Private branch exchange system (PBXs) operates as a connection within private organizations usually a business. Because they incorporate telephones, the general term "extension" is used to refer to any end point on the branch. The PBX handles calls between these extensions. The primary advantage of PBXs was cost savings on internal phone calls: handling the circuit switching locally reduced charges for local phone services. The private branch exchange (PBX) provides internal station-to-station communications for a well-defined set of users. Three distinct generations of private branch exchanges have appeared. In the first generation (1900-1930), a human operator manually set up calls. Second-generation private branch exchanges (mid-1930s to mid-1970s) used mechanical relays to establish the call path. The third generation of private branch exchanges is the stored-program microprocessor-controlled system. Introduced in the mid-1970s, these systems use computer instructions to perform the call set-up and tear-down. The third-generation private branch exchange is physically much smaller than electromechanical models, uses less power, and generates less heat.(Brooks, 1999)
In this project, the design of a 4 line telephone systems with full signaling and switching functions similar to those of the central office systems was embarked upon. Dial tone, busy tone, and ring tone are provided during call process. Switching employs integrated circuit (IC) matrix switches on four buses. Thus, this system is expandable to 8 lines (4 pairs) if more hardware is added. This system is switching on the Dual Tone Multi Frequency (DTMF) dialing signal.
1.1 STATEMENT OF PROBLEM
The major problems this project intends to deal with are:
* Cut down cost of internal calls made within a company.
* Eliminate the need for a central telephone company to help you monitor your internal calls.
* Eliminate Stress of notification of telephone company each time you need a new extension and thereby reducing cost.
* Ensure security of your internal calls which otherwise can be tapped by company operating it.
* Eliminate the need for a manual switchboard and subsequently an operator to connect the calls.
* Reduce man-hours lost through staff walking about in an office in order to pass information to each other.
1.2 AIMS AND OBJECTIVES:
The main aim of this project is to design and implement a 4 line private exchange box that is able to create connection between four different telephone lines internally without having to connect to an external or trunk line.
The objectives include:
* Establishing connections between the telephone sets of any two users. (e.g. mapping a dialed number to a physical phone)
* Maintaining such connections as long as the users require them. (i.e. channeling voice signals between the users)
* Creating an easy means of communication in an office without getting to spend money for their internal calls.
* To switch between telephone users thereby creating connections.
* To make sure the connection remains in place as long as it last, by keeping its resources.
* To properly end the connection when a user hangs up.
1.3 SIGNIFICANCE OF STUDY
The ability or concept of providing an easy and less expensive way of communication within a small office or organization without having to pay for your internal calls or having limits to the rate or length of calls within the office. Also it is not necessary to go from office to office when something is needed, information is to be passed; a call to a colleague saves stress of walking about.
1.4 SCOPE OF STUDY
The Private Exchange System in this project is limited to a four lines which means that internal calls can be made from only four nodes. As such, it is only suitable for very small organization.
1.5 RESEARCH METHODOLOGY
The review of existing and related works to source appropriate information on how to go about the implementation of the project will be carried out. Information shall be gathered from text books, magazines, journals, and World Wide Web to provide answers in relation to the study. Based on the review, the design and implementation of a four line private exchange box system shall be carried out.
1.6 LIMITATIONS OF STUDY
There are several factors that could contribute to the group not delving deeper into this project which could have resulted in a more comprehensive work. Constraints are unavoidable in any system, be it a natural system or a computer system. Due to the extensiveness of this project topic, limitations were encountered some of which include:
* Time constraint.
* Financial constraints.
* Inadequate facilities to work with.
1.7 ORGANIZATION OF WORK
In chapter one, the research topic is introduced, which is followed by the statement of problem after which the aims and objectives of the study are stated, significance of study, scope of study and research methodology are all identified. The second chapter gives us a view of the related works which have been done and how they are related to our work. The third chapter is about our design methodology and this emphasizes on how the whole private exchange system works and its components. The quality of the system is tested and documented in chapter four. Also in chapter four, an in-depth manual of the system functions and contents is given. A summary of all chapters, a conclusion is outlined in chapter five.
2.0 HISTORY OF PRIVATE EXCHANGE BOX
In the field of telecommunications, a telephone exchange or telephone switch is a system of electronic components that connects telephone calls. A central office is the physical building used to house inside plant equipment including telephone switches, which make phone calls "work" in the sense of making connections and relaying the speech information. Early telephone exchanges are a suitable example of circuit switching; the subscriber would ask the operator to connect to another subscriber, whether on the same exchange or via an inter-exchange link and another operator. In any case, the end result was a physical electrical connection between the two subscribers' telephones for the duration of the call. The copper wire used for the connection could not be used to carry other calls at the same time, even if the subscribers were in fact not talking and the line was silent.
The first telephone exchange opened in New Haven, Connecticut in 1878. The switchboard was built from "carriage bolts, handles from tea pot lids and bustle wire" and could handle two simultaneous conversations. Later exchanges consisted of one to several hundred plug boards staffed by telephone operators. Each operator sat in front of a vertical panel containing banks of ¼-inch tip-ring-sleeve (3-conductor) jacks, each of which was the local termination of a subscriber's telephone line. In front of the jack panel lay a horizontal panel containing two rows of patch cords, each pair connected to a cord circuit. When a calling party lifted the receiver, a signal lamp near the jack would light. The operator would plug one of the cords (the "answering cord") into the subscriber's jack and switch her headset into the circuit to ask, "number please?" Depending upon the answer, the operator might plug the other cord of the pair (the "ringing cord") into the called party's local jack and start the ringing cycle, or plug into a trunk circuit to start what might be a long distance call handled by subsequent operators in another bank of boards or in another building miles away.
2.1 PBX SYSTEM COMPONENTS
PBX is a telephone exchange serving a single organization and having no means for connecting to a public telephone system it serves a user company which wants to have its own communication branch to save some money on internal calls. This is done by having the exchanging or switching of circuits done locally, inside the company. There are some important components which play a major role in the implementation of an effective PBX system.
Some of the Component
* The PBX's internal switching network.
* Central processor unit (CPU) or computer inside the system, including memory.
* Logic cards, switching and control cards, power cards and related devices that facilitate PBX operation.
* Stations or telephone sets, sometimes called lines.
* Outside Telco trunks that deliver signals to (and carry them from) the PBX.
* Console or switchboard allows the operator to control incoming calls.
* Uninterruptible Power Supply (UPS) consisting of sensors, power switches and batteries.
* Interconnecting wiring.
* Cabinets, closets, vaults and other housings.
2.2 PRIVATE BRANCH EXCHANGE (PBX)
There are essentially three different types of PBXs that could be deployed within an organization infrastructure. It is necessary to be certain of type in use, so as to be able to identify the essential numbers.
There are currently three different PBX classes:
Centrex; Direct Inward Dialing (DID)/Direct Outward Dialing (DOD) and Megalink.
Centrex is the easiest of the PBX types. This PBX, unlike other types is installed within the telephone company's Central Office (CO) and does not require dialing an extension code (normally 4 numeric characters) after having dialed the 7 to 10 digit number to connect a call to an individual. In a simplistic manner, it could be considered similar to the telephone used at home. It has an area code (NPA), an Exchange (NXX) and a Unique Number, (0000 to 9999) and does not require the dialling of another number after it in order to place a call. These numbers may be entered through a PAD.
2.2.2 Direct Inward Dialing(DID)/ Direct Outward Dialing (DOD)
Unlike a Centrex, these types of PBXs is not installed within the telephone company's Central Office. Secondly, if a cut of the telephone wire occurs outside the building, individuals are still able to dial within it to talk to colleagues by simply dialing their extension number (normally a number between 0000 to 9999) lastly; this PBX is controlled via a computer interface at a control console. Since the PBX requires constant power to function, it may be necessary to hook it with generating plant, in the absence of power from electricity company.
Direct Inward Dialing (DID) and Direct Outward Dialing (DOD) are simply features of an Automated PBX which require that you dial the company's general telephone number followed by the entry of the individual's extension number when prompted to do so. DIDs allow you direct dialing (seven digits) to locate an individual within an organization's PBX. It is a trunk phone number that must be entered into the PAD program and flagged as a PBX to ensure that the outgoing line(s) get priority.
PBXs may be privately owned or telecommunication company owned. If PBX is programmable it is possible to assign specific trunk lines to specific numbers. These trunk line numbers may then be entered on PAD thus providing dial tone protection.
The major difference between this and a Centrex PBX is that the exiting trunk lines from a building to the telephone company central office are comprised of fibre optic cables and not through twisted pair wiring. Another difference is that unlike a Centrex that is identified by its' ten digit telephone number (NPA, NXX, and Unique), Megalinks are identified by a circuit ID number. This number may contain characters and may even resemble a telephone number, however, PAD does not allow for the entry of the circuit switch identifier. The reason is quite simple, fibre optic cabling circuits can handle far more traffic than twisted pair PBXs.
2.3 INTERFACE STANDARDS
Interfaces for connecting extensions to a PBX include:
• POTS (Plain Old Telephone System) - the common two-wire interface used in most homes. This is cheap and effective, and allows almost any standard phone to be used as an extension.
• Proprietary - the manufacturer has defined a protocol. One can only connect the manufacturer's sets to their PBX, but the benefit is more visible information displayed and/or specific function buttons.
• DECT - a standard for connecting cordless phones.
• Internet Protocol - For example, H.323 and SIP.
Interfaces for connecting PBXs to each other include:
• Proprietary protocols - if equipment from several manufacturers is on site, the use of a standard protocol is required.
• QSIG - for connecting PBXs to each other, usually runs over T1 (T-carrier) or E1 (E-carrier) physical circuits.
• DPNSS - for connecting PBXs to trunk lines. Standardised by British Telecom, this usually runs over E1 (E-carrier) physical circuits.
• Internet Protocol - H.323, SIP and IAX protocols are IP based solutions which can handle voice and multimedia (e.g. video) calls.
Interfaces for connecting PBXs to trunk lines include:
• Standard POTS (Plain Old Telephone System) lines - the common two-wire interface used in most domestic homes. This is adequate only for smaller systems, and can suffer from not being able to detect incoming calls when trying to make an outbound call.
• ISDN - the most common digital standard for fixed telephony devices. This can be supplied in either Basic (2 circuit capacity) or Primary (24 or 30 circuit capacity) versions. Most medium to large companies would use Primary ISDN circuits carried on T1 or E1 physical connections.
• RBS - (Robbed bit signaling) - delivers 24 digital circuits over a four-wire (T1) interface.
• Internet Protocol - H.323, SIP, MGCP, and Inter-Asterisk eXchange protocols operate over IP and are supported by some network providers.
Interfaces for collecting data from the PBX:
• Serial interface - historically used to print every call record to a serial printer. Now an application connects via serial cable to this port.
• Network Port (Listen mode) - where an external application connects to the TCP or UDP port. The PBX then starts streaming information down to the application.
• Network Port (Server mode) - The PBX connects to another application or buffer.
• File - The PBX generates a file containing the call records from the PBX.
The call records from the PBX are called SMDR, CDR, or CIL. (Micheal, 1999)
Telephone is one of the most amazing devices ever created. Although most people take it completely for granted, the telephone is one of the most amazing devices ever created. To talk to someone, just pick up the phone and dial a few digits; connection will be established with the person and a two-way conversation can take place. It is an instrument designed for simultaneous transmission and reception of the human voice. It works by converting the sound waves of the human voice to pulses of electrical current, transmitting the current, and then retranslating the current back to sound. The U.S. patent granted to Alexander Graham Bell in 1876 for developing a device to transmit speech sounds over electric wires is often called the most valuable ever issued. Within 20 years, the telephone acquired a form that has remained fundamentally unchanged for more than a century. The advent of the transistor (1947) led to lightweight, compact circuitry . Advances in electronics have allowed the introduction of a number of "smart" features such as automatic redialing, caller identification, call waiting, and call forwarding. The figure 2.1 shows the major components that makes up a telephone set.
2.5 HOW TELEPHONE WORKS
When a person speaks into a telephone, the sound waves created by his voice enter the mouthpiece. An electric current carries the sound to the telephone of the person he is talking to. A telephone has two main parts: (1) the transmitter and (2) the receiver.
The Transmitter of a telephone serves as a sensitive "electric ear." It lies behind the mouthpiece of the phone. Like the human ear, the transmitter has 14 eardrum." The eardrum of the telephone is a thin, round metal disk called a diaphragm. When a person talks into the telephone, the sound waves strike the diaphragm and make it vibrate. The diaphragm vibrates at various speeds, depending on the variations in air pressure caused by the varying tones of the speaker's voice. Behind the diaphragm lies a small cup filled with tiny grains of carbon. The diaphragm presses against these carbon grains. Low voltage electric current travels through the grains. This current comes from batteries at the telephone company. The pressure on the carbon grains varies as sound waves make the diaphragm vibrate. A loud sound causes the sound waves to push hard on the diaphragm. In turn, the diaphragm presses the grains tightly together. This action makes it easier for the electric current to travel through, and a large amount of electricity flows through the grains. When the sound is soft, the sound waves push lightly on the diaphragm. In turn, the diaphragm puts only a light pressure on the carbon grains. The grains are pressed together loosely. This makes it harder for the electric current to pass through them, and less current flows through the grains.
Thus, the pattern of the sound waves determines the pressure on the diaphragm. This pressure, in turn, regulates the pressure on the carbon grains. The crowded or loose grains cause the electric current to become stronger or weaker. The current copies the pattern of the sound waves and travels over a telephone wire to the receiver of another telephone.
The Receiver serves as an "electric mouth." Like a human voice, it has "vocal cords." The vocal cords of the receiver are a diaphragm. Two magnets located at the edge of the diaphragm cause it to vibrate. One of the magnets is a permanent magnet that constantly holds the diaphragm close to it. The other magnet is an electromagnet. It consists of a piece of iron with a coil of wire wound around it. When an electric current passes through the coil, the iron core becomes magnetized. The diaphragm is pulled toward the iron core and away from the permanent magnet. The pull of the electromagnet varies between strong and weak, depending on the variations in the current. Thus, the electromagnet controls the vibrations of the diaphragm in the receiver.
The electric current passing through the electromagnet becomes stronger or weaker according to the loud or soft sounds. This action causes the diaphragm to vibrate according to the speaker's speech pattern. As the diaphragm moves in and out, it pulls and pushes the air in front of it. The pressure on the air sets up sound waves that are the same as the ones sent into the transmitter. The sound waves strike the ear of the listener and he hears the words of the speaker. (www.howstuffworks.com)
2.6 THE RINGER
Simply speaking this is a device that alerts you to an incoming call. It may be a bell, light, or warbling tone. The ringing signal is in an AC wave form. Although the common frequency used can be any frequency between 15 and 68 Hz. Most of the world uses frequencies between 20 and 40 Hz. The voltage at the subscribers end depends upon loop length and number of ringers attached to the line; it could be between 40 and 150 Volts.
The ringing cadence (the timing of ringing to pause), varies from company to company. In the United States the cadence is normally 2 seconds of ringing to 4 seconds of pause. An unanswered phone in the United States will keep ringing until the caller hangs up. But in some countries, the ringing will "time out" if the call is not answered. The most common ringing device is the gong ringer; a solenoid coil with a clapper that strikes either a single or double bell. A gong ringer is the loudest signaling device that is solely phone-line powered.
Modern telephones tend to use warbling ringers, which are usually ICs powered by the rectified ringing signal. The audio transducer is a small loudspeaker via a transformer.
Ringers are isolated from the DC of the phone line by a capacitor. Gong ringers in the United States use a 0.47 uF capacitor. Warbling ringers in the United States generally use a
1.0 uF capacitor. Telephone companies in other parts of the world use capacitors between 0.2 and 2.0 uF. The paper capacitors of the past have been replaced almost exclusively with capacitors made of Mylar film. Their voltage rating is always 50 Volts. The capacitor and ringer coil, or Zeners in a warbling ringer, constitute a resonant circuit. When phone is hung up ("on hook") the ringer is across the line; and it has merely silenced the transducer, not removed the circuit from the line. When the telephone company uses the ringer to test the line, it sends a low-voltage, low frequency signal down the line (usually 2 Volts at 10 Hz) to test for continuity. The company compares result with the expected signals of the line. This is how it can tell whether an added equipment is on the line. If your telephone has had its ringer disconnected, the telephone company cannot detect its presence on the line.
Because there is only a certain amount of current available to drive ringers, if ringers are added to phone lines indiscriminately, a point will be reached at which either all ringers will cease to ring, some will cease to ring, or some ringers will ring weakly. A normal ringer is defined as a standard gong ringer as supplied in a phone company standard desk telephone; Value given to this ringer is Ringer Equivalence Number (REN) 1. It can be as high as 3.2, which means that device consumes the equivalent power of 3.2 standard ringers, or 0.0, which means it consumes no current when subjected to a ringing signal. If there is a problem with ringing, it could be that the REN is greater than 5, disconnecting ringers until REN is at 5 or below will usually solve the problem. Other countries have various ways of expressing REN, and some systems will handle no more than three of their standard ringers. But whatever the system, if an extra equipment was added and the phones stop ringing, or the phone answering machine won't pick up calls, the solution is disconnect ringers until the problem is resolved. Warbling ringers tend to draw less current than gong ringers, so changing from gong ringers to warbling ringers may help spread the sound better.
Frequency response is the second criterion by which a ringer is described. Because a ringer is supposed to respond to AC waveforms, it will tend to respond to transients (such as switching transients) when the phone is hung up, or when the rotary dial is used on an extension phone. This is called "bell tap" in the United States; in other countries, it's often called "bell tinkle." While
European and Asian phones tend to bell tap, or tinkle, United States ringers that bell tap are considered defective. The bell tap is designed out of gong ringers and fine tuned with bias springs. Warbling ringers for use in the United States are designed not to respond to short transients; this is usually accomplished by rectifying the AC and filtering it before it powers the IC, then not switching on the output stage unless the voltage lasts long enough to charge a second capacitor.(Roberts, 2006)
2.7 HOOK SWITCH
This is a lever that is depressed when the handset is resting in its cradle. It is a two-wire to four-wire converter that provides conversion between the four-wire handset and the two-wire local loop. There are two stages, which are off hook and on hook
Off hook: The state of a telephone line that allows dialing and transmission but prohibits incoming calls from being answered. The phone is off-hook when the handset is removed from the base unit of a stationary phone or press Talk on a portable phone. The term stems from the days when the handset was lifted off an actual hook. When the handset was removed, a spring caused contacts to press together, closing the circuit from the telephone to the switchboard.
On hook: The condition that exists when a telephone or other user instrument is not in use, i.e., when idle waiting for a call. Note: on-hook originally referred to the storage of an idle telephone reciever, i.e., separate earpiec, on a swithch hook. The weigth of the recieved depresses the sping leaded switch hook thereby disconnecting the idle instrument (except its bell) from the telephone line. (Roberts, 2006)
2.8 THE DIAL
There are two types of dials in use around the world. The most common one is called pulse, loop disconnect, or rotary; the oldest form of dialing, it's been in use since the 1920's. The other dialing method, is called Touch-tone, Dual Tone Multi-Frequency (DTMF)
Pulse dialing is traditionally accomplished with a rotary dial, which is a speed governed wheel with a cam that opens and closes a switch in series with the phone and the line. It works by actually disconnecting or "hanging up" the telephone at specific intervals. The United States standard is one disconnect per digit, so if a "1," is dailled, the telephone is "disconnected" once. To dial a seven means that it will be "disconnected" seven times; and dialling a zero means that it will "hang up " ten times. Some countries invert the system so "1" causes ten "disconnects" and 0, one disconnect. Some add a digit so that dialing a 5 would cause six disconnects and 0, eleven disconnects. There are even
some systems in which dialing 0 results in one disconnect, and all other digits are plus one, making a 5 cause six disconnects and 9, ten disconnects.
Although most exchanges are quite happy with rates of 6 to 15 Pulses Per Second (PPS), the phone company accepted standard is 8 to 10 PPS. Some modern digital exchanges, free of the mechanical inertia problems of older systems, will accept a PPS rate as high as 20. Besides the PPS rate, the dialing pulses have a make/break ratio, usually described as a percentage, but sometimes as a straight ratio. The North American standard is 60/40 percent; most of Europe accepts a standard of 63/37 percent. This is the pulse measured at the telephone, not at the exchange, where it's somewhat different, having traveled through the phone line with its distributed resistance, capacitance, and inductance. In practice, the make/break ratio does not seem to affect the performance of the dial when attached to a normal loop. However,each pulse is a switch connect and disconnect across a complex impedance, so the switching transient often reaches 300 Volts. Usually, a safe practice is not to have fingers across the line when dialing.
Most pulse dialing phones produced today use a CMOS IC and a keyboard. Instead of pushing finger round in circles, then removing finger and waiting for the dial to return before dialing the next digit, the button can be punched as fast as desired. The IC stores the number and pulses out the number at the correct rate with the correct make/break ratio and the switching is done with a high-voltage switching transistor. Because the IC has already stored the dialed number in order to pulse it out at the correct rate, it's a simple matter for telephone designers to keep the memory "alive" and allow the telephone to store, recall, and redial the Last Number Dialed (LND). This feature enables easy redial by picking up the handset and pushing just one button.
Touch tone is the most modern form of dialing. It is fast and less prone to error than pulse dialing. Compared to pulse, its major advantage is that its audio band signals can travel down phone lines further than pulse, which can travel only as far as the local exchange. Touch-tone can therefore send signals around the world via the telephone lines, and can be used to control phone answering machines and computers.
Bell Labs developed DTMF in order to have a dialing system that could travel across microwave links and work rapidly with computer controlled exchanges. Each transmitted digit consists of two separate audio tones that are mixed together. The four vertical columns on the keypad are known as the high group and the four horizontal rows as the low group; the digit 8
is composed of 1336 Hz and 852 Hz. The level of each tone is within 3 dB of the other. A complete touch-tone pad has 16 digits, as opposed to ten on a pulse dial. Besides the numerals 0 to 9, a DTMF "dial" has *, #, A, B, C, and D. Although the letters are not normally found on consumer telephones, the IC in the phone is capable of generating them.
The * sign is usually called "star" or "asterisk." The # sign, often referred to as the "pound sign." is actually called an octothorpe. Although many phone users have never used these
digits - they are not, after all, ordinarily used in dialing phone numbers. They are used for control purposes, phone answering machines, bringing up remote bases, electronic banking, and repeater control. The one use of the octothorpe that may be familiar occurs in dialing international calls from phones. After dialing the complete number, dialing the octothorpe lets the exchange know you've finished dialing. It can now begin routing your call; without the octothorpe, it would wait and "time out" before switching your call.
Standard DTMF dials will produce a tone as long as a key is depressed. No matter how long you press, the tone will be decoded as the appropriate digit. The shortest duration in which
a digit can be sent and decoded is about 100 milliseconds (ms). It's pretty difficult to dial by hand at such a speed, but automatic dialers can do it. A twelve-digit long distance number
can be dialed by an automatic dialer in a little more than a second - about as long as it takes a pulse dial to send a single 0 digit.(Roberts,2006)
2.9 MODULAR CONNECTORS
Modular connector is the name given to a family of electrical connectors that were originally used in telephone wiring. Even though they are still used for that purpose they are used for a variety of other things as well. A modular connector's advantage over many other kinds include; small size and ease of plugging and unplugging. Many uses that originally used a bulkier connector have migrated to modular connectors. Probably the most well known applications of modular connectors is for telephone jacks and for ethernet jacks, which are nearly always modular connectors. Figure 2.2 shows types of connectors commonly used.
Modular connectors were first used in the registered jack system, so registered Jack specifications describe them precisely. These are the specifications to which all practical modular connectors are built.
Modular connectors come in four sizes: 4-, 6-, 8-, and 10-position. A position is a place that can hold a conductor (pin). The positions need not all be used; a connector can have any even number of conductors. Unused positions are usually the outermost positions. The connectors are designed so that a plug can fit into any jack that has at least the number of positions as the plug. Where the jack has more positions than the plug, the outermost positions are unused. However, plugs from different manufacturers may not have this compatibility, and some manufacturers of eight position jacks now explicitly warn that they are not designed to accept smaller plugs without damage. The positions of a jack are numbered left to right, looking into the receiving side of the jack with the hook (locking tab or clip) side down, starting at 1. The positions of plug are numbered the same as the jack positions with which they mate. The number of a conductor is the same as the number of the position it's in. So for example in a 6P2C plug, only conductors 3 and 4 exist.
Some connector types in the family are indexed, which means their shape is altered from the standard somewhat to prevent them from mating with standard connectors. The indexing is usually a different shape or position of the hook, but can also be an additional tab. The members of the family are typically identified using the format "[number]P[number]C", e.g. "6P2C", which means 6 positions, 2 conductors. Alternate formats "[number]x[number]" (e.g. "6x2") or "[number]/[number" (e.g. "6/2") are also used.
Modular connectors have gender. The male connector is called a plug, while the female connector is called a jack or sometimes a socket. The application of jack versus plug is generally based on physical installation only. Jacks go in walls and panels, while plugs go on wires. Modular connectors also go by the names "modular phone jack/plug", "RJ connector," and "Western jack/plug." The 8P8C modular connector type is often called RJ45, Modular connectors lock together. A spring-loaded tab called a "hook" on the plug snaps into a jack so that the plug cannot be pulled out. To remove the plug, the hook has to be pressed. The most common way to install a jack in a wall or panel is with the hook side down. This usually makes it easier to operate the hook when removing the plug, because the person grabs the plug with thumb on top and presses the hook with the index finger. A disadvantage of modular connectors is that the fragile hook on a plug easily snags and often breaks while trying to pull a cord through other cords or other obstructions. Some higher quality cables have a flexible sleeve called a "boot" over the plug, or a special hook design, to prevent this. Boots are seen mainly on 8P8C data cables.(www.whatis.com)
This chapter emphasizes on the information or content presented, the functions that will be performed and the behaviour that each system component is expected to exhibit. The design focuses on how the system looks like, the layout and technical structure of the system. A breakdown of this system circuits will further ascertain the functionality of the system. The figure below is a block diagram that describes the whole communication system.
The block diagram in figure 3.0 is comprised:
* THE CONTROL UNIT
* TONE GENERATION
* DECODING AND SWITCHING
3.1 THE CONTROL CIRCUIT
This is where all control functions of the system are being carried out. The main device that performs this role is the microcontroller which is the heart of the system. The frequency used to generate the tones is from the microcontroller, It is used to monitor the hook comparator to know when it is off-hook (the condition that exists when a telephone is in use) or on-hook (the condition that exists when a telephone is not in use) .When a number is dialed the microcontroller helps to check the EPROM (Erasable Programmable Read Only Memory) if the number dialed is valid before it allows a call to be successful.
A microcontroller is a computer-on-a-chip used to control interfaces needed
for a simple application. A microcontroller is a single integrated circuit, commonly with the following features:
* bit processors to sophisticated 32- or 64-bit processors
* serial ports
* signal conversion circuits.
* RAM for data storage
* program storage
The 8051 used in this project is an 8-bit microprocessor originally designed in the 1980's by Intel that has gained great popularity since its introduction. Its standard form includes several standard on-chip peripherals, including timers, counters, plus 4kbytes of on-chip program memory and 128 bytes of data memory, making single-chip implementations possible. The 8051 memory architecture includes 128 bytes of data memory that are accessible directly by its instructions. A 32-byte segment of this 128 byte memory block is bit addressable by a subset of the 8051 instructions, namely the bit-instructions. External memory of up to 64 Kbytes is accessible by a special instruction. Up to 4 Kbytes of program instructions can be stored in the internal memory of the 8051, or the 8051 can be configured to use up to 64 Kbytes of external program memory .The majority of the 8051's instructions are executed within 12 clock cycles. The package outline of 8051 microcontroller is shown in fig 3.1
3.1.1 EPROM (Erasable Programmable Read Only Memory)
The function of the EPROM is to store the program that was used .The numbers assigned to each telephone are being programmed into it. It is a type of computer memory chip that retains its data when its power supply is switched off. In other words, it is non-volatile. It is an array of floating-gate transistors individually programmed by an electronic device that supplies higher voltages than those normally used in electronic circuits. Once programmed, the EPROM can be erased only by exposing it to strong ultraviolet light. It is a ROM-type chip that can hold data from 10-20 years. It is different from PROM because it can be programmed more than once. The EPROM is configured or reconfigured using an EPROM programmer.
The EPROM has 8 data lines (bits AD0 - AD7) forming a byte wide data bus. Enabling both ALE (Address latch Enable) and PSEN (Program Store Enable) causes all the bits to appear on the data lines, as such individual bits cannot be enabled. The byte of data appears on these lines as 0's or 1's. The data lines of the EPROM are bi-directional, they are outputs when it is being read and they serve as inputs are programming it. Figure 3.2 shows the package view of EPROM (Erasable Programmable Read Only Memory).(Mano, 2008)
A latch is a kind of bistable multivibrator, an electronic circuit which has two stable states and thereby can store one bit of information its output may depend not only on its current input, but also on its previous inputs.The Latch also separates the Intel 8051 microcontroller address lines from its data lines. The Intel 8255 Programmable Peripheral Interface chip is a peripheral chip. This chip is used to give the CPU access to programmable parallel I/O. The 8255 chip is used together with a microcontroller to expand its I/O capabilities.
3.2 TONE GENERATION
The tone generation deals with how the tones used for the exchange box are generated.
The sine wave is the most familiar AC waveform and is the type of wave used in telecommunication. It derives its name from the fact that the current or voltage varies with the sine of the elapsed time. The sine wave is unique in that it represents energy entirely concentrated at a single frequency. However, the microcontroller only generates a square wave.
A square wave is a waveform that is built up from a series of harmonics derived from the fundamental frequency. A true square wave will have 3rd, 5th, 7th, 9th, 11th, 13th and 15th harmonics. The rise and fall is very abrupt, straight up and straight down. For an audio signal, all of these combined odd order harmonics would not be considered to be a pleasant sound. Hence, to generate the required sine wave, the square wave has to be filtered.
3.2.1 LOW-PASS FILTER CIRCUIT
The low-pass filter circuit is used to cut-off the harmonics in the square waves in order to convert it to a sine wave which is needed. The term "low-pass filter" merely refers to the shape of the filter's response. A low-pass filter is a filter that passes low-frequency signals but attenuates (reduces the amplitude of) signals with frequencies higher than the cutoff frequency. The actual amount of attenuation for each frequency varies from filter to filter. It is sometimes called a high-cut filter, or treble cut filter when used in audio applications. The concept of a low-pass filter exists in many different forms, including electronic circuits. One simple electric circuit that will serve as a low-pass filter consists of a resistor in series with a load, and a capacitor in parallel with the load as shown in figure 3.4. The capacitor exhibits reactance, and blocks high-frequency signals, causing them to go through the load instead. At higher frequencies the reactance drops, and the capacitor effectively functions as a short circuit. The combination of resistance and capacitance gives you the time constant of the filter t = RC .The break frequency, also called the turnover frequency or cutoff frequency (in hertz), is determined by the time constant. The figure below is a low pass filter circuit. (www.answers.com)
The buffer is used to separate the phase of low-pass filter circuit so that it does not affect other operations on the circuit. A buffer amplifier is one that provides electrical impedance transformation from one circuit to another. Typically a buffer amplifier is used to transfer a voltage from a first circuit, having a high output impedance level, to a second circuit with a low input impedance level. The interposed buffer amplifier prevents the second circuit from loading the first circuit unacceptably and interfering with its desired operation. If the voltage is transferred unchanged (the voltage gain is 1), the amplifier is a unity gain buffer; also known as a voltage follower.
3.2.3 PASSIVE COMPONENTS
Some Passive components were utilized in the design. Passive components are electronic component that does not increase the power of the electrical signal on which it acts. It may actually end up decreasing the power of the signal it is acting upon. As such, a passive component may draw the energy it uses for its own operation directly from the signal on which it is operating.
They are constant components and have resistance (R), capacitance (C) and inductance (L) properties respectively.
Resistors: increase the current at the expense of the voltage. A resistor's resistance (R) is a measure of the ratio of its potential difference (V) with the current (I): R = V / I
The value of resistors are usually expressed in Ohms (?),
Capacitors: store electrical energy. A capacitors capacitance (C) is a measure of the amount of charge (Q) stored on each plate for a given potential difference or voltage (V) which appears between the plates. C = Q / V
In SI units, a capacitor has a capacitance of one farad when one coulomb of charge is stored due to one volt applied potential difference across the plates. Since the farad is a very large unit, values of capacitors are usually expressed in microfarads (µF), nanofarads (nF) or picofarad (pF)
Their schematic diagrams and symbols are shown in figure 3.5 below.
A transformer is being used in the circuit used to generate a ring back tone because the voltage required to generate it is 90v - 120v and the voltage in the circuit is 5v - 12v. So the transformer was used to step up the voltage. A transformer is a device that transfers circuit to another through wires. A changing current in the first circuit (the primary) creates a changing magnetic field; in turn, this magnetic field induces a changing voltage in the second circuit (the secondary). By adding a load to the secondary circuit, one can make current flow in the transformer, thus transferring energy from one circuit to the other. The secondary induced voltage VS is scaled from the primary VP by a factor ideally equal to the ratio of the number of turns of wire in their respective windings. (James, 2004)
Four different tones can be obtained from the circuit of figure 3.6
The square waves from the microcontroller are being passed to a low pass filter circuit and are converted to sine waves. The sign waves are then buffered. A buffer is used to separate the phase of low-pass filter circuit so that it does not affect other operations on the circuit. Two of the frequencies are then mixed to generate a tone.
Dial tone: it is a telephony signal used to indicate that the telephone exchange is working and ready to accept a call. The tone stops when the first numeral is dialed, or if there is no response after going off-hook. The tone is generated by adding two sine waves of 350Hz and 440Hz
Ring back tone: this is a tone that is heard when your call is successful. The tone is generated by adding two sine waves of 440Hz and 480Hz.
Busy tone: is a telephony signal used to indicate that the called telephone is being used at the moment, so it cannot accept a call. The tone is generated by adding two sine waves of 480Hz and 620Hz.
Ring tone: is generated when a call goes through to the other party's phone. The tone is composed of a sine wave of 20Hz. The voltage required is 90V - 120V, thus necessitating the use of step-up transformer.
3.3 DECODING AND SWITCHING
This is where the decoding of the tones and the switching of the phone lines are carried out. The switching is performed by use of a relay is used in the phone circuit because of the voltage of the ring back tone which uses 90V - 120V while the system uses 5V - 12V. The relay is used to switch between the ring back tone and the normal phone circuit. Therefore each time a call is made and it gives a ring back tone the relay switches to ring back and when the call is picked it goes back to the normal circuit. The relay is an electromechanical switch operated by a flow of electricity in one circuit and controlling the flow of electricity in another circuit. The relay consists basically of an electromagnet with a soft iron bar, called an armature, held close to it. A movable contact is connected to the armature in such a way that the contact is held in its normal position by a spring. When the electromagnet is energized, it exerts a force on the armature that overcomes the pull of the spring and moves the contact so as to either complete or break a circuit. When the electromagnet is de-energized, the contact returns to its original position. Variations on this mechanism are possible: some relays have multiple contacts; some are encapsulated; some have built-in circuits that delay contact closure after actuation; some, as in early telephone circuits, advance through a series of positions step by step as they are energized and de-energized. They are used in a wide variety of applications throughout industry, such as in telephone exchanges, digital computers.
The comparator is used to detect when the phone is off hook or on hook. The microcontroller is being used to monitor the comparator, once it is off-hook current flows through and it detects it. In electronics a comparator is a device which compares two voltages or currents and switches its output to indicate which is larger. A dedicated voltage comparator will also contain additional feature such as an accurate, internal voltage reference.. In theory the reference and input voltages can be anywhere between zero and the supply voltage but there are practical limitations on the actual range depending on the particular device used(www.wikipedia.com)
3.3.2 DTMF (DUAL-TONE MULTI FREQUENCY)
DTMF GENERATOR AND DECODER (MT8870): DTMF (Dual-tone Multi Frequency) is a tone composed of two sine waves of given frequencies. Individual frequencies are chosen so that it is quite easy to design frequency filters, and so that they can easily pass through telephone lines (where the maximum guaranteed bandwidth extends from about 300 Hz to 3.5 kHz). The dial tone heard when the phone set is picked up is called Dual Tone Multi-Frequency (DTMF). The name was given because the tone that we heard over the phone is actually made up of two distinct frequency tones, hence the name dual tone. The DTMF tone is a form of one way communication between the dialer and the telephone exchange. A complete communication consists of the tone generator and the tone decoder. In this project MT8870 was used as the main component to decode the input dial tone to 4 digital output. The MT8870 is a complete DTMF receiver integrating both the bandsplit filter and digital decoder functions. The filter section uses switched capacitor techniques for high and low group filters; the decoder uses digital counting techniques to detect and decode all 16 DTMF tone-pairs into a 4-bit code.
The tones are generated from phones based on the keys pressed. This table resembles a matrix keyboard. The X and Y coordinates of each code give the two frequencies that the code is composed of. Notice that there are 16 codes; however, common DTMF dialers use only 12 of them. The "A" through "D" are "system" codes. Most end users won't need any of those (are used by some PBX systems for special functions). It uses electronics and computer to assist in the phone line connection. Basically on the caller side, it is a dial tone generator. When a key is being pressed on the matrix keypad, it generate a unique tone consisting of two audible tone frequency. For example, if the key '1' is being press on the phone, the tone you hear is actually consist of a 697hz & 1209hz sine signal. Pressing key '9' will generate the tone form by 852hz & 1477hz. The frequency use in the dial tone system is of audible range suitable for transmission over the telephone cable. On the telephone exchange side, it has a decoder circuit to decode the tone to digital code. For example, the tone of 941hz + 1336hz will be decoded as binary '1010' as the output. This digital output will be read in by a computer, which will then act as a operator to connect the caller's telephone line to the designated phone line. The telephone exchange will generate a high voltage signal to the receiving telephone, so as to ring the telephone bell, to notify the receiving user that there is an incoming call. (Stephen, 1997).
3.3.3 74393 DUAL 4-BIT (0-15) RIPPLE COUNTER
We are using this ripple counter to clock the microcontroller also it is used for the DTMF (Dual Tone Multi Frequency). It used as a replica counter for the DTMF decoder in order to double check the binary bits that have been decoded by the DTMF. The 74393 contains two separate 4-bit (0 to 15) counters, one on each side of the chip, they are ripple counters. The count advances as the clock input becomes low (on the falling-edge), this is indicated by the bar over the clock label. This is the usual clock behavior of ripple counters and it means a counter output can directly drive the clock input of the next counter in a chain. For normal operation the reset input should be low, making it high resets the counter to zero (0000, QA-QD low). The figure below is a 74393 clock.
3.3.4 8 - 1 LINE ANALOG SWITCH
A multiplexer selects one of several input signals and passes it on to the output. It is a circuit that accepts several inputs and selects one of them at any given time to pass on to the output. The routing of the desired data input to the output is controlled by select inputs often referred to as Address inputs). The analog switch is used to switch between the four phone lines and also the dial tone, busy tone, ring back tone.` (Tocci, 2004)
IMPLEMENTATION AND TESTING
The physical realization of this project is very important, because the designer will see the result of his work. The working ability of this project was constructed to meet the desired specifications, and that is in a nutshell, a system that is capable of controlling communication between four telephone lines. This switching system is a four line private exchange box.
In this chapter, the workability and efficiency of the private exchange box system, after its various subsystems have been combined together and examined.
4.1 IMPLEMENTATION AND TESTING
To test this project, at least two (2) touch tone telephone lines will be needed. The first phone A is connected to the private exchange box and the second phone B also to be connected to the exchange box.
When the phone A is picked up, the microcontroller detects that the phone has been picked which will then make the hook comparator know that it is off hook. It then sends a control signal to the The 8-1 line analog switch and it switches to dial tone. When phone A recieves a dial tone, it means it is ready to make a call. When phone A makes a call by dialing the number assigned to phone B, the DTMF(Dual tone multi-frequency) Decoder decodes the number into four binary bits one by one and temporarily stores them in the RAM (random access memory) of the microcontroller till it matches with one of the numbers stored in the EPROM, the microcontroller checks the hook comparator of phone B if it off-hook or on-hook. If phone B is on-hook it sends a control signal to the 8-1 line analog switch via the processor to switch to ring back tone therefore the relay has to switch to the ring generator and disconnect from the normal circuit because the ring generator uses 90V while the normal circuit uses 5V- 12V. Phone B will then ring and when the phone is picked you are able to communicate with each other . If the microcontroller detects that the phone is off-hook, it sends a control signal to the 8-1 line analog switch and it switches to busy tone. That means the phone B is in use so you cannot talk to that person at that time you have to call back later.
The project is working as expected. Calls can be made from any of the phones and the ring tone, busy tone and dial tone is heard. Calls can be made simultaneously and you can hear the voice of the person talking clearly. There is no limit to the time at which you can talk.
RECOMMENDATION AND CONCLUSION
This chapter gives a summary of all the chapters in this project, a conclusion and also recommends where the system can be used.
Chapter one talks about the introduction of the project and its layout. In Chapter two we talked about the history of the project, the system components, reviewed other works that have been carried out relating to our project, how telephones work. The Third chapter analyzes the components that make up the system and how they relate to each other in order to make the system function. Chapter four explains how the system works during testing and how it is expected to function.
The execution of this project has left possibilities for improvement which could not be carried out during the design and implementation because of the limited time and resources. The possible improvement on this project is that the amount of phones used can be increased and you can have one of the phone lines connected to an external line for trunk calls.
This project is recommended for use in offices, at home, hospitals or hotels
For full implementation in a bigger area you can increase the number of telephone lines in the system so that more people can have access to each other.
The beauty of this project is the fact that a means of communication that is cheap, efficient, secure and technology driven has been designed. The Four Line Telephone Exchange Box is a device that enhances the internal communication within an office or home. The idea of Communication is reliability, security, efficiency, connectivity and cost. This Project has put all this into consideration, and has come out with a very efficient way of communicating in perhaps an Organization, offices and even for our Case Study Babcock University.
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