signaling Diagnostics

LD 31 – Telephone and Console Diagnostics

LD 32 – Network and peripheral Diagnostics

LD 34 – TDS and DTR Diagnostics

1.1 LD 2 – Traffic

Set time and date Set traffics report schedules, print reports

LD 1 – Audit (template audit)

Audits the PBX and BCS template data structures

1.2 LD 44 – Software Audit

Audits data structures: Call registers, queues, etc Checks/Repairs network connections, linkages

PRINT OVERLAYS

LD 20 – Prints TN related data (sets, trunks, DTRs, Dn blocks)

LD 21 – Prints Customer Data Block

LD 22 – Configuration Record Data Block

LD 81 – Print a list or count of sets with selected features

LD 82 – Prints hunting patterns and DN blocks

LD 83 – Prints a list of TNs in DES order

LD 97 – Administrates and Print some portions of Configuration record

To show the loops on D channel

ld 96

DCH000 .stat dch

DCH 002 : OPER EST ACTV AUTO DES: XYZ DCH 003 : OPER EST ACTV AUTO DES : XYZ

DCH 005 : OPER EST ACTV AUTO DES : XYZ

DCH 009 : OPER EST ACTV AUTO DES : XYZ

DCH 010 : OPER EST ACTV AUTO DES : XYZDCH 011 : OPER EST ACTV AUTO DES : XYZDCH 018 : OPER EST ACTV AUTO DES : XYZ

DCH000 .stat dch 10

DCH 010 : OPER EST ACTV AUTO.****

>OVL000 > To check the B channel status

>ld 60

DTI000 SL-1 --- SYS-12/AXE-10 SWE/NUMERIS/SWISS/TCNZ /EUROISDN/SING/THAI/MSIA/INDO/CHNA /INDI/PHLP CHANNEL TIMESLOT MAPPING

. SL-1 NETWORK TIMESLOT B-CHANNEL 1 -15 1 -15 1 -15 16-30 17-31 17-31D-CHANNEL 31 16 16

DTI000 .stat 10

PRI2 LOOP 10 - ENBL REF CLK: DSBLSERVICE RESTORE: YES ALARM STATUS: ACCEPTABLECH 01 - IDLE TIE VCE * CH 02 - IDLE TIE VCE * CH 03 - IDLE TIE VCE * CH 04 - IDLE TIE VCE * CH 05 - IDLE TIE VCE * CH 06 - IDLE TIE VCE * CH 07 - BUSY TIE VCE * CH 08 - BUSY TIE VCE * CH 09 - BUSY TIE VCE * CH 10 - BUSY TIE VCE * CH 11 - IDLE TIE VCE * CH 12 - IDLE TIE VCE * CH 13 - IDLE TIE VCE * CH 14 - IDLE TIE VCE * CH 15 - IDLE TIE VCE * CH 16 - IDLE TIE VCE * CH 17 - IDLE TIE VCE * CH 18 - IDLE TIE VCE * CH 19 - IDLE TIE VCE * CH 20 - BUSY TIE VCE * CH 21 - BUSY TIE VCE * CH 22 - BUSY TIE VCE * CH 23 - IDLE TIE VCE * CH 24 - IDLE TIE VCE * CH 25 - IDLE TIE VCE * CH 26 - IDLE TIE VCE * CH 27 - IDLE TIE VCE * CH 28 - IDLE TIE VCE * CH 29 - IDLE TIE VCE * CH 30 - IDLE TIE VCE * CH 31 - DCH 10 .ERR049 36 0 7 1

ERR049 28 0 13 3 ****

>OVL000

To check the system health

>ld 135

CCED000.stat cpu

cp 1 16 PASS -- ENBL

SYSTEM STATE = REDUNDANTDISK STATE = REDUNDANTHEALTH = 30VERSION = May 23 2007, 00:01:30Side = 1, DRAM SIZE = 512 MBytescp 0 16 PASS -- STDBYSYSTEM STATE = REDUNDANTDISK STATE = REDUNDANTHEALTH = 30VERSION = May 23 2007, 00:01:30Side = 0, DRAM SIZE = 512 MBytesDTA301 18 CCED000.stat cni

cni 1 9 0: remote = group 0 ENBLcni 1 9 1: remote = group 1 ENBLcni 1 10 0: remote = group 2 ENBL

cni 0 9 0: remote = group 0 ENBLcni 0 9 1: remote = group 1 ENBLcni 0 10 0: remote = group 2 ENBLCCED000.stat memSide = 1, DRAM SIZE = 512 MBytesSide = 0, DRAM SIZE = 512 MBytesERR049 84 0 4 1 ****>

>OVL000 >

To check the configuration of all super loops

>ld 97

SCSYS000 MEM AVAIL: (U/P): 42893194 USED U P: 8255342 464133 TOT: 51612669 DISK SPACE NEEDED: 776 KBYTESREQ prt

TYPE supl

SUPL TIM185 15:00 13/6/2010 CPU 1

SUPL SUPT SLOT XPEC0 XPEC1 SHLF ZONE0/1 IPR0/1

012 STD LEFT 04 0 1 -- - - --- --- - --- --------------- 028 STD LEFT 04 2 3 -- - - --- --- - --- --------------- 036 STD RGHT 05 0 1 -- - - --- --- - --- --------------- 040 STD RGHT 05 2 3 -- - - --- --- - --- --------------- 052 STD RGHT 01 0 1 -- - - --- --- - --- --------------- 056 STD RGHT 01 2 3 -- - - --- --- - --- --------------- 068 STD RGHT 03 0 1 -- - - --- --- - --- --------------- 072 STD RGHT 03 2 3 -- - - --- --- - --- --------------- 084 STD RGHT 02 0 1 -- - - --- --- - --- --------------- 088 STD RGHT 02 2 3 -- - - --- --- - --- --------------- 152 ---- ---- VIRTUAL -- - - --- --- - --- --------------- 156 ---- ---- VIRTUAL -- - - --- --- - --- --------------- 184 ---- ---- PHANTOM -- - - --- --- - --- --------------- 188 ---- ---- PHANTOM -- - - --- --- - --- ---------------

REQ prg

SCH0101 REQ prt

TYPE supl

SUPL 12

SUPL SUPT SLOT XPEC0 XPEC1 SHLF ZONE0/1 IPR0/1

012 STD LEFT 04 0 1 -- - - --- --- - --- ---------------

REQ ERR049 28 0 13 7 ****DTA301 5

OVL000 >

To check the FIJI card status

>ld 39

ISR000 .stat ring ERR049 72 0 14 14 0

RING STATE: DRIVES HALF (000 - 479)RING AUTO RECOVERY IS ONFIJI 0 0 ENBL

FIJI 1 0 ENBLFIJI 2 0 ENBLFIJI 3 0 UNEQ FIJI 4 0 UNEQ FIJI 5 0 UNEQ FIJI 6 0 UNEQ FIJI 7 0 UNEQ

.

ISR000 .stat ring 1

RING STATE: DRIVES HALF (480 - 959 )RING AUTO RECOVERY IS ONFIJI 0 1 ENBLFIJI 1 1 ENBLFIJI 2 1 ENBLFIJI 3 1 UNEQ FIJI 4 1 UNEQ FIJI 5 1 UNEQ FIJI 6 1 UNEQ FIJI 7 1 UNEQ

.****>OVL000 >

To check the status of ELAN/CLAN

>ld 48

LNK000 .ld 48

LNK001 .stat elnk

LNK002 .elanLNK001 .stat elanSERVER TASK: ENABLED ELAN #: 016 DES: elan APPL_IP_ID: X LYR7: ACTIVE EMPTY APPL ACTIVE ELAN #: 017 DES: Callpilot APPL_IP_ID: X LYR7: ACTIVE EMPTY APPL ACTIVE DTC001 ***LNK000

.****>OVL000

To check the elink status

>ld 137

CIOD000

.stat elnk

ELNK ENABLEDAuto Negotiation: EnabledAuto Negotiation Completed: YESActual Line Speed: 100 MbpsActual Duplex Mode: Half Duplex

Ethernet (gei unit number 0):Host: MERIDIANA104Internet address: 172.16.25.1Broadcast address: 172.16.25.255Ethernet address: 00:c0:8b:0a:d5:c6Netmask: 0xffff0000; Subnetmask: 0xffffff00 0 packets received; 143268948 packets sent0 input errors; 0 output errors0 collisions.*****

To check the host ip

>ld 117

OAM000

=> prt host ID Hostname IP Address 1 LOCAL_PPP_IF 137.135.192.4 2 REMOTE_PPP_IF 100.1.1.1 3 .255.255.0 4 MERIDIANA104 172.16.25.1 5 MERIDIANA104_SEC 172.16.25.2

=> ****

>

MEM AVAIL: (U/P): 42893194 USED U P: 8255342 464133 TOT: 51612669 DISK SPACE NEEDED: 776 KBYTESACD DNS AVAIL: 23510 USED: 490 TOT: 24000 REQ

SCH0701 REQ *****

>OVL000 >logo

TTY #15 LOGGED OUT ADMIN1 15:31 13/6/2010 SESSION DURATION: 00:41

>****