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The PBX: What It Is, How It Works

Lee Goeller's "The PBX: What It Is, How It Works" was Chapter One of The BCR Manual of PBXs, a ring-bound information service published by Business Communications Review from 1980 to 1990.

To speed downloading, we have posted the chapter in five parts, and configured diagrams to open as separate windows.

The PBX: What It Is, How It Works

Part Three: The User Interface

Telephone sets are the interface between the telephone system and the users. Consoles, seldom more than glorified telephone sets in modern PBXs, fall into the same category. Additional interfaces, provided for maintenance administration and other information exchange, are often standard data terminals although sets or consoles with special displays are sometimes provided. Needless to say, the instrument and the switching system must be able to work with one another if the system is to support communications.

Telephone sets must, in general, convert acoustical energy to electrical energy and vice versa to permit voice communication. For non-voice services, electrical signals from machines have to be mapped into the electrical patterns internal to the PBX, an analogous function. In addition, the station user must be able to signal toward the system to "place his order," and the system must be able to signal toward the caller to let him know what the system is doing to carry out his instructions. The system must also signal toward the called party to encourage him to answer his phone and, in some instances, to tell him which line he is supposed to answer.

Conventional telephone sets

Conventional telephones are completely unsuitable for use with electronic PBXs, based on purely technical reasons relating to transmission and ringing. However, they are standard, universal and inexpensive, and customers, with no idea of the technical issues involved, feel so secure with the familiar that some PBX manufacturers who sought a better solution had to back off and return to the 1890s.

The basic mechanism of the universally standard 500-type telephone set was perfected just about the time transistors were announced (circa 1950). Thus, conventional station apparatus has not taken any particular advantage of developments in solid state physics. The 2500-type telephone set, identical to the 500 except for the use of DTMF (the generic name for AT&T’s Touch-Tone) for signaling rather than rotary dialing, does, it is true, use transistors to generate tones. But that is sort of an add-on application.

The 500/2500-type telephone does use some rather early solid-state components (called varistors) to compensate for the distance between the central office and the telephone. Obviously, a long pair of wires from phone to CO will attenuate voice and DTMF signals more than a short pair. Further, the DC current, used for supervision and for powering the telephone set, will also be less on a long loop. Varistors, sensitive to this current, attenuate voice and DTMF signals more when the DC is large, and less when it is small. In this way, voice and signaling levels at the switch are relatively constant, and no adjustment has to be made by the installer.  This reduces labor and administrative costs both at installation and over the life of the set, in addition to improving transmission.

However, a PBX telephone, unlike a phone served directly by a central office, is usually within a few hundred feet of its switch. Compared to the average distance to a CO phone (measured in miles), this is, for all practical purposes, a "zero loop." The question is, what does automatic level adjustment do under such circumstances? As the set was intended to operate, a zero loop draws a large amount of current from the switch; if an electronic PBX supplied this current to the 4, 8 or perhaps 16 lines on each of its line cards, the power drawn would be considerable and the cards would be in danger of burning up. To prevent this, the current is greatly reduced, but now the automatic voice-level compensation comes into play and the audio is much too loud. Thus the switching matrix is designed to reduce this level by some fixed amount.

So what’s the problem? Well, when a PBX telephone is making a call through the CO to a distant phone, the PBX loop plus the loop length of the trunk to the CO is much greater, and the matrix loss has to be switched back out again. Note that the electromechanical PBXs, provided by the telco in years gone by, had no problems delivering zero-loop power and, when they connected the PBX user to a CO trunk, a metallic path through the matrix let the PBX phone draw power directly from the CO. Thus compensation for inside vs. outside calls was automatic.

With the coming of interconnect in the early 1970s, the telcos insisted on a "coupler" to privately owned PBXs to protect themselves from "harms" such equipment supposedly produced. The coupler blocked the level-compensating capabilities of direct extension-trunk connections, but by then it didn’t matter much. No electronic switching matrix permits direct connection in the first place. Other much more complex means must be found to adjust call level for inside vs. outside. A telephone designed specifically for PBX use would have made the job a lot easier.

This is not the only transmission problem that the 500/2500 set produces. On a zero loop, the set is the main source of the "return loss" mentioned above in connection with echo. And it doesn’t do very well. That is, it produces a very low return loss, and makes transmission on tie-trunk networks precarious. Further, all digital PBXs must be four-wire internally, and need a hybrid in each line circuit. Thus, even for intra-PBX calls on a digital PBX, echo can easily degenerate into singing or oscillation.

Note that all these problems are well known and understood, and have been fixed with large, sloppy band-aids so that the customer is not even aware of their existence. But the better solution would be to eliminate the problems by suitable set design. Now, in the mid-1980s, many PBX manufacturers are doing just this. It is to the credit of Danray and Tele/Resources that they made the effort a decade ago by extending four-wire transmission all the way to the telephone set, eliminating its potential for creating echo.

There are other problems with conventional residential telephones on modern electronic PBXs, which we will examine in due course.

Conventional 500 or 2500-type sets have a microphone or transmitter for converting acoustical energy to electrical energy to be transmitted via wires. Similarly, they have a receiver for making the opposite conversion. The two are connected via a "network" which is actually one version of a hybrid coil. This permits a single pair of wires to be used for transmission in each direction when two separate devices are obviously required at the telephone set to interface a caller’s ear and mouth. See Figure 7.

[ Figure 7: The parts of a 500 type telephone set. ]

The hybrid coil serves an additional purpose. It acts as a "side-tone network." That is, it allows a certain amount of the spoken energy to be fed back to the ear of the speaker (without delay). This is necessary because we are used to hearing ourselves speak (cover both ears and speak aloud to detect the impact of not hearing yourself); if we don't, we change our speaking patterns. As it happens, the side-tone network feeds back slightly less sound than we hear through the air; this tends to make us talk a little louder, and provides better volume for the person on the far end of the connection. This "zero cost" amplifier is a typical example of the subtle ways in which Bell Labs moves, its miracles to perform.

The switch-hook is the next device we encounter in our tour of the telephone set. The switch-hook completes a path for current from the PBX or CO switch though the telephone set when the caller picks up the handset. The handset, of course, contains the transmitter and receiver, and, when not in use, sits in a cradle on the telephone base. While placed in the cradle, it holds the switch-hook open to turn off the power drawn from the PBX or CO; when lifted, it releases the switch-hook and current flows.

A PBX or CO must always monitor a line for its on-hook or off-hook status. This is called "supervision." The switch must know if the user is originating a call, answering a call to his phone, or ending a call that has been in progress. In recent years, the switch-hook "flash" from conventional single line phones has regained the importance it had in manual systems. A flash is a momentary on-book. The system detects it, and knows that it is not a hang-up followed by the origination of a new call. (In equipment made to AT&T specifications, the system also knows that a flash is not a digit 1 from a rotary dial, but GTE Automatic Electric has used a dial 1 signal for a flash.) Clearly, accurate timing and careful user behavior are required.

Some older multi-line telephone sets (with 10 buttons or more) do not make a clean flash—they bounce, and confuse the system. This makes life even more difficult for the station user. Several traditional key telephone systems and single line sets, to facilitate their use with modern PBXs, include a "service button" which, when depressed, sends a properly timed flash. (Some systems use the service button to send a "ground.")

The purpose of the flash is to tell the system that the user wants to send an additional command, a "feature code." Particularly if the system uses DTMF, but often when dial pulsing is used as well, a digit receiver must be connected to unscramble the user's new command. Use of the switch-hook flash is common in modern PBXs, followed by a choice of 20 or more feature codes.

During 1980 and 1981, a new approach to feature codes appeared. Repertory dialers, capable of sending a switch-hook flash, detecting dial tone, and then sending feature codes, all at the push of a single button, came on the market. Built into conventional sets, they simplified PBX feature access and also made "speed calling" available to the station user without affecting the PBX in any way. However, to handle multiple line requirements, such phones had to have separate access to two or more ports on the switching matrix.

The rotary dial was invented to work with Strowger SXS switches; it came from Automatic Electric (now part of GTE) because the Bell System did not, for about 30 years after the invention of automatic switching, have an automatic system of its own. A dial makes momentary opens in the path of current-flow from the PBX or CO. A relay or other inexpensive device detects these opens (one open for the digit 1, two opens for the digit 2, etc., up to ten opens for the digit 0). The system has to use timing to detect the difference between a pulse, a flash and a hang-up, and additional timing to differentiate the interpulse interval from the interdigital interval and the time when all digits are received. This was done with three relays in the old days, and uses a rather elaborate computer program today.

Rotary dials go at 10 pulses per second (PPS), and are tricked up to insure something over half a second between digits. These specifications are required by all SXS switches, but are not needed by switches with a register to receive the digits and a separate control system to reach out and make the appropriate connection. The relatively large mass of the SXS switch needed time to move from level to level on each pulse and then, after timing and identifying an interdigital interval, to hunt over ten possible paths, one after another, to the next rank of switches. It is not clear why its requirements are imposed on modern electronic PBXs (or modern computer controlled COs, for that matter).

DTMF (Touch-Tone) signaling is tending to replace rotary dials today. A DTMF key-pad generates two tones, one from a low group and one from a high group. With four tones in each group, 16 combinations are possible and 12 are normally found on a DTMF key-pad. The tones, in the voice-frequency band, are carefully selected to minimize their resemblance to human speech, but additional techniques are required to prevent "talk offs" in the digit receivers used to convert DTMF whistles into something switch controls can use.

DTMF is supposed to save users a lot of time. The actual saving is about five seconds per call; for an ordinary user making, say, 20 calls a day, this comes to 100 seconds or about half the time it takes to smoke one cigarette. The only place where DTMF "speed" actually earns its keep is in digit receivers in central offices and at telephone switchboards or consoles where an attendant dials hundreds of calls a day. In each instance, there are enough calls to permit the accumulated time to mount up to something reasonable, and speed in dialing each call is important. But watch out for PBX consoles. Anybody who uses a pad to complete a lot of calls gets pretty fast at keying. Indeed, such a person can easily outrun the timing circuits in the DTMF receiver that prevent voice simulations (talk-offs) from being mistaken for valid digits. Even the Dimension PBX, from the same Bell Labs that invented Touch-Tone, uses digital signaling from its console.

DTMF is good in that it does not break long fingernails. Further, it is a dandy status symbol. However, its most useful feature is its ability to go through the telephone network, end to end. This allows a conventional DTMF phone to be a sort of data terminal, and a variety of schemes have been devised to permit accessing computer data-bases, particularly when the computer is provided with a voice synthesizer so that it can talk back to the caller. Electronic banking has made some use of DTMF in this sense, but the principal beneficiaries of DTMF’s through-signaling capability are the specialized common carriers and resellers such as MCI, Sprint, SBS, etc. Until "equal access" becomes universal, these various companies should be thankful for AT&T's innovation in providing a means for customers to dial up, identify themselves, and then send the called number to them through the public telephone network.

Calling someone to the telephone requires ringing. Mr. Watson’s ringer, invented before the turn of the century, still reigns supreme and represents one of the main problems of using conventional phones on electronic PBXs. Ringing is a power signal, 86 volts at 20 Hz. This is almost the size and one-third the frequency of the power that comes out of the wall to run your electric iron or toaster. It cannot be switched through a digital switching matrix from a common source to a called line, and, indeed, only AT&T's 5ESS has an electronic space division crosspoint capable of handling this dinosaur. Thus it must be applied via a relay or other power-handling device on a per-line basis, adding greatly to the cost of electronic switching,

On the good side, it is readily differentiated from voice signals (appreciably less than one volt in amplitude and in the frequency range between 300 and 3000 Hz.) and does not resemble the battery voltage (50 volts, DC) which may be pulsed by rotary dials. Further, it has enough power to operate directly specific devices such as ring-up relays; these, in turn, can light switchboard lamps, turn on telephone answering machines, etc. Power ringing is deeply embedded in the real world, even if it is not compatible with electronic switching. It represents a de facto standard, but we might have wished for a standard invented a little later than 1890.

In any event, conventional telephones require power ringing; to be able to use a $20 phone ($40 if DTMF replaces a rotary dial) in a thousand dollar a line PBX, we hark back to the 19th century. Tone ringing, capable of passing through an electronic matrix as easily as speech or DTMF, is well understood and can be made available, but it may be that digital signaling including a code that tells the set to operate its sounder will get here first for all practical purposes.

Conventional ringers hang across the line. In some systems, they are disconnected by taking the hand-set off hook, but this is not always the case. When bridged stations are used (two or more phones on the same line), the ringer in the other set remains across the line regardless of the switch-hook status in the answering set. Ringers are usually resonant at about 20 Hz, draw only a small current during operation, and are a very high impedance at voice frequencies. Thus they have little effect on speech and, unlike tone ringers which respond to signals in the voice band, speech has no effect on them.

Although they can be made to work well, it should be obvious that "standard" 500/2500 type telephone sets leave much to be desired.

Key telephone sets

Key telephone sets, identified by the keys or buttons that permit connection to one of several lines (and not by the key pad used to transmit DTMF address signals), have the basic 500/2500 set components plus some additional equipment. Normally accessed through 25-pair or larger cable, they allow for a hold button (the first and most important of the feature buttons we will consider), private intercoms with associated signaling, and direct selection of one or more specific lines. These lines may be PBX extensions, CO lines, or intercom lines internal to the key system itself. They may even be tie trunks or foreign exchange (FX) lines.

The buttons on a key telephone set are illuminated to tell the status of each line (busy, idle, ringing or on-hold). This is a great help to people using the system. Indeed, the simplicity of operation of a key system recommends it to almost everyone with complex communication needs; for many years, key telephone systems provided the main user features required in business telecommunications.

In conventional key equipment, each button actually makes a physical connection to a line or other facility, inside the telephone set itself. Each line has its own hold relay, lamp control equipment, etc., mounted on a circuit board called a KTU (key telephone unit) which plugs into a KSU (key service unit). To let a particular phone pick up a particular line, that line must come to the appropriate button. Thus, a cross connect panel is required between the KTUs and the multi-pair cables to each phone; a craftsman must make the modifications in station wiring when different lines are to be picked up or different features to be accessed. This is costly in terms of both labor and administration, and is the main disadvantage of 1A2 and similar key systems.

Another advantage of traditional key equipment is that three lines, say, can serve six or seven people with about the same grade of service that six lines can serve them if they have individual phones. Actually, three lines in a hunt group, shared, provide better service; a second call coming in can be answered by someone else and a message taken, and a second line is available to another outgoing call to get information while the first call is on hold. In any event, the bridged stations do not add very much in cost, and they permit a somewhat smaller number of terminals on the switching matrix to serve a given number of people. When replacing a PBX using 1A2 key equipment, it is usually necessary to allow for about 20% more matrix terminals for the same number of users.

Some modern PBXs build in key system control equipment (the KTU), and permit existing phones, including 1A2 key, to be used directly. This can be a powerful marketing argument when the customer already owns the telephones on the system being replaced. Further, because most of the control is in the PBX’s software, PBX administrative systems can make some use of program capability in handling moves, changes, feature upgrades, etc. Adding conventional key to modern PBXs is a mixed blessing, but it gives the customer some added flexibility in making a choice.

Electronic telephone sets

The current trend is to proprietary telephone sets designed to work directly with the PBX to combine PBX capabilities with the user convenience of 1A2 key. Such sets, available from Northern Telecom, AT&T, Rolm, Mitel and others, have a talk path, a signaling path, and a power path from the set to the PBX. At the set, they usually have a group of buttons that can be used either for line pick-up (as in conventional key telephone sets) or feature invocation. The meaning of each button on each set is defined in the PBX’s memory; when a button is pushed, its identity is sent to system control over the signaling path, and the required operation read out of memory. If a line-select button is pushed, the system connects an incoming call to the talk path to the set via the switching matrix; on an outgoing call, charging and restriction are based on the class of service of the line selected. If a feature key has been pushed (the hold button, for instance, although we now have many other possibilities), the system does whatever is required. If visual cues are required by the user, the PBX control sends the appropriate signal down the signaling path to light or blink the appropriate lamp, cause the tone ringer to sound, or provide some other information. Many proprietary sets today have fairly elaborate alphanumeric displays, as will be discussed.

These modern electronic sets have a number of advantages. First, they are much easier to use when contrasted with flashing the switch-hook, dialing a number of feature codes, and then identifying a variety of call progress tones. Second, they normally use two or three pairs, and three pairs will work on any kind of phone from a simple single line set to a 30-button answering position. This is a great improvement over 1A2 key, where 25 pair cables sometimes have to be replaced with 75 pair cables. Finally, they permit complete program control of changes in lines picked up, features available, etc. This kind of control can often be handled for remote locations via a data terminal in a dial-up connection, used by the customer's communication manager.

The saving in "OCC" (other charges and credits) for moves and changes can be considerable in an active location. Further, when the communication manager can reduce the time it normally takes to write the order, and then be relieved of the aggravation of trying to honcho the order through a reluctant vendor, "productivity improvement" takes on a whole new dimension.

One problem must be noted, however. Single line sets usually require different line cards in the PBX than do these modern sets. Thus more than a program change is required when one goes from a single line to a multi-line set. Further, it is evident that when one changes from a 5-button set to a 30-button set, the sets themselves will have to be interchanged even though programming can do most of the rest. Button identification, too, may have to be done at the set, although "soft function keys" are beginning to attack this problem. In any event, administrative savings and user convenience are the main arguments to justify the cost of these proprietary sets.

Data via proprietary sets

Data should be switched over public and private voice networks like any other signal. On a more modest scale, within any given location, the PBX is an ideal vehicle for moving electrical information around. Just how this could be done was demonstrated first by Danray and then taken up by Northern Telecom upon their purchase of Danray; others have followed using a variety of implementation techniques. One simply puts an RS-232C interface on an electronic telephone set (there are better and less expensive interfaces, but 232 exists in vast quantities) and plugs in a terminal or a computer.

Danray used the power pair (one of the three pairs required by their electronic set) to carry off 9,600 bps full-duplex data to a separate auxiliary data switch. The voice telephone was used to set up the connection, but once the data path was set up, the voice equipment was free to make or receive calls. This kind of simultaneous voice/data operation seems to be highly favored by those who normally use terminals, particularly when compared with some version of alternate voice/data where the voice phone is tied up with the data connections.

Northern Telecom uses the signaling channel for their SL-1 phone (one of its three pairs) to move data. The actual signaling information between set and line card is quite small, so data and signaling can share the same pair of wires. Because the SL-1 switching matrix is digital, a separate data matrix is not required as with the analog Danray. However, voice and data from one set require two appearances on the matrix, and the data path, like the voice path, is the full 64 Kbps T-Carrier signal in each direction.

Rolm, with a 144 Kbps per voice channel internal to the switching matrix rather than the 64 Kbps standard, uses a different approach. Carving out one voice channel, they submultiplex it to handle up to 40 data connections and make very efficient use of the bandwidth available. To get to the user's work station, they originally used two pairs (one for each direction), and located the RS-232C interface in a box that looked like a modem but actually used digital techniques between the work location and the digital line card. In their later approaches, they use a single pair to the work location, transmit voice and data simultaneously in both directions (using a technique called Manchester encoding), and separate incoming from outgoing with hybrid circuits at each end. See Figure 8.

[ Figure 8: Rolm’s hybrid approach to full duplex voice/data on a single pair. ]

Northern Telecom, in their "Meridian" announcement in February 1985, also went to single pair wiring, but used the ping-pong approach (known more formally as time compression multiplexing) as shown in Figure 9. Ping-pong works by transmitting 8 bits for voice, 8 for data and a couple of extra bits for signaling and synch from PBX to set, and then a similar "word" from set to PBX. A guard interval is left between the ping and the pong so that they do not overlap. Transmission is at the rate of 256 Kbps while the information is moving, but part of the time it is going one way, part the other, and, during the guard Intervals, there is silence.

[ Figure 9: The ping pong approach to voice and data on a single pair. Similar to ISDN 2B+D. ]

AT&T, Ericsson, Mitel, InteCom, Lexar, GTE and others have used a variety of techniques, both to get from the workstation to the switch, and to make the interconnection from one matrix terminal to another. See the individual system descriptions, and note the incredible number of ways the job can be done.

Many people have suggested that a telephone set in a business area should contain a full alphanumeric keyboard and display for non-voice communication, and a handset for voice calls. Line and feature activation can be carried out with "soft function keys" defined as needed at a particular time by software, and described for the user on a related general-purpose display. Northern Telecom's Displayphone and Rolm's Cypress are examples of such general-purpose telephone sets.

However, technology is moving rapidly these days, and full micro-processor-based computer systems are available for a fraction of the cost of a proprietary voice/data terminal. Personal computers, general-purpose machines useful for whatever office function is desired as long as the proper floppy disk is in place, are proliferating faster than the slow moving telephone industry can design terminals. Thus the personal computer may well turn out to be the standard business communication instrument of the future. All it needs is a standard interface to take all the marbles.

ISDN Standards

The standard interface may be here, offering escape from the 1950 500/2500 standard with its 1890 ringing, and the incredible variety of gold plated proprietary sets and terminals that cost two or three times as much per line as the PBX and only work on the PBX they were designed to interface. ISDN may be the way out.

At present, ISDN offers two standards, referred to loosely as 23B+D for connecting two switches such as a PBX and a CO, and 2B+D for connecting a PBX or CO to a new-standard telephone set. The "B" in each specification refers to a 64 Kbps T-compatible channel, 8,000 8-bit bytes, full duplex, in each direction. The "D" designates a signaling channel. In 23B+D, it is a 64 Kbps CCIS channel using the CCITT No. 7 international standard protocol, and the whole thing is seen to be a standard North American 24-channel T-carrier digroup (see Section V of this chapter). In 2B+D, the D channel consists of 2 bits for signaling, supervision, and packet switching to the set, 8,000 times a second. With 8 bits for voice and 8 bits for data and 2 bits for signaling, the bit rate in each direction is 144 Kbps; however, since time compression multiplexing with guard intervals between pings and pongs is used, the actual speed during transmission will be a little more than twice that.

We now have a chance to make ISDN interface boards or boxes for our personal computers and telephone sets, and have reason to hope, in the not too distant future, for digital connectivity with bit and byte integrity maintained end-to-end. And manufacturers have an opportunity to make truly modern telephone sets, not just for their PBXs, but for all PBX, CO, and Centrex switches that meet the standard. It shouldn't be too hard to modify most of the present proprietary sets to the standard, either, now that a standard more or less exists. And even where the PBX and its proprietary tel-set cannot be adapted, it would seem that a T-compatible PBX could Interface the rest of the world via 23B+D and still offer a high degree of flexibility in the people and machines it can reach. Keep your fingers crossed.

Consoles

Consoles used with PBXs are an interface to a human who can provide assistance when required to complete calls or execute other services. Early consoles connected to trunks directly as shown in Figure 10 and, working through the trunk appearances on the switching matrix, completed calls. Today, the trend is to use something very much like an electronic telephone set as a console (see Figure 11); the switch connects calls needing assistance to the console as to any other telephone, and provides the specialized information required by driving suitable displays. Consoles, like electronic telephone sets, usually have digital signaling paths to the system control to interchange information directly without using the voice path. Companies like Mitel and CXC are actually using CRT-based terminals as consoles. The general-purpose display can be used, among other things, for presenting directory information and for storing and recalling messages.

[ Figure 10: Direct trunk appearances on a key per trunk attendant console. ]

[ Figure 11: Switched loop operation of attendant console. ]

Specialized consoles are also provided, notably in the hotel/motel business. Message unit charging by room for local calls is now stored in the PBX's electronic memory; message register displays, replacing the electromechanical registers that used to lurk behind all hotel cashier desks in the old days, are now common. Further, these consoles can change the class of service of the room phone to permit outside calls when the guest checks in, and to block outside calls when the guest checks out. It is not too hard to activate and deactivate the room television and air conditioner at the same time. Energy management is now being offered in several PBXs, and may turn out to be an important factor in overall facilities management in the not too distant future.

Calling number displays are standard on main consoles, but some PBXs have separate consoles just to display the calling number to facilitate room service operations, etc. Other systems have separate consoles to handle messages and send a message-waiting signal toward the phone to notify the guest. In the last few years, electronic telephone sets and CRT-based phone-terminals have opened new markets and new possibilities for messaging; message waiting is no longer trapped in hotel-motel service alone, but is often considered a standard business feature.

Maintenance consoles are somewhat similar, but access the PBX differently. There is often a data bus associated with the control that can interface one or more data terminals or printers. With a modem to an outside telephone line (or one of the CO trunks on a throw-over basis) in place of a local terminal, a dial-up connection from a terminal at a remote maintenance center can be set up quickly. It is often very useful to be able to call in the factory experts when the local maintenance people get in over their heads.

Another approach to maintenance and administration, pioneered by Harris, NEC, Solid State Systems and others, is to use a personal computer for system access. With software from the PBX manufacturer running on the PC, the entire PBX data base can be called up, tested, modified, edited, etc. Then, after the changes in station features, directory, ARS, etc., are entered into the PC, the PC, interfacing the PBX control via an RS-232C port, downloads the new information. The old data-base is stored just in case something goes wrong, and can be reloaded in an emergency.

Given access to a data bus that runs through the control processor, it is not very hard to add on extra features running on "applications processors." Call detail recording systems, running on floppy-disk based personal computers, were in this game very early. Plugging into a port formerly used by a tape recorder, these small systems process the call records formatted by the PBX and passed on to them. The communication manager thus has available individual station bills, along with reports sorted in a variety of ways for overall budgeting and system administration.

The telephone directory, energy management, messaging systems (both voice and text), data protocol conversions, etc., etc., are today easily incorporated into a PBX to combine its universal access to different parts of the company with inexpensive modern machinery for data manipulation.

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Copyright 2006 Lee Goeller. All Rights Reserved.