AT&T MERLIN LEGEND Release 2.0 Analog Multiline Telephone Equipment And Operations Manual

AT&T 555-620-115

Issue 1

October 1992

MERLIN LEGEND

TM

Communications System

Release 2.0

Equipment and Operations Reference

Copyright © 1992 AT&T

All Rights Reserved

Printed in U.S.A.

AT&T 555-620-115

Issue 1

October 1992

Notice

Every effort was made to ensure that the information in this printing. However, information is subject to change.

book was complete and accurate at the time of

Federal Communications Commission (FCC)

Electromagnetic Interference Information

This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful interference, in which case the user will be required to correct the interference at his own expense.

Canadian Department of Communications (DOC)

Interference Information

This digital apparatus does not exceed the Class A limits for radio noise emissions set out in the radio interference regulations of the Canadian Department of Communications.

Le Présent Appareil Numérique n'émet pas de bruits radioélectriques dépassant Ies Iimites applicable aux appareils numériques de la class A prescribes clans Ie Réglement sur Ie brouillage radioélectrique

édicté par Ie ministère des Communications du Canada.

Trademarks

5ESS, ACCUNET, CONVERSANT, Dimension, Horizon, Magic On Hold, MEGACOM, MERLIN, SYSTIMAX, and MultiQuest are registered trademarks and 4ESS, AUDIX Voice Power, FAX Attendant System,

InnManager, MERLIN LEGEND, MERLIN MAIL, MLX-10, MLX-10D, MLX-20L, MLX-28D, and MERLIN PFC are trademarks of AT&T in the U.S. and other countries.

Intel is a registered trademark of Intel Corporation.

MS-DOS is a registered trademark of Microsoft Corporation.

UNIX is a registered trademark of UNIX System Laboratories, Inc.

PagePac is a registered trademark and ZoneMate is a trademark of DRACON, a division of Harris

Corporation.

Mirage, StarSet, Supra, and Supra NC are registered trademarks of Plantronics, Inc.

Support Telephone Number

AT&T provides a toll-free customer Helpline (1 -800-628-2888) 24 hours a day (U.S.A. only). Call the

Helpline, or your authorized dealer, if you need assistance when installing, programming, or using the system.

Contents

1

2

3

About This Book

■ Terms and Conventions Used

■ Product Safety Labels

■ Security

■ Related Documents

■ How to Comment on This Document

Introduction

■ Components

■ Functional Description

■ Modes of Operation

■ Programming

■ System Capacities and Requirements

■ Release Differences

Hardware Components

■ Control Unit

■ Telephones and Consoles

■ Adapters and Adjuncts

■ Power-Related Hardware

Lines and Trunks

■ Loop-Start Lines/Trunks

■ Ground-Start Lines/Trunks

■ Tie Trunks

■ Direct Inward Dialing (DID) Lines/Trunks

■ DS1 Facilities

3-1

3-1

3-2

3-2

3-11

3-16

2-1

2-1

2-13

2-28

2-51

3

3

5

2

1

3

1-1

1-2

1-9

1-12

1-30

1-33

1-43

Table of Contents i

5

Contents

4

Applications

■ Voice Messaging Systems

■ MERLIN MAIL Voice Messaging System

■ MERLIN Attendant

■ Call Accounting System (CAS)

■ Call Accounting Terminal (CAT)

■ Call Management System (CMS)

■ InnManager Guest Management System

■ System Programming and Maintenance (SPM)

■ Integrated Solution II (IS II)

■ Integrated Solution Ill (IS Ill)

■ Primary Rate Interface (PRI) Applications

■ Centrex Operation

■ MERLIN PFC Telephone

■ Automated Document Delivery System (ADDS)

■ CONVERSANT Intro

■ Applications Printers

Data Communications Support

■ Data Communications Configuration Overview

■ Outside Trunks

■ System Features Used For Data

■ Endpoint Communications Features

■ PRI Applications

5-1

5-2

5-13

5-16

5-18

5-18

4-26

4-30

4-31

4-33

4-40

4-50

4-52

4-56

4-58

4-60

4-62

4-1

4-4

4-9

4-16

4-20

4-23

ii Table of Contents

Contents

A

GL

IN

Product Ordering Information

Glossary

Index

A-1

GL-1

IN-1

Table of Contents

Contents

iv Table of Contents

Figures

1

2

Introduction

Figure 1-1.

System Components

Figure 1-2.

Figure 1-3.

Figure 1-4.

Functional Units

Lines Labeled for Key System Telephones

Figure 1-5.

Lines Labeled for Modified Key System

Telephones

Hybrid/PBX Mode

Figure 1-6.

Figure 1-7.

Figure 1-8.

Behind Switch Mode

Behind Switch Mode with Direct Outside

Trunks

Labeled Line Buttons for Behind Switch

Telephones


Figure 2-1, Carriers

Figure 2-2.

Processor Module

Figure 2-3.

Power Supply

Figure 2-4.

Line/Trunk and Station Modules

Figure 2-5.

Control Unit Cover

Figure 2-6.

MLX-28D Telephone

Figure 2-7.

MLX-20L Telephone

Figure 2-8.

MLX-10D Telephone

Figure 2-9.

MLX-10 Telephone

Figure 2-10. Direct Station Selector

Figure 2-11. 551 T1 L1 Channel Service Unit Connections

Figure 2-12, ESF T1 Channel Service Unit Connections

Figure 2-13. Multi-Function Module

Figure 2-14. GPA Connections

Figure 2-15. 7500B Data Module Front Panel

Figure 2-16. 7500B Data Module Back Panel

Figure 2-17. SAA Connections

Figure 2-18. Headsets

Figure 2-19. Analog Multi line Telephone Headset

Figure 2-20. Analog IROB Connection

Figure 2-21, MLX IROB Connection

2-2

2-3

2-4

2-6

2-10

2-15

2-16

2-17

2-18

2-19

2-32

2-33

2-34

2-36

2-38

2-39

2-43

2-48

2-49

2-54

2-55

1-6

1-10

1-15

1-17

1-19

1-23

1-24

1-25

Table of Contents v

Figures

3

5

Figure 2-22.

Figure 2-23.

Figure 2-24.

Figure 2-25.

Surge Protectors

Trouble Alarm Connections

Power Failure Alarm Connections

Power Failure DID Busy-Out Connections


Figure 3-1.

Setting the 400EM Module DIP Switches for

E&M Signaling Types 1C and 5

Figure 3-2.

Nontandem Tie-Trunk Network


Figure 5-1.

Individual Use Data Station Configurations

Figure 5-2.

Direct RS-232 Interface

Figure 5-3.

RS-232 to V.35 Interface Conversion

Figure 5-4.

Direct V.35 Interface

Figure 5-5. Video Conferencing Connections

2-57

2-59

2-59

2-60

3-6

3-11

5-3

5-20

5-22

5-24

5-26

vi Table of Contents

Tables

1

2

3

4

Introduction

Table 1-1, Modes of Operation Summary

Table 1-2.

FCC Registration Numbers

Table 1-3.

Hardware and Software Capacities

Table 1-4.

Environmental Specifications


Table 2-1.

Line/Trunk and Station Modules

Table 2-2.

Table 2-3.

Table 2-4,

Table 2-5.

Table 2-6.

Table 2-7.

Table 2-8.

Table 2-9.

Reusable MERLIN II Modules

Reusable MERLIN II Hardware

Analog Multiline Telephones

Single-Line Telephones

Telephones and Adjuncts Not Supported

Maximum Number of System Operator

Positions

Adjunct Summary

Local Auxiliary Power Requirements

1-12

1-28

1-34

1-37


Table 3-1.

Setting the 400EM Module DIP Switches

Table 3-2.

Sample DIP Switches for the 400EM Module

Table 3-3.

Tie-Trunk Compatibility

Table 3-4.

Type 1 Standard and Type 1 Compatible

E&M Switch Settings

Table 3-5.

Line Compensation Settings

Applications

Table 4-1.

Application Capacities and Modes of

Operation

Table 4-2.

Mode Codes

4-3

4-6

Table 4-3.

TTRs Required by Voice Messaging Systems 4-7

Table 4-4.

MERLIN MAIL Voice Messaging System

Ports Required 4-15

3-7

3-7

3-9

3-1o

3-20

2-7

2-11

2-12

2-21

2-22

2-23

2-24

2-28

2-52

Table of Contents vii

Tables

5

Table 4-5.

MERLIN Attendants Required

Table 4-6.

Voice Channels Required

Table 4-7.

Voice Channels Required

Table 4-8.

Applications Printers


Table 5-1.

Configurations of Data Stations

4-19

4-35

4-49

4-62

5-9

viii Table of Contents

he exclamation point in an equilateral triangle is intended to alert the user to the presence of important operating and maintenance (servicing) instructions in the literature accompanying the product.

IMPORTANT SAFETY INSTRUCTIONS

When installing telephone equipment, always follow basic safety precautions to reduce the risk of fire, electrical shock, and injury to persons, including:

■ Read and understand all instructions.

■ Follow all warnings and instructions marked on or packed with the product.

■ Never install telephone wiring during a lightning storm,

■ Never install a telephone jack in a wet location unless the jack is specifically designed for wet locations.

■ Never touch uninsulated telephone wires or terminals unless the telephone wiring has been disconnected at the network interface.

■ Use caution when installing or modifying telephone lines.

■ Use only AT&T-manufactured MERLIN LEGEND™ Communications

System circuit modules, carrier assemblies, and power units in the

MERLIN LEGEND Communications System (511A) control unit.

■ Use only AT&T-recommended/approved MERLIN LEGEND

Communications System accessories.

■ If equipment connected to the analog station modules (008, 408,

408 GS/LS) or to the MLX telephone modules (008 MLX, 408 GS/LS-MLX) is to be used for in-range out-of-building (IROB) applications, IROB protectors are required.

■ Do not install this product near water, for example, in a wet basement location.

■ Do not overload wall outlets, as this can result in the risk of fire or electrical shock.

■ The MERLIN LEGEND Communications System is equipped with a three-wire grounding-type plug with a third (grounding) pin. This plug will fit only into a grounding-type power outlet. This is a safety feature, If you are unable to insert the plug into the outlet, contact an electrician to replace the obsolete outlet. Do not defeat the safety purpose of the grounding plug.

■ The MERLIN LEGEND Communications System requires a supplementary ground.

ix

■ Do not attach the power supply cord to building surfaces. Do not allow anything to rest on the power cord. Do not locate this product where the cord will be abused by persons walking on it.

■ Slots and openings in the module housings are provided for ventilation.

To protect this equipment from overheating, do not block these openings.

■ Never push objects of any kind into this product through module openings or expansion slots, as they may touch dangerous voltage points or short out parts, which could result in a risk of fire or electrical shock. Never spill liquid of any kind on this product.

■ Unplug the product from the wall outlet before cleaning. Use a damp cloth for cleaning. Do not use cleaners or aerosol cleaners. Auxiliary equipment includes answering machines, alerts, modems, and fax machines. To connect one of these devices, you must first have a

Multi-Function Module (MFM).

WARNING:

■ For your personal safety, DO NOT install an MFM yourself.

■ ONLY an authorized technician or dealer representative shall install, set options, or repair an MFM.

■ To eliminate the risk of personal injury due to electrical shock,

DO NOT attempt to install or remove an MFM from your MLX telephone. Opening or removing the module cover of your telephone may expose you to dangerous voltages.

SAVE THESE INSTRUCTIONS

Customer Support Information

Support Telephone Number

In the U.S.A. only, AT&T provides a toll-free customer Helpline

(1-800-628-2888) 24 hours a day. Call the Helpline, or your authorized dealer, if you need assistance when installing, programming, or using your system.

Outside the U. S. A., if you need assistance when installing, programming, or using your system, contact your authorized AT&T dealer.

Federal Communications Commission (FCC) Electromagnetic Interference

Information

This equipment has been tested and found to comply with the limits for a

Class A digital device, pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful interference to radio communications, Operation of this equipment in a residential area is likely to cause harmful interference, in which case the user will be required to correct the interference at his own expense.

Canadian Department of Communications (DOC) Interference Information

This digital apparatus does not exceed the Class A limits for radio noise emissions set out in the radio interference regulations of the Canadian

Department of Communications,

Le Présent Appareil Numérique n'émet pas de bruits radioelectriques depassant Ies Iimites applicable aux appareils numériques de la class A prescribes clans Ie reglement sur Ie brouillage radioelectrique edicté par Ie ministère des Communications du Canada.

FCC Notification and Repair Information

This equipment is registered with the FCC in accordance with Part 68 of its rules, In compliance with those rules, you are advised of the following:

Customer Support Information xi

Customer Support Information

■ Means of Connection. Connection of this equipment to the telephone network shall be through a standard network interface jack:

USOC RJ11C, RJ14C, RJ21X. Connection to E&M tie trunks requires a

USOC RJ2GX. Connection to off-premises stations requires a

USOC RJ11C or RJ14C. Connection to 1.544-Mbps digital facilities must be through a USOC RJ48C or RJ48X. Connection to DID requires a

USOC RJ11C, RJ14C, or RJ21X. These USOCs must be ordered from your telephone company.

This equipment may not be used with party lines or coin telephone lines.

■ Notification to the Telephone Companies. Before connecting this equipment, you or your equipment supplier must notify your local telephone company’s business office of the following:

—

The telephone number(s) you will be using with this equipment.

— The appropriate registration number and ringer equivalence number (REN), which can be found on the back or bottom of the control unit, as follows:

If this equipment is to be used as Key System, report the number AS593M-72914-KF-E.

If the system provides both manual and automatic selection of incoming/outgoing access to the network, report the number AS593M-72682-MF-E.

If there are no directly terminated trunks, or if the only directly terminated facilities are personal lines, report the number AS5USA-65646-PF-E.

The REN for all three systems is 1.5A.

—

For tie line connection, the facility interface code (FIC) is TL31 M and the service order code (SOC) is 9.0F.

— For connection to off-premises stations, the FIC is OLI3C and the SOC is 9.0F.

— For equipment to be connected to 1.544-Mbps digital service, the FIC is 04DU9-B for D4 framing format or 04DU9-C for extended framing format, and the SOC is 6.0P.

— For equipment to be connected to DID facilities, the FIC is

02RV2-T and the SOC is 9.0F.

— The quantities and USOC numbers of the jacks required.

— For each jack, the sequence in which lines are to be connected: the line types, the FIC, and the REN by position when applicable.

You must also notify your local telephone company if and when this equipment is permanently disconnected from the line(s).

The REN is used to determine the number of devices that may be connected to the telephone line. Excessive RENs on the line may result in the devices not ringing in response to an incoming call. In most, but not all, areas the sum of the RENs should not exceed five (5.0). To be certain of the number of devices that may be connected to the line, as determined by the total RENs, contact the telephone company to determine the maximum REN for the calling area.

xii Customer Support Information


Installation and Operational Procedures

The manuals for your system contain information about installation and operational procedures.

■ Repair Instructions. If you experience trouble because your equipment is malfunctioning, the FCC requires that the equipment not be used and that it be disconnected from the network until the problem has been corrected. Repairs to this equipment can be made only by the manufacturers, their authorized agents, or others who may be authorized by the FCC. In the event repairs are needed on this equipment, contact your authorized T&T dealer or, in the U.S.A. only, contact the National

Service Assistance Center (NSAC) at 1-800-628-2888.

■ Rights of the Local Telephone Company. If this equipment causes harm to the telephone network, the local telephone company may discontinue your service temporarily. If possible, they will notify you in advance. But if advance notice is not practical, you will be notified as soon as possible. You will also be informed of your right to file a complaint with the FCC.

Your local telephone company may make changes in its facilities, equipment, operations, or procedures that affect the proper functioning of this equipment. If they do, you will be notified in advance to give you an opportunity to maintain uninterrupted telephone service.

■ Hearing Aid Compatibility. The custom telephone sets for this system are compatible with inductively coupled hearing aids as prescribed by the FCC.

■ Automatic Dialers. WHEN PROGRAMMING EMERGENCY NUMBERS

AND/OR MAKING TEST CALLS TO EMERGENCY NUMBERS:

— Remain on the line and briefly explain to the dispatcher the reason for the call.

— Perform such activities in off-peak hours, such as early morning or late evening.

■ Direct Inward Dialing (DID).

a. This equipment returns answer supervision signals to the Public

Switched Telephone Network when:

(1)

(2)

(3) answered by the called station answered by the attendant routed to a recorded announcement that can be administered by the customer premises equipment user

(4) routed to a dial prompt b. This equipment returns answer supervision on all DID calls forwarded back to the Public Switched Telephone Network,

Permissible exceptions are when:

(1) a call is unanswered

(2) a busy tone is received

(3) a reorder tone is received

Customer Support Information xiii


Allowing this equipment to be operated in such a manner as not to provide proper answer supervision signaling is in violation of

Part 68 rules.

DOC Notification and Repair Information

NOTICE: The Canadian Department of Communications (DOC) label identifies certified equipment. This certification means that the equipment meets certain telecommunications network protective, operational, and safety requirements.

The DOC does not guarantee the equipment will operate to the user’s satisfaction.

Before installing this equipment, users should ensure that it is permissible to connect it to the facilities of the local telecommunications company. The equipment must also be installed using an acceptable method of connection.

In some cases, the company’s inside wiring for single-line individual service may be extended by means of a certified connector assembly (telephone extension cord). The customer should be aware that compliance with the above conditions may not prevent degradation of service in some situations.

Repairs to certified equipment should be made by an authorized Canadian maintenance facility designated by the supplier. Any repairs or alterations made by the user to this equipment, or any equipment malfunctions, may give the telecommunications company cause to request the user to disconnect the equipment.

Users should ensure for their own protection that the electrical ground connections of the power utility, telephone lines, and internal metallic water pipe system, if present, are connected. This precaution may be particularly important in rural areas.

CAUTION: Users should not attempt to make such connections themselves, but should contact the appropriate electrical inspection authority or electrician, as appropriate.

To prevent overloading, the Load Number (LN) assigned to each terminal device denotes the percentage of the total load to be connected to a telephone loop used by the device. The termination on a loop may consist of any combination of devices subject only to the requirement that the total of the Load

Numbers of all the devices does not exceed 100.

DOC Certification No. 230 4095A

CSA Certification No. LR 56260

Load No. 6

Renseignements sur la notification du ministère des Communications du

Canada et la réparation

AVIS: L’étiquette du ministère des Communications du Canada identifie Ie matériel homologué. Cette étiquette certifie que Ie matériel est conforme à certaines normes de protection, d’exploitation et de sécurité des réseaux de télécommunications. Le Ministère n’assure toutefois pas que Ie matériel fonctionnera à la satisfaction de I’utilisateur.

Avant d’installer ce matériel, I’utilisateur doit s’assurer qu’il est permis de Ie raccorder aux installations de I’entreprise locale de télécommunication. Le xiv Customer Support Information

Customer Support Information matériel doit également être installé en suivant une méthode acceptée de raccordement. Dans certains cas, Ies fils intérieurs de I’enterprise utilisés pour un service individual à Iigne unique peuvent être prolongés au moyen d’un dispositif homologué de raccordement (cordon prolongateur téléphonique interne). L’abonné ne doit pas oublier qu’il est possible que la conformité aux conditions énoncées ci-dessus n’empêchent pas la degradation du service clans certaines situations. Actuellement, Ies entreprises de télécommunication ne permettent pas que I’on raccorde Ieur matériel à des jacks d’abonné, sauf clans Ies cas précis prévus pas Ies tarifs particuliers de ces entreprises.

Les réparations de matériel homologué doivent être effectuées par un centre d’entretien canadien autorisé désigné par Ie fournisseur. La compagnie de télécommunications peut demander à I’utilisateur de débrancher un appareil à la suite de reparations ou de modifications effectuées par I’utilisateur ou à cause de mauvais fonctionnement.

Pour sa propre protection, I’utilisateur doit s’assurer que tous Ies fils de mise à la terre de la source d’énergie électrique, des Iignes téléphoniques et des canalisations d’eau métalliques, s’il y en a, sent raccordés ensemble. Cette precaution est particulièrement importance clans Ies régions rurales.

AVERTISSEMENT: L’utilisateur ne doit pas tenter de faire ces raccordements lui-même; il doit avoir recours à un service d’inspection des installations

électriques, ou à un electrician, selon Ie cas.

L’indite de charge (IC) assigné à chaque dispositif terminal indique, pour éviter toute surcharge, Ie pourcentage de la charge totale qui peut être raccordée à un circuit téléphonique bouclé utilisé par ce dispositif. La terminaison du circuit bouclé peut être constitute de n’importe quelle combinaison de dispositifs, pourvu que la somme des indices de charge de I’ensemble des dispositifs ne dépasse pas 100.

No d’homologation: 230 4095A

Node certification: CSA LR 56260

L’indite de charge: 6

Customer Support Information xv


MERLIN LEGEND D.O.C. Ministère des Communications

Location Label Placement du Canada emplacement de l'étiquette

xvi Customer Support Information


Security of Your System—Preventing Toll Fraud

As a customer of a new telephone system, you should be aware that there exists an increasing problem of telephone toll fraud. Telephone toll fraud can occur in many forms, despite the numerous efforts of telephone companies and telephone equipment manufacturers to control it. Some individuals use electronic devices to prevent or faIsify records of these calls. Others charge calls to someone else’s number by illegally using lost or stolen calling cards, billing innocent parties, clipping on to someone else’s line, and breaking into someone else’s telephone equipment physically or electronically. In certain instances, unauthorized individuals make connections to the telephone network through the use of remote access features.

The Remote Access feature of your system, if you choose to use it, permits offpremises callers to access the system from a remote telephone by using an 800 number or a 7- or 10-digit telephone number. The system returns an acknowledgement signaling the user to key in his or her authorization code, which is selected and administered by the system manager. After the authorization code is accepted, the system returns dial tone to the user. If you do not program specific egress restrictions, the user will be able to place any call normally dialed from a telephone associated with the system. Such an offpremises network call is originated at, and will be billed from the system location.

The Remote Access feature, as designed, helps the customer, through proper administration, to minimize the ability of unauthorized persons to gain access to the network. Most commonly, phone numbers and codes are compromised when overheard in a public location, through theft of a wallet or purse containing access information, or through carelessness (writing codes on a piece of paper and improperly discarding it). Additionally, hackers may use a computer to dial an access code and then publish the information to other hackers. Enormous charges can be run up quickly. It is the customer’s responsibility to take the appropriate steps to properly implement the features, evaluate and administer the various restriction levels, protect access codes, and distribute access codes only to individuals who have been fully advised of the sensitive nature of the access information.

Common carriers are required by law to collect their tariffed charges. While these charges are fraudulent charges made by persons with criminal intent, applicable tariffs state that the customer of record is responsible for payment of all long-distance or other network charges. AT&T cannot be responsible for such charges and will not make any allowance or give any credit for charges that result from unauthorized access.

To minimize the risk of unauthorized access to your communications system:

■ Use a nonpublished Remote Access number.

■ Assign authorization codes randomly to users on a need-to-have basis, keeping a log of ALL authorized users and assigning one code to one person.

Customer Support Information xvii


■ Use random sequence authorization codes, which are less likely to be easily broken.

■ Deactivate all unassigned codes promptly.

■ Ensure that Remote Access users are aware of their responsibility to keep the telephone number and any authorization codes secure.

■ When possible, restrict the off-network capability of off-premises callers, via use of Call Restrictions and Disallowed List capabilities.

■ When possible, block out-of-hours calling.

■ Frequently monitor system call detail reports for quicker detection of any unauthorized or abnormal calling patterns.

■ Limit Remote Call Forward to persons on a need-to-have basis.

Limited Warranty and Limitation of Liability

AT&T warrants to you, the customer, that your MERLIN LEGEND

Communications System will be in good working order on the date AT&T or its authorized reseller delivers or installs the system, whichever is later (“Warranty

Date”). If you notify AT&T or its authorized reseller within one year of the

Warranty Date that your system is not in good working order, AT&T will without charge to you repair or replace, at its option, the system components that are not in good working order. Repair or replacement parts may be new or refurbished and will be provided on an exchange basis. If AT&T determines that your system cannot be repaired or replaced, AT&T will remove the system and, at your option, refund the purchase price of your system, or apply the purchase price towards the purchase of another AT&T system.

If you purchased your system directly from AT&T, AT&T will perform warranty repair in accordance with the terms and conditions of the specific type of AT&T maintenance coverage you selected. If you purchased your system from an

AT&T-authorized reseller, contact your reseller for the details of the maintenance plan applicable to your system.

This AT&T limited warranty covers damage to the system caused by power surges, including power surges due to lightning.

The following will not be deemed to impair the good working order of the system, and AT&T will not be responsible under the limited warranty for damages resulting from

■ failure to follow AT&T’s installation, operation, or maintenance

■ instructions unauthorized system modification, movement, or alteration

■ unauthorized use of common carrier communication services accessed through the system

■ abuse, misuse, or negligent acts or omissions of the customer and persons under the customer’s control

■ acts of third parties and acts of God

AT&T’S OBLIGATION TO REPAIR, REPLACE, OR REFUND AS SET FORTH

ABOVE IS YOUR EXCLUSIVE REMEDY.

EXCEPT AS SPECIFICALLY SET FORTH ABOVE, AT&T, ITS AFFILIATES,

SUPPLIERS, AND AUTHORIZED RESELLERS MAKE NO WARRANTIES,

EXPRESS OR IMPLIED, AND SPECIFICALLY DISCLAIM ANY WARRANTIES OF

MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

xviii Customer Support Information


Limitation of Liability

EXCEPT FOR PERSONAL INJURY, DIRECT DAMAGES TO TANGIBLE

PERSONAL PROPERTY PROXIMATELY CAUSED BY AT&T, AND LIABILITY

OTHERWISE EXPRESSLY ASSUMED IN A WRITTEN AGREEMENT SIGNED BY

AT&T, THE LIABILITY OF AT&T, ITS AFFILIATES, SUPPLIERS, AND

AUTHORIZED RESELLERS FOR ANY CLAIMS, LOSSES, DAMAGES, OR

EXPENSES FROM ANY CAUSE WHATSOEVER (INCLUDING ACTS OR

OMISSIONS OF THIRD PARTIES), REGARDLESS OF THE FORM OF ACTION,

WHETHER IN CONTRACT, TORT OR OTHERWISE, SHALL NOT EXCEED AN

AMOUNT EQUAL TO THE LESSER OF THE DIRECT DAMAGES PROVEN OR

THE PURCHASE PRICE OF THE SYSTEM. IN NO EVENT SHALL AT&T OR ITS

AFFILIATES, SUPPLIERS, OR AUTHORIZED RESELLERS BE LIABLE FOR

INCIDENTAL, RELIANCE, CONSEQUENTLY, OR ANY OTHER INDIRECT LOSS

OR DAMAGE (INCLUDING LOST PROFITS OR REVENUES) INCURRED IN

CONNECTION WITH THE SYSTEM. THIS LIMITATION OF LIABILITY SHALL

SURVIVE FAILURE OF THE EXCLUSIVE REMEDY SET FORTH IN THE LIMITED

WARRANTY ABOVE.

Voice Mail Systems

Your Voice Mail system permits callers to leave verbal messages for system users or gain access to the back-up position in an emergency as well as create and distribute voice messages among system users.

The Voice Mail system, through proper administration, can help you reduce the risk of unauthorized persons gaining access to the network. However, phone numbers and authorization codes can be compromised when overheard in a public location, are lost through theft of a wallet or purse containing access information, or through carelessness (writing codes on a piece of paper and improperly discarding them). Additionally, hackers may use a computer to dial an access code and then publish the information to other hackers. Substantial charges can accumulate quickly. It is your responsibility to take appropriate steps to implement the features properly, evaluate and administer the various restriction levels, protect and carefully distribute access codes.

Under applicable tariffs, you will be responsible for payment of toll charges.

AT&T cannot be responsible for such charges and will not make any allowance or give any credit resulting from unauthorized access.

To reduce the risk of unauthorized access through your Voice Mail system, please observe the following procedures:

■ Employees who have voice mailboxes should be required to use the passwords to protect their mailboxes.

— Have them use random sequence passwords.

— Impress upon them the importance of keeping their passwords a secret.

— Encourage them to change their passwords regularly.

■ The administrator should remove any unneeded voice mailboxes from the system immediately.

Customer Support Information xix


■ AUDIX Voice Power™ has the ability to limit transfers to subscribers only.

You are strongly urged to limit transfers in this manner.

■ Use the PBX or Key system administration capability to do the following:

—

Block direct access to outgoing lines and force the use of account codes/authorization codes.

— Disallow trunk-to-trunk transfer unless required.

— Assign toll restriction levels to all AUDIX Voice Power ports.

—

If you do not need to use the Outcalling feature, completely restrict the outward calling capability of the AUDIX Voice Power ports.

■ Monitor SMDR reports or Call Accounting System reports for outgoing calls that might be originated by AUDIX Voice Power ports.

Remote Administration and Maintenance

The Remote Administration and Maintenance feature of your telecommunications system, if you choose to use it, permits users to change the system features and capabilities from a remote location.

The Remote Administration and Maintenance feature, through proper administration, can help you reduce the risk of unauthorized persons gaining access to the network. However, telephone numbers and authorization codes can be compromised when overheard in a public location, are lost through theft of a wallet or purse containing access information, or through carelessness

(writing codes on a piece of paper and improperly discarding them).

Additionally, hackers may use a computer to dial an access code and then publish the information to other hackers. Substantial charges can accumulate quickly. It is your responsibility to take appropriate steps to implement the features properly, evaluate and administer the various restriction levels, and protect and carefully distribute access codes.

Under applicable tariffs, you will be responsible for payment of toll charges.

AT&T cannot be responsible for such charges and will not make any allowance or give any credit resulting from unauthorized access.

To reduce the risk of unauthorized access through Remote Administration and

Maintenance, please observe the following procedures:

■ The System Administration and Maintenance capability of a PBX or Key system is protected by a password.

— Change the default password immediately.

— Continue to change the password regularly.

— Only give the password to people who need it and impress upon

— them the need to keep it secret.

If anyone who knows the password leaves the company, change the password immediately.

xx Customer Support Information


■ If you have a special telephone line connected to your PBX or Key system for Remote Administration and Maintenance, you should do one of the following:

— Unplug the line when it is not being used.

— Install a switch in the line to turn it off when it is not being used.

— Keep the Remote Administration and Maintenance telephone number secret. Only give it to people who need to know it, and impress upon them the need to keep it a secret. Do not write the telephone number on the PBX or Key system, the connecting equipment, or anywhere else in the system room.

■ If your Remote Administration and Maintenance feature requires that someone in your office transfer the caller to the Remote Administration and Maintenance extension, you should impress upon your employees the importance of only transferring authorized individuals to that extension.

Customer Support Information xxi

About This Book

This document covers all aspects of the MERLIN LEGEND™ Communications

System Release 2.0, a state-of-the-art telephone switching system that provides both voice and data communication features.

The document is intended for use by anyone who needs detailed information about the hardware and software that apply to the communications system, including support personnel, technicians, sales representatives, and account executives. It describes system components and capabilities, modes of operation, lines and trunks, applications, and data communications support.

The following documents may be referenced for additional information:

■ Feature Reference

■ System Planning

■ System Programming

See “Related Documents” later in this section.

In the U.S.A. only, AT&T provides a toll-free customer Helpline (1-800-628-

2888) 24 hours a day. Call the Helpline, or your authorized dealer, if you need assistance when installing, programming, or using your system,

About This Book 1

About This Book

Terms and Conventions Used

In this document, the following terms are used to describe components of the communications system:

■ telephone (synonymous with voice terminal)

■ extension (synonymous with station)

■ control unit (synonymous with switch)

Although the terms line and trunk technically refer to different facilities, they are often used interchangeably. In this document, trunk refers to either facility.

Typographical conventions are used in this document to distinguish certain kinds of information. The conventions are as follows:

■ Bold type is used for emphasis and for telephone buttons.

Press Drop to delete the current entry.

■ Constant width type is used for information on telephone display screens or on a PC screen,

Select Sys Program,

■ Bold constant width type indicates information that you enter exactly as shown.

Type install; dial #55.

2 Terms and Conventions Used

About This Book

Product Safety Labels

An exclamation point inside a triangle and the word “caution” or “warning” indicate hazardous situations. These product safety labels appear as follows.

WARNING:

Warning indicates the presence of a hazard that could cause death or severe personal injury if the hazard is not avoided.

CAUTION:

Caution indicates the presence of a hazard that could cause minor

personal injury or property damage if the hazard is not avoided.

Security

The use of passwords prevents unauthorized users from abusing the communications system. It is strongly recommended that passwords be assigned wherever possible and that the passwords are provided only to those persons directly responsible for system administration and maintenance.

Non-displaying access codes and telephone numbers provide another layer of security. The following cautionary note pertains to security:

CAUTION:

For more information about the security of your communications

system to prevent toll fraud, see the “Customer Support Information”

section at the front of this document.

Related Documents

A number of related documents are available, providing additional information about the communications system. Whenever a reference to a related document is given within a document, the reference uses a shortened version of the document’s title. For example, MERLIN LEGEND Communications System

Release 2.0 System Programming is referred to as System Programming.

Within the continental United States, these documents can be ordered from the

AT&T Customer Information Center (CIC) by calling 1-800-432-8600 or by contacting your local sales representative or authorized dealer.

Related Documents 3

About This Book

Document No.

555-620-114

555-620-110

555-620-115

555-620-116

555-620-111

555-620-112

555-62~113

555-620-122

555-620-123

555-620-150

555-620-152

555-620-124

555-620-125

555-620-151

555-620-120

555-620-121

555-620-128

555-620-126

555-620-127

555-620-134

555-620-135

555-620-132

555-620-133

555-620-136

555-620-137

555-620-130

555-620-131

555-620-129

555-620-140

555-620-141

555-620-142

555-620-143

555-620-144

Title

System Documents

System Overview

Feature Reference

Equipment and Operations Reference

Pocket Reference

System Programming

System Planning

System Planning Forms

Telephone User Support

MLX-10D, MLX-28D, and MLX-20L

Display Telephones User’s Guide

MLX-10D, MLX-28D, and MLX-20L

Display Telephones Quick Reference

MLX-10D (Display) Telephone Tray Cards (6 cards)

MLX-28D and MLX-20L Telephone Tray Cards (5 cards)

MLX-10 Non-Display Telephone User's Guide

MLX-10 Non-Display Telephone Quick Reference

MLX-10 (Non-Display) Telephone Tray Cards (6 cards)

Analog Multiline Telephones User’s Guide

Analog Multiline Telephones Quick Reference

ML C-5 Cordless Telephone Quick Reference

Single-Line Telephones User’s Guide

Single-Line Telephones Quick Reference

System Operator Support

MLX Direct-Line Consoles Operator’s Guide

MLX Direct-Line Consoles Quick Reference

Analog Direct-Line Consoles Operator’s Guide

Analog Direct-Line Consoles Quick Reference

MLX Queued Call Console Operator's Guide

MLX Queued Call Console Quick Reference

Miscellaneous User Support

Calling Group Supervisor’s Guide

Calling Group Supervisor’s Quick Reference

Data User’s Guide

Documentation for Qualified Technicians

Installation, Programming, & Maintenance (lP&M) Binder

(consists of 555-620-141,555-620-142, 555-620-143, and 555-620-1 44)

Installation

System Programming & Maintenance (SPM)

Maintenance and Troubleshooting

Programming Summary

4 Related Documents

About This Book

How to Comment on This Document

We welcome your comments about the usefulness of this document. Please tell us what you like, as well as what you would improve. You may use the feedback form on the next page to let us know how we can continue to serve you. If the feedback form is missing, write directly to:

A. Sherwood

AT&T

99 Jefferson Road

Room 2A25

Parsippany, NJ 07054.

How to Comment on This Document 5

Introduction

The MERLIN LEGEND Communications System is an advanced digital switching system that integrates voice and data communications features. Voice features combine traditional telephone features, such as Transfer and Hold, with advanced features, such as Group Coverage and Park. Data features enable the transmission of voice and data over the same system wiring. This chapter describes the following aspects of the system:

■ Components—the required and optional equipment that makes up the

system

■ Functional Description-the functional units that make up the control unit,

their relationships, and the process of signaling

■ Modes of Operation—the three modes for which the system can be

configured: Key mode, Hybrid/PBX mode, and Behind Switch mode

■ Programming-general information about programming the system and

telephones

■ System Capacities and Requirements—the technical requirements and

capacities of the system, for example, hardware and software capacities and environmental, power, and grounding requirements

Release Differences-enhancements provided by Release 1.1 and

Release 2.0 of the system

Introduction

1-1

Introduction

Components

The system consists of the following basic components and optional auxiliary components:

■ Basic components

— Control unit

— Telephones

■ Auxiliary components

— Adjuncts

— Adapters

— Applications

Control Unit

The control unit consists of the basic carrier and up to two expansion carriers.

The basic carrier contains the processor module, power supply module, and line/trunk and station modules. Each expansion carrier contains a power supply module and line/trunk and station modules.

Telephones

The telephones that can be used with the system are MLX (digital) telephones, analog multiline telephones (including cordless telephones), and single-line telephones.

MLX

Telephones

The following MLX telephones can be used:

■ MLX-10D™ (10 buttons with display)

■ MLX-20L™ (20 buttons with display)

■ MLX-28D™ (28 buttons with display)

■ MLX-10™ (10 buttons, no display)

Analog Multiline Telephones

The following analog multiline telephones can be used:

■ 5-button* (5 buttons, membrane, no adjuncts supported)

1-2 Components

Introduction

■ 10-button* (10 buttons, membrane)

■ 34-button* (34 buttons, membrane)

■ 34-button Deluxe* (34 buttons, membrane)

■ 10-button HFAI* (10 buttons, hands-free-answer, no adjuncts supported)

■ 34-button BIS* (34 buttons, built-in speakerphone)

■ 34-button BIS/DIS* (34 buttons, built-in speakerphone, 16-character display)

■ BIS-10 (10 buttons, built-in speakerphone)


■ BIS-22D (22 buttons, built-in speakerphone, 16-character display)


■ BIS-34D (34 buttons, built-in speakerphone, 16-character display)

■ MLC-5 Cordless (5 buttons, cordless)

■ MERLIN PFC™ Telephone (telephone, fax machine, and copier)

*

Vintage telephone; no longer available for sale or lease

† Requires two analog multiline ports

Single-Line Telephones

■ 2500MMGB (desk telephone)

■ 2554MMGJ (wall telephone)

■ 2500YMGK* (desk telephone, message light, Recall button)

■ 2500SM (desk telephone used with 4A speakerphone)

■ 2514 BMW (desk telephone with built-in headset jack)

■ 2526BMG (outdoor telephone used with waterproof enclosure)

■ 7101A* (desk telephone, message light, Recall and Disconnect buttons, no adjuncts supported)

■ 7102A (desk telephone, message light, Recall button, supports 101 and

201 speakerphones and 500 headsets)

■ CS6402U0IA* (desk telephone, built-in speakerphone, memory, redial)

■ 2500MMGJ (desk telephone)

■ 2500MMGK (desk telephone, timed Recall button action activates Hold and Transfer)

■ 8102 (desk telephone, data jack for connecting a modem, slot for headset adapter, and jack for speakerphone adjuncts)

Components 1-3

Introduction

■ 8110 (desk telephone, built-in speakerphone with volume control, auxiliary power jack for improving quality of built-in speakerphone, Mute button with LED indicator, and data jack for connecting a modem)

■ 500MM, 554BMPA, 500SM (rotary dial)

* Vintage telephone; no longer available for sale or lease

Adjuncts

Adjuncts are pieces of equipment that connect directly to the control unit or to a telephone through an adapter (see Adapters). Answering machines, credit card verification terminals, and alerts are examples of adjuncts.

Adapters

Adapters enable the connection of equipment or, in the case of a channel service unit (CSU), of Digital Signal 1 (DS1 ) facilities to the control unit. Some adapters connect directly to the control unit (system adapters) while others connect to telephones (telephone adapters).

■ System adapters

— ESF T1 CSU

— 551 T1 L1 CSU

— Universal Paging Access Module (UPAM)

— Loop trunk adapter for paging

■ Telephone adapters

— Multi-Function Module (MFM) for MLX telephones

WARNING:

The MFM can be installed or repaired only by a qualified technician or an authorized dealer representative. To eliminate the risk of electrical shock, the MLX telephone should not be disassembled.

— General Purpose Adapter (GPA) for analog multiline telephones

7500B Data Module for connecting digital data equipment either directly to the control unit or to an MLX telephone (for simultaneous voice and data transmission)

— 7500B Data Module for connecting digital data equipment modem for connecting digital equipment (such as a personal computer) to a tip/ring (T/R) interface

1-4 Components

Introduction

— Supplemental Alert Adapter (SAA) for connecting an alert (such as a horn or strobe) to an analog multiline telephone

Applications

The following applications for the system consist of software and/or hardware

that add functions to the system. See the Applications chapter for details.

■ MERLIN MAIL™ Voice Messaging System

■ MERLIN® Attendant

■ Call Accounting System (CAS)

■ Call Accounting Terminal (CAT)

■ Call Management System (CMS)

■ InnManager™ Guest Management System

■ System Programming and Maintenance (SPM)

■ Integrated Solution II (IS II)

— Integrated Voice Power Automated Attendant (IVP AA—IS II)

— AUDIX Voice Power™—lS II (AVP—IS II)

— Call Accounting System—lS II (CAS-IS II)

— System Programming and Maintenance-lS II (SPM—IS II)

■ Integrated Solution Ill (IS Ill)

— Integrated Voice Power Automated Attendant (IVP AA—IS Ill)

— AUDIX Voice Power—lS Ill (AVP—IS Ill)

– Call Accounting System–lS Ill (CAS-IS Ill)

— System Programming and Maintenance—lS Ill (SPM–IS Ill)

— FAX Attendant System™

■ Primary Rate Interface (PRI) Applications

— Group IV (G4) Fax

— Video Conferencing

■ Centrex service

■ MERLIN PFC

■ Automated Document Delivery System (ADDS)

■ CONVERSANT® Intro

Components 1-5

Introduction

Figure 1-1 shows the components of the system. The numbered paragraphs following the figure correspond to the numbered items in the figure.

Figure 1-1. System Components

1-6 Components

Introduction

1.

2.

Control Unit. The backbone of the system, consisting of the basic and expansion carriers, power supply module, processor module, and line/trunk and station modules. The control unit connects telephone company lines/trunks with stations such as telephones and adjuncts.

Line/Trunk and Station Modules. The components that connect telephone company lines/trunks and terminal equipment such as telephones, external alerts, and fax machines via jacks to the control unit.

3.

4.

5.

6.

7.

8.

Carrier (Basic). The component attached to the backboard used to hold the modules needed for system operation. The basic carrier houses the processor module, power supply module, and up to five line/trunk and station modules. Each expansion carrier houses its own power supply and up to six additional line/trunk and/or station modules. One or two expansion carriers can be added.

Processor Module. A miniature computer that controls most of the system’s features, and supplies the system’s diagnostics. The processor provides two jacks, one for Station Message Detail Recording (SMDR) and the other for system programming and maintenance via a personal computer (PC).

Power Supply Module. The component that supplies DC power for the modules and telephones (one power supply unit is needed per carrier).

If the system’s power requirements exceed the capacity of the power supply, an auxiliary power supply unit can be added.

Direct Station Selector (DSS). A console that adds 50 buttons for onetouch extension dialing to the MLX-20L or MLX-28D telephone and speeds call handling.

Analog Data Station. A data terminal such as a PC, printer, or optical reader that connects, via a modem (for transmitting and receiving analog signals), to a 012 basic telephone module or a 008 off-premises telephone (OPT) module. A data terminal can also be connected to an

MLX telephone using an MFM or to an analog multiline telephone using a

GPA.

Magic On Hold®. Optional equipment that connects to the system through a ground-start/loop-start (GS/LS) jack programmed for Musicon-Hold. (A customer-provided music source can be connected instead of Magic On Hold.)

9.

General Purpose Adapter (GPA). An adapter used to connect a variety of tip/ring (T/R) adjuncts to an analog multiline telephone (shown here with an answering machine).

10.

Analog Multiline Telephone. A 34-button telephone with built-in speakerphone that connects to the system via an analog station jack.

Other analog multiline telephones compatible with the system include the

22- and 34-button with built-in speakerphone and a l-line, 16-character display, and the 10- and 22-button with built-in speakerphone, without display. In addition, the following vintage membrane telephones (no

Components 1-7

Introduction longer available for sale or lease) are compatible with the system:

5-button, 10-button, 34-button, and 34-button Deluxe.

11. MLC-5 Cordless Telephone. A cordless multiline telephone that connects to the control unit via an analog station jack.

12. Industry-Standard Single-Line Telephone. A touch-tone or rotary industry-standard telephone connected to the system via a 012 basic telephone module, a 008 OPT module, or an MLX telephone via an MFM.

13. Off-Premises Telephone (OPT). A single-line, touch-tone or rotary, industry-standard telephone located in a different building from the control unit.

14. External Alert. Alerting devices such as balls, chimes, and strobe lights that connect to a jack on a 012 basic telephone module or a 008 OPT module, or to an MFM or SAA.

15. Digital Data Station. A data terminal such as a PC, printer, or optical reader that connects via a 7500B Data Module to a 008 MLX or 408

GS/LS-MLX module (Release 2.0 only) and that can also include an MLX telephone.

16. MLX-20L Telephone. An MLX telephone with 20 line buttons and a display with seven lines of 24 characters each. The MLX-20L telephone can be used as a system programming console. Other MLX telephones are as follows:

— MLX-10 Telephone: a 10-button MLX telephone without a two-

Iine, 24-character display

— MLX-10D Telephone: a 10-button MLX telephone with a two-line

24-character display

17. MLX-28D Telephone with Multi-Function Module. An MLX telephone with 28 line buttons and a two-line, 24-character display.

A Multi-Function Module (MFM) is a circuit board mounted inside an MLX telephone that provides a jack to connect optional equipment such as answering machines, fax machines, external alerts, and modems to the telephone.

18. Fax. Industry-standard fax machines connected to the control unit via a jack on a 012 basic telephone module or a 008 OPT module, an MFM, or a GPA.

19. SMDR Printer. A printer for SMDR call records, connected via a RS-232 jack on the processor.

20. Applications. Software and hardware for the system that can be connected to the control unit to provide more functions.

21. Loudspeaker Paging. A single-zone or multizone system such as

PagePac® with Zonemate™ 9 or 39 that connects via an administered jack on a GS/LS module.

1-8 Components

Introduction

22. AC Power Outlet. A dedicated 115-VAC wall outlet (not controlled by an on/off switch) that supplies power to the control unit.

Functional Description

This section describes the functional units that make up the control unit, their relationships, and how signals are processed.

The control unit contains the following functional units:

■ Processor module

■ Power supply module

■ Carrier

■ Line/trunk and station modules

Functional Units

The functional units are the processor module, which controls the operation of the system and its features; the power supply module, which supplies power to the control unit; the carrier with its backplane assembly, which contains the input/output (I/0) bus and the time-division multiplex (TDM) bus; and the line/trunk and station modules, which connect the outside lines/trunks to the stations. All the modules are electrically connected to the backplane, which provides common circuitry for the I/O bus, the TDM bus, and power distribution,

The processor is connected to intelligent ports on the line/trunk and station modules, through the I/0 bus, by the digital switch element (DSE) on each line/trunk and station module. The TDM bus is also connected to the DSE of

each line/trunk and station module. The two buses are illustrated in Figure 1-2.

Functional Description 1-9

Introduction

Figure 1-2.

Functional Units

1-10 Functional Description

Introduction

■ Input/Output Bus. The I/O bus contains a 16-bit address bus and an 8bit data bus. The address bus selects the module that receives instructions from the 68000 microprocessor in the processor module.

The microprocessor provides instructions to the port processors and

DSEs through the 8-bit data bus.

■ Time-Division Multiplex Bus. The TDM bus connects the DSEs to allow voice or data to flow in and out of the system. The TDM bus is parallel, 8 bits wide, and runs at 2.048 MHz (256 time slots x 8 kHz = 2.048 MHz).

Each TDM cycle has 256 time slots for voice, data, tones, and clocks.

The frame repetition rate is 8 kHz, providing a 64-kbps channel on each time slot (8-bit bus x 8 kHz = 64 kbps).

The built-in modem connects to the TDM bus; this permits access from a local or remote PC or workstation equipped with a 1200-bps modem,

The TDM bus connects with the built-in diagnostics that enable the processor to read and write to dedicated TDM test slots.

The TDM bus carries analog signals encoded in Mu-Law 255 pulse code modulation (PCM) format for domestic use. The system provides a circuit-switched connection for transmission of digital data signals up to

64 kbps.

Digital Switching

Because the system is internally a digital system in a world of both analog and digital devices, it must accurately translate analog signals, Doing this involves signal conversion and switching. Codecs provide analog-to-digital and digitalto-analog conversion. The digitally encoded signals are routed from one interface port to another interface port by assigning source and destination information to specific time slots on the TDM bus. In this way, signals can be transmitted to one or several destinations and reconstructed at the original amplitude. The result is no signal loss during switching and transmission from one point to another.

The TDM bus allows many users to communicate over a common electrical connection because it is physically distributed across the backplane of the control unit and connects all line/trunk and station modules.

The processor uses the DSE to specify time slots for various functions, For example, during a conversation between station A and station B, a time slot is reserved for station A to transmit on and for station B to receive on. For example, station A talks on time slot 150 and listens on time slot 160. Station B talks on time slot 160 and listens on time slot 150.

A digital tone plant in the processor module provides touch-tone and callprogress signals to stations via time slots 0 to 39. Unlike other bus configurations, the DSEs on the TDM bus receive all transmissions. If a DSE is not assigned to any of the time slots, the DSE ignores the data.

Functional Description 1-11

Introduction

Each module has a DSE to interface codecs or digital transceivers to the TDM bus. The actual digital switching occurs when the DSE is programmed by the system I/O bus to transmit data on or receive data from the TDM bus in specific time slots. In addition, the DSE can sum digital signals from designated TDM slots to provide conferencing for up to five parties.

Modes of Operation

NOTE:

Although the terms “line” and “trunk” technically refer to different facilities, the differentiation is not as clear as it once was, and the terms are usually treated as if they are interchangeable. A “line” traditionally connects a piece of equipment to a switching system; for example, it connects your home telephone to the local telephone company. A “trunk” connects one switching system to another switching system; for example, it connects a communications system like this (except for facilities line personal lines that pass transparently through the system) to the local telephone company’s central office (CO). Since the industry trend seems to be toward using

“trunk” to refer to either facility, this standard is used in this section.

The system can be programmed to operate in Key, Hybrid/PBX, or Behind

Switch mode. The mode of operation determines the following:

■ The types of outside trunks that can be connected to the system

■ How telephone users access outside trunks

■ The types of system operator consoles allowed

■ The features available and how they work

The following sections describe each mode of operation, and Table 1-1 summarizes the modes.

Table 1-1. Modes of Operation Summary

Key Hybrid/PBX Behind Switch

Trunks connected directly to control unit:

■ Ground-start

KF registration (FCC)

MF registration (FCC)

PF registration (FCC)

■ Loop-start

■ PRI

■ DSI

No

Yes

No

Yes

Yes

Yes

No

Yes

Yes

Yes

Yes

Yes

No

Yes

No

Yes

Yes

Yes

Continued on next page

1-12 Modes of Operation

Introduction

Table 1-1. - Continued

Key Hybrid/PBX Behind Switch

■ Tie

■ FX

■ WATS

■ DID

Trunk pools

ARS

ICOM buttons

SA buttons

Line buttons, that is, outside trunks assigned to buttons on telephone

Shared trunks

Yes

Yes

Yes

No

No

No

Yes

No

Yes

Yes

(outside trunks only)

Prime Lines

Queued Call Console (QCC)

Number of extensions:

■ <5(3

■ >50

FCC registration

No

No

Good

Not recommended

KF or MF

Yes

No

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Good

Good

MF or PF

Yes

Yes

Yes

No

No

No

Yes

No

Yes

Yes

(outside trunks and SA buttons)

Yes

(outside trunks only)

No

Yes

Yes

No

Good

Good up to 80

KF or MF

Key

Mode

A Key system is the simplest way to provide users with more than one line from a single telephone. Older Key systems have telephones that look like single-line telephones except for a row of buttons, illuminated by incandescent lights when active, across the bottom. The leftmost button is labeled Hold, and the other buttons are labeled with telephone numbers.

When the communications system operates in Key mode, telephones are programmed with two kinds of buttons:

■ Line buttons (or keys) are associated with specific outside (telephone company) trunks. Line buttons allow users to see activity on other telephones, join conversations, and make and receive calls.

■ Intercom buttons are used to make and receive internal calls.

Modes of Operation 1-13

Introduction

The Key mode of operation accommodates the following kinds of outside trunks:

■ Loop-start trunks, including basic lines, WATS, and foreign exchange

(FX)

■ Ground-start trunks (only if registered as MF and if not strapped for Key mode, as described below) and emulated ground-start trunks on T1 facilities

■ DS1 facilities

■ Tie trunks and emulated tie trunks on T1 facilities

A standard Key system’s trunks are all loop-start. A loop-start trunk introduces a slight delay between the time the telephone company’s CO recognizes a call attempt and the time the call is processed. This delay is minimal and virtually unnoticeable. Most residence and small business telephones have loop-start trunks.

The communications system is configured for Key mode operation by system programming, depending on how the system is registered with the Federal

Communications Commission (FCC), as described later in this section. If the system is modified for Key-only operation by the hardware strap in the processor module, no ground-start trunks can be connected to it. However, ground-start emulation on a T1 facility is allowed.

NOTE:

■ The default programmed mode is Key.

■ On initialization of a Release 1.0 system, all loop-start and groundstart trunk programming reverts to loop-start. In Releases 1.1

and 2.0, if the system is programmed for Key mode, the strap is checked on initialization. If the strap is in (Key-only operation), all trunks revert to loop-start. If the strap is not in, any programmed designation of ground-start trunks is retained.

The following features are not available in Key mode:

■ Direct inward dialing (DID) trunks

■ Trunk pools

■ Automatic Route Selection (ARS)

■ QCCs and associated features

■ System Access buttons


Introduction

Line Access

In Key mode (whether strapped and/or programmed), each outside line must be assigned to a line button on at least one telephone. As a result, the telephones most commonly used in Key mode are multibutton telephones. See Figure 1-3.

Telephone # 1 Telephone # 2 Telephone #

3

Figure 1-3. Lines Labeled for Key System Telephones

A user selects an outside line by pressing a personal line button-a button labeled with a telephone number. Upon hearing dial tone, the user can dial out.

When a line is in use, the green LED goes on next to the corresponding personal line button on any telephones that share that personal line.

The telephones in a Key system also have Intercom buttons, labeled ICOM, that allow users to make and receive calls to and from other extensions within the system. An inside talk path is provided when an ICOM button is pressed. The user hears a system dial tone, which can be programmed to sound different from outside dial tone. The factory setting is to provide a different internal dial tone.


Introduction

The following types of ICOM buttons can be used to make and receive inside calls in Key mode:

■ An ICOM Ring button is used to make inside calls and to receive inside and outside calls transferred from another extension. When an

ICOM Ring button is used to make an inside call, the telephone at the destination extension rings once per ring cycle to indicate an inside call.

■ An ICOM Voice button is used to make inside calls and to receive inside and outside calls transferred from another extension. When an

ICOM Voice button is used to make an inside call, the person at the destination extension hears the caller’s voice on the speakerphone after a beep, rather than ringing. (If he or she has a single-line telephone, does not have a speakerphone, or has disabled voice announcements, the telephone rings the same as if the call had been made on an

ICOM Ring button.)

■ An ICOM Originate Only button is used only to make inside calls.

Neither inside nor outside calls are received on an ICOM Originate Only button. This type of button ensures that the user always has a button available to make or transfer a call, establish a conference call, answer a

Call Waiting call, or pickup parked calls. The button can be programmed for either voice or ring operation.

A combination of up to 10 ICOM Voice, ICOM Ring, and ICOM Originate Only buttons can be assigned to each telephone on buttons 1 through 10. The number of personal line buttons that can be assigned to a telephone is limited only by the number of trunks in the system and the number of buttons available on the telephone. See System Planning for button diagrams.

Key

System Configurations

In Key mode, the system can be configured as a square, modified, or hybrid

Key system, as described in the following sections. (The communications system does not distinguish among these configurations. )

In Key mode, the first eight trunks connected to the system are automatically assigned to the same eight buttons on all multiline telephones; all trunks are automatically assigned to each Direct-Line Console (DLC).

Square Key System

In a square Key system, every outside trunk in the system terminates on a personal line button on every telephone in the system.

The system illustrated in Figure 1-3 is a square Key system in an office with

three outside trunks and three telephones. Each of the three trunks is assigned to the same button on each telephone. When a trunk is assigned to a personal line button on more than one telephone, it is considered a shared line.


Introduction

Modified Key System

A Key system can be modified through system programming to provide trunk access for special business needs. For example, some business do not require every user to have access to a tie trunk, so the system can be programmed so that some telephones do not have access to all outside trunks.

Figure 1-4 shows an example of personal line button assignments in a modified

Key system with two outside trunks and one tie trunk. Each trunk is not assigned to a button on every telephone.

Figure 1-4. Lines Labeled for Modified Key System Telephones

Hybrid Key System

A hybrid Key system allows ground-start trunks to be connected directly to the system’s control unit. To program ground-start trunks in Key mode, the system may not be strapped for Key-only operation and must be registered with the

FCC under an MF classification, as described later in this chapter, in the section

FCC Registration. In this configuration, all outside trunks, including ground-start

trunks, are assigned to personal line buttons on each telephone.

Key Mode Considerations

■ The multibutton telephones most commonly used in a Key system provide easy access to outside trunks. To get a dial tone, the user simply lifts the handset and an outside line is automatically selected.

■ The intercom path helps ensure that outside trunks are available when needed.


Introduction

Key mode has the flexibility to provide trunk access according to user needs. For example, tie trunks can be terminated on the telephones of only those users who need them.

The loop-start trunks traditionally associated with Key mode operation can cost less than the trunks used in the other modes.

Key systems are best suited to smaller businesses.

To take advantage of the features and functionality of the system, all users should have multibutton telephones when the system is operating in Key mode.

If the number of trunks connected to the system is larger than the number of buttons available on the DLC, Hybrid/PBX mode, which offers the QCC, may be more functional.

To make more efficient use of outside trunks by grouping them into pools for shared use, or to use ARS, the system must be programmed to operate for Hybrid/PBX operation.

Hybrid/PBX Mode

A private branch exchange (PBX) originally was a large switchboard installed at a customer’s office that functioned like a small, self-contained telephone company. The switchboard was manually operated, and the system operators physically connected calls by plugging cords into the board’s jacks. Today’s

PBX is a processor in the communication system control unit programmed to connect both inside and outside calls on a single button. In Hybrid/PBX mode, this button is called a System Access button, and is labeled SA.

Although there is no longer a person handling cords, the communications system operating as a PBX still requires the user to request an outside trunk. A user simply dials a dial-out code (usually a 9) and the telephone number on an

SA button, and the system routes the call to an available outside trunk.

Thus, the major distinction of Hybrid/PBX mode, is that both inside and outside calls can be made on the same button.

The Hybrid/PBX mode of operation accommodates the following kinds of outside trunks:

■ Loop-start trunks, including basic lines, WATS, and FX

■ Ground-start trunks, including basic lines, WATS, and FX

■ DS1 facilities

■ Tie trunks and emulated tie trunks on T1 facilities

■ DID trunks

Programming the system for Hybrid/PBX mode automatically arranges the

outside trunks in functional groups, or pools, within the control unit. (See Figure

1-5. ) The system can have up to 11 separate trunk pools. The number of pools

programmed depends on both the kinds of trunks and the special needs of the users.

1-18 Modes of Operation

Introduction

Since the outside trunks are pooled, outside numbers are not associated with individual telephones. When a pool is assigned to a line button during system programming, it is called a pool button. Users request specific trunk pools by dialing the trunk pool number (870-879) for the pool or by pressing a pool button, which gives one-touch access to a group of trunks.

Telephone # 1 Telephone # 2 Telephone # 3 Telephone # 4

Figure 1-5. Hybrid/PBX Mode

Outside calls are normally answered by a system operator and transferred to individual system users.


Introduction

A feature commonly used in Hybrid/PBX mode is ARS. When an SA button is used to make an outside call and the ARS dial-out code is entered, the system selects the next available trunk from the type of pool that is most cost-effective for the call and gives the user access to that trunk.

The system automatically provides three types of trunk pools and assigns trunks to the appropriate pool type:

■ Loop-start trunks, by default, are assigned to pool number 70. This pool is called the loop-start pool or main pool.

■ Ground-start trunks, by default, are assigned to pool number 890. This pool is called the ground-start pool.

NOTE:

On initialization of a Release 1.0 system, all Imp-start and groundstart trunk programming reverts to Imp-start. The ground-start pool never has trunks assigned to it automatically, but must be programmed after the ground-start ports are designated. In

Releases 1.1 and 2.0, ground-start trunks are assigned to the ground-start pool on initialization, except in a system strapped for

Key mode operation.

■ Tie trunks, by default, are assigned to pool number 891. This pool is called the tie pool.

Through system programming, the three automatically assigned pools can be rearranged and special-function or special-user pools can be created. For example, the main pool can be divided and smaller pools of loop-start trunks can be assigned to different groups of users.

Line Access

To make an outside call, the single-line telephone user dials a pool access or

ARS dial-out code, and the system automatically selects an outside trunk. In addition, SA buttons on multiline telephones allow different kinds of calls to be made from the same buttom--outside calls on basic loop-start or ground-start trunks or on tie trunks or special service facilities such as WATS, and inside calls to other extensions in the system.

The following types of buttons can be assigned to multiline telephone users:

■ An SA Ring button is used to make and receive inside and outside calls.

When an SA Ring button is used to make an inside call, the telephone at the destination extension rings once per cycle to indicate an inside call.

■ An SA Voice button is used to make and receive inside and outside calls. When an SA Voice button is used to make an inside call, the person at the destination extension hears the caller’s voice on the speakerphone after a single beep, rather than ringing. (If the person at the destination extension has a single-line telephone, does not have a speakerphone, or has disabled voice announcements, the telephone rings the same as if the call had been made on an SA Ring button.)


Introduction

■ An SA Originate Only button is used only to make inside and outside calls. Neither inside nor outside calls are received on an SA Originate

Only button. The purpose of this type of button is to ensure that the user always has a button available to make or transfer a call, establish a conference call, answer a Call Waiting call, or pick Up parked calls. For inside calls, the button can be programmed for either voice or ring operation.

■ A Shared SA button is used to allow two or more users to answer each other’s calls, join conversations, or make or receive inside or outside calls on each other’s SA Ring or SA Voice buttons. In a Shared System

Access arrangement, one extension is designated as the principal (or primary) extension. This extension is the telephone from which SA Ring,

SA Voice, and/or SA Originate Only buttons are assigned as

Shared SA buttons on one or more telephones in the Shared System

Access arrangement.

Shared SA buttons are often provided to secretaries and their bosses, as well as to others who work closely together, such as a customer service department. For inside calls, the button can be programmed for either voice or ring operation.

■ A pool button is used to make outside calls using a specific trunk pool,

To make an outside call, the user presses the appropriate pool button— no dial-out code is necessary.

■ A personal line button is used to dedicate an outside trunk for use by one or more telephones in the system. The personal line button is used to make and receive only outside calls. To make a call, the user presses the appropriate personal line button—no dial-out code is necessary.

A combination of up to 10 SA Voice, SA Ring, SA Originate Only, and

Shared SA buttons can be assigned to each telephone (except for the QCC) on buttons 1 through 10. See System Planning for button diagrams, The number of personal line buttons that can be assigned to a telephone is limited only by the number of trunks in the system and the number of buttons available on the telephone.

Queued Call Console

The type of system operator position typically used in Hybrid/PBX mode is the

QCC, which allows calls to come to the operator one at a time. This is especially useful when the number of outside trunks connected to the system control unit exceeds the number of buttons available on a DLC.

A call is held in queue until the system operator is available, After the call is delivered to a Call button on the QCC, the operator transfers it to the desired internal destination. The operator can switch between the caller and the called person to screen calls.


Introduction

Hybrid/PBX Mode Considerations

■ Hybrid/PBX mode provides the most efficient use of outside trunks since they can be pooled and are more readily available to users. The ARS feature can be programmed for more cost-effective use of pools.

■ Hybrid/PBX mode provides greater functionality for single-line telephones than other modes of operation. The telephone user can make both inside and outside calls by accessing a pool of trunks.

■ The QCC, available only in Hybrid/PBX mode, ensures efficient call handling and is especially useful when the number of lines exceeds the number of buttons available on a DLC system operator position.

Behind Switch Mode

The system operates in the Behind Switch mode when the control unit is connected to (is “behind”) another system. The other system is referred to as the host, and can be either a PBX or Centrex service (a telephone company service that provides PBX-like capabilities but is housed at the CO).

Figure 1-6 illustrates a very simple Behind Switch configuration in which the

outside trunks are connected to the host system. The lines connecting the two control units are like extensions that provide users with access to the host system’s trunks, as if via a trunk pool.

Each extension number from the host system is assigned to individual telephones as a Prime Line. This line rings when the user receives an outside call and the user is connected to this line (even when the line is in use) unless the user manually selects a different line.

Each Prime Line can be assigned to additional telephones as secondary lines so that users can see activity on other telephones sharing the button and join their co-worker’s conversations.

All Prime Lines are also automatically assigned to DLC system operator positions. The first line assigned is the system operator’s Prime Line and the rest are assigned as secondary lines. This allows the system operator to answer calls received by other users on their Prime Lines.


Introduction

Telephone # 1 Telephone # 2

Figure 1-6. Behind Switch Mode

Telephone # 3 Telephone # 4

Modified Configurations

In addition to accessing the host’s outside trunks, the Behind Switch system can be modified to bypass the host and provide direct access to outside trunks—for example, to connect WATS lines directly to the control unit so that they are

available only for this system’s users. Figure 1-7 illustrates this configuration,


Introduction

Telephone # 1 Telephone # 2 Telephone # 3 Telephone # 4

Figure 1-7. Behind Switch Mode with Direct Outside Trunks

Depending on business needs, the following kinds of direct outside trunks can be added:

■ Loop-start trunks, including basic lines, WATS, and FX

■ DS1 facilities


Introduction

■ Tie trunks

■ Ground-start trunks (only if not registered as KF and not strapped for

Key mode)

The direct outside trunks must be terminated on individual telephones and must appear on the telephones’ line buttons. For example, if tie trunks are assigned to buttons on telephones 1 and 2, the buttons on those telephones appear as shown in Figure 1-8.

Telephone # 2

Figure 1-8. Labeled Line Buttons for Behind Switch Telephones

The bottom button on each telephone is reserved for the Prime Line from the host system. The outside lines appear on the labeled line buttons above the

Prime Line. In Figure 1-8, the first line (5678) is assigned to buttons on both telephones, and the second (5679) is assigned to a button only on the second telephone.

The following features are not available in Behind Switch mode:

■ Trunk pools

■ A R S

■ Q C C

■ SA buttons


Introduction

Line Access

In Behind Switch mode, the telephones used most commonly are multibutton telephones. Single-line telephones can also be used and can be set up in two ways. In one configuration, the user has constant access to the Prime Line buttons, but cannot make intercom calls or use system features. In the second configuration, the user can make and receive intercom and Prime Line calls and can use system features, but must dial a dial-out code to make an outside call.

When the system is programmed for Behind Switch mode, the system assigns a single Prime Line, an ICOM Ring button, and an ICOM Voice button to each multiline telephone. Two ICOM Ring buttons are assigned to each single-line telephone; no Prime Lines are assigned. When the telephone handset is lifted, the Prime Line is selected automatically (even when it is busy) unless the user has first selected a different button. The Prime Line connects only with the host system, not directly with an outside trunk.

To call another person connected to the host system, the user dials the host system extension number assigned to that person. To access an outside trunk, the user dials the host system’s dial-out code (usually a 9), and the host system selects an available outside trunk.

In a Behind Switch system, ICOM buttons allow users to call other people connected to the system. When a user presses an ICOM button, the system provides an inside talk path and dial tone from the Behind Switch system (not from the host system). The user can then reach co-workers without tying up a

Prime Line.

The following types of lCOM buttons can be used to make and receive inside calls in Behind Switch mode:

■ An ICOM Ring button is used to make inside calls and to receive inside calls and outside calls transferred from another extension. When an

ICOM Ring button is used to make an inside call, the telephone at the destination extension rings with a one-burst ring to indicate an inside call.

■ An ICOM Voice button is used to make inside calls and to receive inside calls and outside calls transferred from another extension. When an

ICOM Voice button is used to make an inside call, the person at the destination extension hears the caller’s voice on the speakerphone after a single beep, rather than ringing. (If he or she has a single-line telephone, does not have a speakerphone, or has disabled voice announcements, the telephone rings the same as if the call had been made on an ICOM Ring button.)

■ An ICOM Originate Only button is used only to make inside calls.

Neither inside nor outside calls are received on an ICOM Originate Only button. The purpose of this type of button is to ensure that the user always has a button available to make or transfer a call, establish a conference call, answer a Call Waiting call, or pick up parked calls. The button can be programmed for either voice or ring operation.

1-26 Modes of Operation

Introduction

A combination of up to 10 ICOM Voice, ICOM Ring , and ICOM Originate

Only buttons can be assigned to each telephone, except for single-line telephones, on buttons 1 through 10. See System Planning for button diagrams. The number of Prime Line buttons that can be assigned to a telephone is limited only by the number of trunks in the system and the number of buttons available on the telephone.

In Behind Switch mode, users have access to the special features of both the

Behind Switch system and the host system. When both systems have common features, the customer must decide which system will be used for those features through system programming, A fixed Conference, Drop, or Transfer button automatically activates that feature on the host system. A user can program an unlabeled button for any of those features to access that feature on the Behind

Switch system. Each system must be programmed accordingly, and the users must be given the appropriate access instructions.

When users press a fixed Conference, Drop, or Transfer button, the respective host features are activated: a timed switchhook flash is sent to the host, followed by a programmed feature access code. The person programming the system must obtain the feature access code for the host system and program it to the appropriate button.

NOTE:

A single-line telephone user has access only to the host system’s

Conference, Drop, and Transfer features.

Behind Switch Considerations

Behind Switch mode is appropriate for users who are part of a large organization. For example, a department might not want (or be able) to support a large-capacity PBX. Programming the communications system for Behind

Switch operation provides the advantage of the host’s features and capabilities.

A business with multiple locations can use Centrex services to provide the appearance of a single system at all locations.

FCC Registration

The account representative or authorized dealer who planned the system’s mode of operation provides the FCC registration number that the customer reports to the local telephone company. Depending on mode of operation and the hardware strap in the processor module, this number includes the letters

KF, MF, or PF, loosely corresponding to “key function,” “multi-function,” or “PBX function, ” respectively. (The FCC has no Behind Switch classification. )


Introduction

NOTE:

The communications system’s modes of operation (Key, Behind Switch, and

Hybrid/PBX) do not correspond directly to these designations.

Table 1-2 lists the registration number(s) used for each mode of operation.

Table 1-2. FCC Registration Numbers

Mode of Operation Registration Number

Key or Behind Switch AS593M-72914-KF-E

Key, Hybrid/PBX, or Behind Switch AS593M-72682-MF-E

Hybrid/PBX AS5USA-65646-PF-E

The following guidelines are used to determine which classification is used.

KF Classification

The system’s KF classification number is AS593M-72914-KF-E. This classification is applicable only to the Key and Behind Switch modes of operation. The system is registered under the KF classification if any of the following conditions are met.

Key Mode of Operation

■ The system is strapped for Key-only operation.

■ All outside trunks terminate on one or more telephones.

■ All outside trunks are loop-start, tie, DS1 facilities, or ground-start emulation on DID (or a combination of these).

■ No trunks are pooled.

Behind Switch Mode of Operation

■ The system is strapped for Key-only operation.

■ No outside trunks are connected directly to the control unit. The communications system accesses only trunks connected to the host switch.

■ No ground-start trunks are connected directly to the control unit, except for ground-start emulation on DID.

■ No trunks are pooled.


Introduction

MF Classification

The system’s MF classification number is AS593M-72682-MF-E. This classification is applicable to all three modes of operation—Key, Hybrid/PBX, and Behind Switch. The system is registered under the MF classification if any of the following conditions are met.

Key Mode of Operation

■ One or more ground-start trunks are connected directly to the control unit. These trunks connect to a 400 GS/LS/TTR, 408 GS/LS, 800 GS/LS, or 408 GS/LS-MLX module (Release 2.0 only).

Hybrid/PBX Mode of

Operation

■ All outside trunks are pooled; no trunks are terminated directly on a telephone.

■ The only directly terminated trunks are personal lines, not shared lines— only one telephone accesses the trunk.

Behind Switch Mode of Operation

■ One or more ground-start trunks are connected directly to the control unit. These trunks connect to a 400 GS/LS/TTR, 408 GS/LS, 800 GS/LS, or 408 GS/LS-MLX module (Release 2.0 only), The processor module must not be strapped for Key-only operation.

PF Classification

The system’s PF classification number is AS5USA-65646-PF-E. This classification is applicable only to the Hybrid/PBX mode of operation. The system is registered under the PF classification if either of the following conditions is met.

Hybrid/PBX Mode of Operation

■ All outside trunks are pooled; no trunks are terminated directly on a telephone.

■ The only directly terminated trunks are personal lines, not shared lines— only one telephone accesses the trunk.


Introduction

Programming

System Programming

The system can be programmed with options for the following:

■ Basic system operating conditions

■ System renumbering

■ Settings for lines/trunks

■ Telephones and operator consoles

■ Adjuncts

■ Applications

■ Optional features

The system can be programmed by using one of the following:

■ An MLX-20L telephone connected to one of the first five ports on the first

MLX module in the control unit

■ The built-in modem in the processor, which permits remote programming via the public network. For example, support personnel can access the system by using a PC with a modem and with SPM software; support personnel call the system and enter a password to gain access. The system must be programmed for Remote Access.

■ A PC with SPM software connected to the lower RS-232 port on the processor

The programming options are accessed from display screen menus. For more information, see System Programming. To use SPM to program your system on a personal computer, you need the SPM diskette and an PC with version 3.3 (or a later version) of MS-DOS® .

Your PC should include the following:

At least 640 kbytes of random access memory (RAM)

A floppy disk drive that will accommodate the SPM diskette

A serial port that can use either a DB-9 or DB-25 connector

NOTE:

For a DB-9 connector, use a 9-pin to 25-pin adapter to convert the

25-pin connector to a modular connector.

Either a 355AF modular adapter (if a male connector is on the interface cable) or 355A modular adapter (if the connector is female)

1-30 Programming

Introduction

■ A 4-pair modular cord (D8W)

The monitor can be either monochrome or color.

In addition, the following equipment is useful:

■ A parallel printer (the PC needs a parallel port for the connection)

■ A 1200-or 2400-bps modem

NOTE:

SPM uses Interrupt 4 and l/O address 3F8 for COM1. It uses Interrupt 3 and I/O address 2F8 for COM2.

When you use a PC with SPM to program the system, the maximum number of

MLX-20L telephones that can be connected to the system is reduced by one.

Telephone Programming

There are two kinds of telephone programming: Centralized Telephone

Programming and Extension Programming.

Centralized Telephone Programming

Centralized Telephone Programming is an option you can choose from the

System Programming menu to program any feature onto a telephone. Although many features can also be programmed by individual telephone users Or system operators, the following features can be programmed only by centralized telephone programming (and not by individual users):

■ Barge-In

■ Headset Hang Up

■ ICOM buttons—all types (Key and Behind Switch only)

■ System Access buttons—all types (Hybrid/PBX only)

Extension Programming

Extension Programming allows telephone users and system operators to tailor their telephones to meet personal needs. Multiline telephone users can assign a wide range of features to buttons on the telephone. In addition, many other features can be programmed on both multiline telephones and single-line telephones that do not require button assignment, such as Call Waiting.

Users can program theier telephones by dialing programming codes or, on MLX display telephones, selecting features from the display. When a telephone is in the program mode, the system considers it busy; therefore, no incoming calls ring at the telephone until it is back in the call-handling mode.

Programming 1-31


I

System

I

Restart

SProg Port

M o d e

● Key

●

Hybrid/PBX

● BehindSwtch

Board Renum

●

Mainten Busy

Enable

-Auto Busy Tie

Trunks:

- Enable

-Disable

●

Disable

Date

T i m e

I

SysRenumber

I

Default Numbering

●

2-Digit

●

3 - D i g i t

● SetUp Space

●

Single

Lines*

●

Extensions*

●

P o o l s *

●

Group Page*

●

GrpCalling*

● Adjuncts*

●

Park*

● ARS DialOut

●

RemoteAccs

● DSS Buttons*

● L i s t D i r c t N o

B l o c k

●

Lines

● Extensions

●

Adjuncts

Operator

Positions

●

Direct Line*

●

Queued Call*

Queued Call

● Hold Rtrn

- Return to

Queue

- Remain On

Hold

● HoldRelease

- Auto Hold

- Auto Release

●

Threshold

● ElvatePrior

● lnQue Alert*

- InQue Alert

Enable

-InQue Alert

Disble

● Call Types

- Dial 0

- Priority

- Operator*

- Follow/Frwd

- UnasslgnDID

- Priority

-Operator*

- ListedNumbr

- Priority

- Operator*

-QCC Ext

-Returning

- Priority

- Operator*

-GrpCoverage

- Priority

-Operator*

●

Msg Center*

●

ExtndComplt

- Automatic

Complete

- Manual

Complete

●

Return Ring

● QCC Backup

HoId Timer

DCL Hold

● Auto Hold Enable

●

Auto Hold Disable

LS/GS/DS1

●

(DS1)

-Type

-T1*

- Active

-Not Active

- ChannelUnit

-Foreign

Exchange

-Special

Access

●

(4xx GS/LS)

-GroundStart*

-Loop Start*

- All Ground

- All Loop

●

(8xx GS/LS)

- GroundStart*

- Loop Start*

- All Ground

- A l l L o o p

TIE Lines

●

Direction

-Two Way

- OutGoing

-Incoming

●

Intype

- Wink

- Delay

- Immed

-Auto

● Outtype

- Wink

-Delay

- Immed

-Auto

● E&M Signal

- Type 1S

- Type 1C

-Type5

● Inmode†

●

Outmode†

●

Dialtone†

●

AnsSupvr

●

Disconnect

- GroundStart*

- Loop Star*

- TIE

TIE-PBX*

- Toll*

- Unequipped*

- All Ground

- All Loop

- All TIE

- TIE-PBX

- Toll

-All Unequlp

D I D *

- All DID

- PRl

- FrameFormat

-D4

Compatible

-Extended

Super Frame

- Suppression

- AMI-ZCS

- B8ZS

- Signaling

- Robbed Bit

- Common

Channel

- Line Comp

- Clock Sync

- Priority

- Primary

-Secondary

- Tertiary

-None

- Source

- Loop

-Local

- Activation

I

LinesTrunks Extension O p t i o n s

TT/LS Disc

● OutMode†

●

LS Disconnect

- Yes

-No

DID

●

Block†

●

Type

- lmmed

- Wink

●

Disconnect

●

ExpectDgit

●

DeleteDigit

●

Add Digits

●

Signaling

- Rotary

- Touch Tone

● lnvalDstn

- Send to

Backup

Extension

-Return FastBusy

P R l

●

PhoneNumber

● B-ChannlGrp

- B Channels*

- Lines*†

- NetworkServ

- AT&T Toll

-Megacom

WATS

- ACCUNET

SDS

- Soft DefNetw

- Megaom

800

- MULTI

QUEST

- Long Distnce

-Local

- OUTWATS

- 56/64 Digtl

- VirtPrivNet

- IN WATS

-Misc

- Other

-CallByCall

- Copy Number

- Copy PhnNum to NumToSend

- Do not Copy

-Phone Number

-lncomingRtg

- Routing by Dial

Plan

-Route by Line

Appearance

●

NumbrToSend

- Extension Only

-Base Number with Ext.

- Line Telephone

Number

●

Test TelNum

● Protocol

- Timers

- T200 Timer

- T203 Timer

- N200Counter

-N201Counter

- K Counter

- T303 Timer

-T305 Timer

- T308 Timer

- T309 Timer

-T310 Timer

- T313 Timer

- T316 Timer

-TEI

●

DiaIPlanRtg

-Service

- AT&T Toll

- Megacom

800

- ACCUNET SDS

-SoftDefNetw

-LongDistnce

- Local

-OUTWATS

- 56/64 Digtl

- VirtPrivNet

-M

ISC

-Other

- No Service

- Delete Digit

Copy

●

Single

●

Block

RemoteAccss

● LinesTrunks *

- Dedicated

- Shared

-No Remote

●

Non-TIE

- BarrierCode

- Barrier Code

Required

-BarrierCode

Not Required

- Restriction

-Unrestricted

- Outward

Restrict

-Toll Restrict

-ARS Restrct

- Allow List*

-DisallowLst*

● TIE Lines

- BarrierCode

- Barrier Cede

Required

-BarrierCode

Not Required

- ACCUNET

SDS

- SoftDefNetw

-MULTI

QUEST

- Megacom

WATS

- LongDistance

- Local

- INWATS

-56/64 Digtl

- VirtPrivNet

- OUTWATS

-M ISC

- Other

- Any Service

- No Service

- Patterns

- TotalDigits

- DeleteDigit

-Add Digits

●

OutgoingTbl

-NetwkSelect

- SpecialServ

Pattern

- Operator

- Local

Operator

-Presubscribed

Carrier

- No Operator

-TypeOfNumber

-National

-International

-DeleteDigit

- CBC Service

-Patterns

-Voice/Data

-VoiceOnly

-Data Only

- Voice/Data

-NetworkServ

-AT&T Toll

- Megacom

WATS

LinesTrunks†

Line copy

●

Single

●

Block

Dial OutCd†

Restriction

●

Unrestricted

● Outward Restrict l Toll Restrict

RestrctCopy l Single

●

Block

Account*

BIS/HFAl*

Call PickUp*

VoiceSignl

Ext Status*

Group Page*

Group Cover*

Grp Calling

● Hunt Type

- Circular

- Linear

●

DelayAnnce

●

GrpCoverage†

●

Message

●

Queue Alarm

●

Xtnl Alert

● Overflow

●

Members*

● Line/Pool*

●

Group Type

- Auto Login

-Auto Logout

- Integ VMI

-Generic VMI

ARS Restrct

Mic Disable*

Remote Frwd*

Lines Trunks

Continued

-Restriction

-Unrestricted

- Outward

Restrict

- Toll

Restrict

- ARS Restrct

- Allow L IS t*

-DisallowLst*

●

BarrierCode

-SProg/Maint

- Codes

- Restriction

-Unrestricted

- Outward

Restrict

- Toll Restrict

- ARS Restrct

- Allow List*

●

-DisallowLst*

AutoQueuing

- Enable

- Disable

Pools†

Toll Type†

HoldDiscnct†

PrncipalUsr

QCC Prior†

QCC Oper†

●

Format

- Basic SMDR

-ISDN SMDR

● Call Length

●

Call Report

- In/Out

-Out Only

●

New Page

InsideDial

●

Inside

●

Outside

ReminderSrv

Unassigned

●

QCC Queue

● Extension

● Grp Calling

BehindSwitch

●

Transfer

●

●

Conference

D r o p

RecallTimer

Transfer

● Return Time

●

One Touch

- Transfer

- Manual

- Automatic

- Hold

● Audible

- M USIC On

Hold

- Ringback

●

Type

- Voice

Announce

- R i n g

C a m p O n

C a l l P a r k R t n

Delay Ring

Callback

Ext Status

● Hotel l GrpCall/CMS

SMDR

●

350 ms

●

450 ms

●

650 ms

●

1 sec

Rotary

●

Delay

● No Delay

Cover Delay

T a b l e s

Allow List

AllowTo†

Disallow

DisallowTo†

ARS

●

ARS1+7Dial

- Within Area

Code

-Not Within

Area Code

● ARS Input

-6 -Digit

- Area Code†

- E.xchange†

- 1+7†

●

Sub A Pools

● Sub A FRL

●

SubAAbsorb

●

Sub A Digit

●

Sub B Start

● Sub B Stop

●

Sub B Pool

● Sub B FRL

●

SubB Absorb

● Sub B Digit

●

SpeclNumber

- ARS FRL

- ARS Digit

●

Dial 0

- ARS Pool

-ARS FRL

- ARS Digits

●

Sub A Data

- Vote Only

-Data Only

-Voice/Data

●

Sub B Data

- Voice Only

- Data Only

-Voice/Data

AuxEquip

MusicOnHold

Ldspkr Pg*

Fax

●

Extension*

●

●

Meg Waiting*

Threshold

MaintAlarms

VMS/AA

●

TransferRtn

● TT Duration

●

TT Interval

NightSrvce

GroupAssign†

●

Extensions*

● Calling Grp

OutRestrict

Emergency

ExcludeList*

Start*

Stop*

Day of Week

Labeling

Directory

● System

● Extension

● Personal

LinesTrunks

PostMessage

Grp Calling

Data

Voice/Data*

Print

A l l

SysSet-up

Dial Plan

Lables

Trunk Info

●

T I E

●

●

D I D

Loop/Ground

●

General

T1 Info

PRI Info

RmoteAccess

Oper Info

A l l o w L i s t

A l l o w T o

DisallowLst

DisallowTo

A R S

Ext Direct

Sys Direct

Group Page

E x t I n f o

GrpCoverage

Grp Calling

Night Service

Call Pickup

ErrorLog

Cntr-Prg

Program Ext

Copy Ext

* The Inspect feature can be used with this menu option. Press Inspect or PgDn.

† The Inspect feature can be used in entry mode with this menu option, Press

Inspect or PgDn while in entry mode.

Language

SystemLang

●

●

English

French

●

Spanish

Extensions

●

Single

- English

- French

- Spanish

● Block

- English

- French

-Spanish

SMDR

●

English

● French

●

Spanish

Printer

● English

●

French

● Spainsh

1-32 Programming

Introduction

System Capacities and Requirements

This section details the technical requirements and capacities of the system:

■ Hardware and software capacities for the system

■ Environmental requirements for placement of the control unit

■ Power and grounding requirements for operating the system

Capacities

The system can be arranged as a stand-alone system or as part of a private network. Maximum system capacities are as follows:

■ Up to 108 simultaneous two-party conversations

■ Up to 80 line/trunk jacks, including loop-start, ground-start, DID, and tie

■ Up to 255 station endpoints that support a combination of the following:

— Up to 144 physical station jacks for telephones and adjuncts

— Up to 127 logical digital data ports (via 7500B Data Modules connected to jacks on the MLX module) providing RS-232 connections to data terminals and personal or multiport computers

■ System call-handling capability of 3888 hundred call seconds per hour

(ccs/hr)

■ Up to three 100D DS1 modules

The system has a total capacity of 224 jacks (80 outside lines/trunks plus 144 stations); however, each MLX module station jack supports two logical endpoints (station devices that can operate simultaneously and independently).

For example, an MLX telephone with a Multi-Function Module (MFM) plugs into only one station jack, but the jack supports the telephone and the equipment connected to the MFM (such as a fax machine or an answering machine).

In a similar way, although the 100D module has only one jack, it can serve up to

24 endpoints (emulated lines/trunks or PRI lines/trunks).

Thus, the entire system can be configured to connect up to 80 lines/trunks and

255 station endpoints, a total of 335 endpoints.

NOTE:

The system has a time-slot capacity of 216. If more than 216 endpoints are in use at the same time, blocking can occur.

The following table lists the hardware and software capacities of the system.

System Capacities and Requirements 1-33

Introduction

Table 1-3, Hardware and Software Capacities

Limit

Allowed Lists

■ Number of lists

■ Entries per list

■ Digits per entry

Automatic Route Selection

■ Number of ARS patterns

■ Subpatterns per pattern

■ Routes per subpattern

■ Number of fully programmable ARS tables

■ Entries per table

■ Entries across all tables

■ Default tables

Callback

■ Number of calls in queue

Calling groups

■ Number of groups

■ Members per group

■ Groups per member

■ Delay announcements per system

■ Delay announcements per group

■ Groups per delay announcement

■ External alerts per group

■ Coverage groups per group

Carriers

■ Line/trunk and station module slots per basic earner

■ Line/trunk and station module slots per expansion carrier

■ Maximum slots available for line/trunk and station modules

Coverage groups


■ Senders per group

■ Groups per sender

■ Receiver buttons per group

■ Groups per QCC receiver

Data hunt groups




Direct Inward Dialing

■ Number of blocks

■ Number of trunks

Directories

■ System Directory

■ Number of directories

■ Listings per directory

■ Extension Directory


1

130

1

8

10

7

32

20

1

32

1

32

1

1

3

5

6

17

16

100

1600

4

18

2

6

64

30

144

1

8

30

32

20

1

2

80

Constraining Factor

QCCs cannot be members

First slot of basic earner used for processor module

QCCs cannot be senders


1-34 System Capacities and Requirements

Introduction



■ Personal Directory (MLX-20L only)



Disallowed Lists

■ Number of lists

■ Entries per list


100D module (maximum 2 per carrier)

Endpoints (devices)

Fax machines with message-waiting

Lines/Trunks

Night Service

■ Groups

■ Members per group (including one group calling number)


■ Emergency Allowed List entries

System operator consoles

■ DCLs

■ MLX-20L or MLX-28D

■ BIS-22D, BIS-34, BIS-34D, or MERLIN II

System Display Console

QCCs

DSSs

Limit

144

48

50

8

10

11

3

255

16*

80

8

144

8

10

8

8

4

16

Constraining Factor

Software real-time limit

■

■

Combination of DLCs and QCCs

Number of consoles per module

Park codes

■ Number of codes

Personal lines

Pickup




Pools (trunk groups)

■ Maximum number of pools

■ Maximum number of trunks in a pool

Pool buttons

8

2

8

64

30

15

11

1

80

64

*

The system can support more than 16 fax machines, but those in excess of 16 cannot use the fax message-waiting indication.

2 per MLX module

2 per analog module

2 per MLX module

2 per MLX module

(built into MERLIN II System

Display Console)


System Capacities and Requirements 1-35

Introduction


Limit Constraining Factor

Ports (not achievable simultaneously)

■ Total

■ Voice and data (physical ports)

■ Voice Announce to Busy stations

■ Voice-mail interface

■ Digital data via 7500B Data Module

■ Paging

■ Delay announcements

Remote Access

■ Number of barrier codes

■ Digits per code

224

144

127

20*

127

3

32

16

4



RAM limit

RAM limit

Software real-time limits, loop-start only

Software real-time limits

Shared System Access buttons

■ Number of buttons per principal station

Speed Dial

■ Personal Speed Dial

16

Single-line and 5- or 10-button telephones only

■ Entries per telephone

■ Entries per system


■ System Speed Dial

■ Entries per system


Stations

■ Total physical jacks

■ Total endpoints

System programming equipment

■ MLX-20L

■ RS-232 jack (for connection of PC with SPM)

■ Modem (built-in processor module)

24

1200

28

130

40

144

255

1

1

1

Remote access overrides on-site programming except during backup or restore

Telephones (not achievable simultaneously)

■ Single-line

■ Analog multiline

Without Voice Announce to Busy

With Voice Announce to Busy

■ MLX-20L

■ All other MLX telephones

(with or without 7500B Data Module or MFM)

■ Power failure transfer

Traffic (hundred call seconds/hr/system)

Two-party conversations

144

136

68

48†

127

20

RAM limit

17 slots x 8 ports/board

RAM limit

RAM limit

1 per 4 LS/GS trunk jacks

3888 ccs/hr. Assuming 20% internal traffic

108 216 time slots

Voice-mail systems 1

‘ Although the system software supports up to 24 voice-mail interface

(VMI) ports, all the VMI ports must be in the same calling group, and the maximum number of stations in a calling group is 20.

† Total includes the MLX-20L telephone used for system programming.

1-36 System Capacities and Requirements

Introduction

Environmental Requirements

The control unit requires a regulated environment and can be located in any room or closet that is temperature controlled and clean. Do not mount the control unit where it will be exposed to direct sunlight.

In addition, the control unit should not be co-located with air conditioning or ventilation units, compressors, fans and blowers, heaters, arc welders, or other such machinery that produces electrical interference.

The control unit is mounted on a customer-provided plywood backboard. The backboard should be wide enough to accommodate additional carriers if system growth is anticipated. Allow enough space on either side of the control unit for any necessary wiring fields.

Once installed, it is important to keep the control unit site clear of hazards, such as stacked paper or boxes, that block ventilation. Installing any machinery in the vicinity of the control unit should be avoided. If any pollution-producing work (such as sanding or spray painting) is to be done in the area, care should be taken to protect the unit.

Table 1-4 gives the environmental specifications for the control unit.

Table 1-4. Environmental Specifications

Control Unit

Fully loaded basic carrier

Fully loaded 2-carrier system

(basic carrier plus one expansion carrier)


(basic carrier plus two expansion carriers)

45lb (20.4kg) 14" W x 23" H x 12"D

(35.6 cm x 58.4 cm x 30.5 cm)

90 lb (40.8 kg) 25" W x 23" H x 12"D

(63.5 cm x 58.4 cm x 30.5 cm)

135lb (61.2 kg) 37" W x 23" H x 12"D

(94 cm x 58.4 cm x 30.5 cm)

Mean Time Between Failures 2.1 years

(mean or average time the system is expected to operate before any type of failure occurs)

Mean Time Between Outages 4.0 years

(mean or average time the system is expected to operate before a failure affecting more than 25% of extensions or lines for more than 15 seconds occurs)

Backboard (minimum needed)

■ Without SYSTIMAX®

■ With SYSTIMAX

6' W x 3' H x 3/4" D(182.9 cm x 91.4 cm x 1.9 cm)

7' W x 4' H x 3/4" D(213.4 cm x 121.9 cm x 1.9 cm)



Introduction


Backboard Mounting Hardware Requirements

Wood surface Wood screws

Concrete surface, brick, cinder block

Plaster, plasterboard

Sheet-metal surface

Masonry anchors

Toggle bolts

Sheet-metal screws

■ Hardware should have a combined pullout force of 650 lb (294.8 kg).

■ When mounting to sheet-metal walls, attach to structural members.

Location

■ Within 5 feet (1.5 meters) of dedicated AC power outlet (1 plug per carrier)

■ Within 1000 cable feet (304.8 m) of telephones

Heat Dissipation

Fully loaded basic carrier


(basic carrier plus one expansion carrier)


(basic carrier plus two expansion carriers)

500 Btu/hr

1000 Btu/hr

1500 Btu/hr

(35 cal/sec)

(70 cal/sec)

(105 cal/sec)

Power Requirements

Basic carrier

2-carrier

3-carrier

Temperature/Humidity Range

40°-104°F (4°-40°C)

20%-80% relative humidity

Ventilation

1 inch (2.5 cm) on right and left sides

117VAC

117VAC

117VAC

6 0 H z ± 5 % 3 A

6 0 H z ± 5 % 6 A

60 Hz±5% 9A

Radio Frequency Interference (RFI) Tolerance 1 V/m

In most cases, electrical noise is introduced to the system through trunk or telephone cables, However, electromagnetic fields near the control unit may also induce noise in the system. Therefore, the control unit and cable runs should not be placed in areas where a high electromagnetic field strength exists. Radio transmitters (AM or FM), television stations, induction heaters, motors (with commutators) of 0.25 horsepower (200 watts) or greater, and similar equipment are leading causes of interference. Small tools with universal motors are generally not a problem when they operate on separate power lines. Motors without commutators generally do not cause interference. Field strengths below 1.0 volts per meter are unlikely to cause interference.

The field strength produced by radio transmitters can be estimated by dividing the square root of the emitted power in kilowatts by the distance from the antenna in kilometers. This yields the approximate field strength in volts per meter and is relatively accurate for distances greater than about half a wavelength (150 meters for a frequency of 1000 Hz).


Introduction

CAUTION:

Do not use switch control on AC outlet for control unit.

Use approved ground (AC receptacle for 3-prong plug).

Do not install control unit outdoors.

Do not place control unit near extreme heat (furnaces, heaters, attics, or direct sunlight).

Do not expose control unit to devices that generate electrical interference (such as arc welders or motors).

Do not place anything on top of carriers.

Do not install control unit under any device that may drip fluid, such as an air conditioner.

Each auxiliary power unit requires one outlet.

Do not expose the control unit to moisture, corrosive gases, dust, chemicals, spray paint, or similar materials.

Power and Grounding

Proper power and grounding are essential for correct and safe functioning of the system.

Power Specifications

The system control unit plugs into a 117-VAC outlet. To avoid accidental disconnection of the system, this outlet should not be controlled by a wall switch.

Each carrier unit requires its own power supply. Each power supply requires a maximum current of 3 amps. Therefore, if expansion carrier units are added to the system, extra AC outlets maybe needed.

Grounding Requirements

Proper grounding of the installation site protects the system against the following:

■ Lightning

■ Power surges

■ Power crosses on outside lines/trunks

■ Electrostatic discharge (ESD)


Introduction

The telephone company is responsible for providing protection of outside lines/trunks at the entrance to the site. The protection should consist of the following:

■

Carbon blocks or gas discharge tubes connected to an approved ground

■ Adequate bonding of the outside line/trunk protector ground and the power company ground

WARNING:

An improper ground can result in equipment failures and service outages. Verify that the AC power uses an approved ground for its primary ground, that all voltage-limiting devices are grounded to an approved ground, and that the ground is one of the approved grounds below.

The following is a list of approved grounds, starting with the most preferred:

■ Building steel

■

Acceptable water pipe-must be a metal, underground water pipe at least 1/2-inch (30.4 cm) in diameter, and in direct contact with the earth for at least 10 feet (3 meters).

It must be electrically continuous so that the protector ground is connected. (Check for insulated joints, plastic pipe, and plastic water meters that might interrupt electrical continuity.)

A metallic underground water pipe must be supplemented by the metal frame of the building, a concrete-encased ground, or a ground ring. If these grounds are not available, the water pipe ground can be supplemented by one of the following types of grounds:

— Other local metal underground systems or structures— local underground structures such as tanks and piping systems

—

Rod and pipe electrodes—a 5/8-inch (1.6-cm) solid rod or 3/4-inch (1.9-cm) conduit or pipe electrode driven to a minimum depth of 8 feet (244 cm)

— Plate electrode—a minimum of 2 square feet

(61 square cm) of metallic surface exposed to the exterior soil


Introduction

■ Concrete-encased ground--must be an electrode, consisting of one of the following:

— At least 20 feet (6.1 meters) of one or more steel reinforcing rods, each being at least 1/2-inch (1.27 cm) in diameter

— 20 feet (6.1 meters) of bare copper conductor not smaller than #4 AWG, encased in 2 inches (5 cm) of concrete.

— This electrode must be located within and near the bottom of a concrete foundation or footing that is in direct contact with the earth.

— Ground ring---consists of at least 20 feet (6. 1 meters) of bare copper conductor not smaller than #2 AWG encircling the building. The ground ring must be in direct contact with the earth and buried at least 2.5 feet (77 cm) below the earth’s surface.

WARNING:

Do not use metal underground gas piping system—this is a safety risk.

For most surge occurrences, the following standard grounding requirements provide adequate lightning and power surge protection:

■ Properly wired/grounded/bonded outside line protectors

■ Properly wired/grounded AC outlet

■ Properly grounded single-point ground bar

■ Properly wired connection between single-point ground and power supplies

Additional Power Surge Protection

The 391A1 power supply has built-in AC line protection. This built-in protection handles almost all situations.

occasionally, additional protection may be needed if the customer is located in a heavy lightning area. A 147A surge protector can be connected to the system to limit surges from the AC lines and outside lines. One 147A protector provides protection for four outside lines. Up to three 146A protectors can be added to the 147A to provide protection for a maximum of 16 outside lines. For more than sixteen lines, additional 147A protectors are required.


Introduction

NOTE:

The 147A protector is usually not needed with the 391A1 power supply. It may be needed with the older 391A power supply module in heavy lightning areas.

Complete installation instructions are provided with the protectors.

Unit Loads

A unit load is a measure of power (1.9 watts) used to determine the electrical load that the following components have on each carrier’s power supply:

■

Telephones and adjuncts

Only the telephones and adjuncts that connect to the analog and digital ports on the control unit require unit load calculation. Do not include any equipment with its own power supply, for example, a fax machine, an

MFM, or an answering machine, in the unit load calculation.

■ 800 DID modules

Before installation, unit load and auxiliary power requirements for a new system are computed by qualified service personnel or an authorized dealer, and any necessary auxiliary power equipment is ordered automatically. However, in the event of maintenance or equipment changes, unit loads should be calculated to ensure proper operation under all conditions.

The power supply module provides 45 unit loads to each carrier. If the unit load requirement per carrier exceeds 45, an auxiliary power unit is needed to allow the carrier to support an additional 27 unit loads.

CAUTION:

Running the system with more than 45 unit loads per carrier may not appear to do harm. However, this can cause the system to malfunction, creating “NO Trouble Found” situations.

An auxiliary power uni redirects the power requirements from the last two slots on the carrier. Any telephone connected to the modules in the last two slots receives power from the auxiliary power unit instead of from the power supply module.

Checking Unit Loads

In the event of maintenance or equipment changes, recalculate the unit loads for each carrier resulting in a different configuration

Use the worksheet in Appendix B of Installation.


Introduction

General Rule: If you can distribute the 800 DID modules and telephone modules equally across the carriers, you will prevent unnecessary drain on any one carrier.

Also, depending on the system’s mode, the rules vary. The next two sections provide the rules for calculating unit loads in various modes.

Unit Loads for the Hybrid/PBX Mode

The power supply module generally supports six modules of any type in a

Hybrid/PBX system—without requiring an auxiliary power unit.

If, however, both of the following conditions are true, the unit loads on a carrier can exceed the 54-unit maximum, and therefore require auxiliary power:

■ All six carrier slots are occupied by MLX telephone or analog multiline telephone modules

■ The carrier has a total of more than 45 MLX-20L telephones or 34-button analog multiline telephones installed

Unit Loads for Key or Behind Switch Mode

In a Key or Behind Switch system with four or fewer modules, no calculation is needed, The power supply module generally supports four modules of any type in Key or Behind Switch mode.

Release Differences

Release 1.1 Enhancements

Release 1,1 includes all Release 1.0 functionality plus the following enhancements:

Language selection allows the system to be programmed for prompts, menus, and messages on MLX display telephones to appear in English, French, or

Spanish. Each of the following can also be programmed for any of these languages, independently of the system language:

■ Individual extensions with MLX telephones

■ SPM

■ System programming reports

■ SMDR report headers

MLX-10D, MLX-20L, and MLX-28D display telephones and MLX-10 non-display telephones are available in three separate versions, with factory-imprinted buttons in English, Spanish, or French.

Release Differences 1-43

Introduction

In addition, user and operator guides, quick reference cards, and telephone tray cards are available in all three languages.

Programming and maintenance enhancements include the following:

■

Additional Inspect capability in system programming

■ Editing capability (Backspace selection) in extension programming

■ Improvements to system reports

■ An access log that records the last 20 times maintenance or system programming has been accessed

■

Longer (20-second) gap between ring cycles for Program Mode and

Forced Idle tone

System operational enhancements include the following:

■ Automatic selection of an SA button when Conference is invoked (in

Hybrid/PBX mode)

■

Prompting through Conference feature (on MLX display telephones)

■ Relocation of the More prompt on the MLX-20L display

■ Display of the number saved on a programmed Last Number Dial or

Saved Number Dial button when the button is Inspected

SPM enhancements include operation in English, French, or Spanish, faster backup and restore, and automatic on-screen display of reports as they are created, with a Browse capability for reading the reports.

Additional equipment includes the 8102 and 8110 analog voice telephones, four headsets, two headset amplifiers, and a transparent protective cover for the

MLX-10 and MLX-10D telephones. The 8102 and 8110 telephones are backward compatible with Release 1.0.

PF registration number AS5USA-65646-PF-E is assigned by the FCC for operating the MERLIN LEGEND Communications System in Hybrid/PBX mode in the United States. (The PF registration is also applicable to Release 1.0

systems.)

The Release 1.1 enhancements are described in detail in MERLIN LEGEND

Communications System Release 1.1 Notes (555-610-119).

Release 2.0 Enhancements

Release 2.0 includes all Release 1.1 functionality plus the following enhancements:

Programming enhancements include the following:

■ Extension Copy is a new feature that reduces programming time by allowing the use of any extension as a template for programming another extension or block of extensions through centralized programming.

1-44 Release Differences

Introduction

■ Integrated Administration provides a single interface through Integrated

Solution Ill (IS-III) for programming entries common to the MERLIN

LEGEND Communications System and AUDIX Voice Power.

■ Any SPM Version 2.

XX

(where xx is replaced by numbers) provides a

Convert function for use in upgrading the system from Release 1.0 or 1.1.

This function converts a backup file from a Release 1.0 or 1.1 system to

Release 2,0 format, allowing reuse of existing system programming on the upgraded system.

■ Forced idle reductions keep system interruptions at a minimum. In general, the smallest necessary component is forced idle during programming activities. For example, renumbering a single extension force-idles only one extension. Only a few system-wide programming activities, such as setting the system mode and system renumbering, force-idle the entire system.

System operational enhancements include the following:

Coverage VMS is a new feature that prevents incoming external calls from going to voice mail. (All other Coverage remains active as programmed.) The feature is programmed extension-by-extension, either through extension programming or through centralized programming.

A Night Service group can be programmed to include a calling group as a member. This allows a call that receives Night Service treatment to be queued when all Night Service group members are busy or unavailable.

When AUDIX Voice Power sends a Leave Word Calling message to an extension, the system identifies the voice mail system as the sender of the message. When the voice mail subscriber uses the Return Call feature, the call goes to any available voice mail port, not just to the specific port that generated the message. This reduces the chance of getting a busy port.

Coverage receivers can call Coverage senders and have the call receive

Coverage treatment. If a receiver calls a sender for whom he or she is covering, and the sender is busy or unavailable, the call proceeds to other points of Coverage. it does not come back to the receiver who originated the call.

Enhancements to display prompts include automatic posting of a

DO

Not Disturb message when a user activates the Do Not Disturb feature, and confirmation messages when a user activates Hold, Privacy,

Saved Number Dial, and Transfer.

Direct Inward Dialing (DID) trunk emulation on a T1 facility provides

24 DID channels on a single DS1 trunk interface, instead of requiring

24 separate physical trunks.

Release Differences 1-45

Introduction

■

A telephone user can send a timed flash (switchhook flash) on a loopstart trunk call on a System Access (SA) button.

FAX Attendant System is a new application for sending and receiving fax messages; its interface is similar to the voice mail interface provided by AUDIX

Voice Power. FAX Attendant System, which co-resides with AUDIX Voice Power on the IS-III platform, provides the following services:

■ Fax Call Answer receives and holds messages for subscribers whose fax machines are busy or out of paper. This service also allows a subscriber to have a personal fax number without having a fax machine.

■

Fax Mail allows subscribers to create and use fax distribution lists, send and receive fax messages, and record personal greetings for incoming fax calls.

■ Fax Response prompts callers to select and receive faxes from a customer-created menu of choices, using touch-tone responses.

408 GS/LS-MLX module (Release 2.0 only) is a new module that combines four ports for ground-start or loop-start trunks and eight ports for MLX telephones on a single module in the control unit.

Primary Rate Interface (PRI) enhancements include the following:

■ Connectivity to the 5ESS® Generic 6

■ Multiple incoming calls to directory number

■ Call-by-Call Service Selection

■ Authorization Code handling for FTS2000

■ Station ID (SID) as Calling Party Number for Automatic Number ID (ANI)

Maintenance enhancements include the following:

■ Clear descriptions of module test failures

■ Optional printing of hard copy of error logs

■ Display that correlates extension numbers to slot/port and logical ID

■ Display showing which slots, trunks, and extensions are maintenancebusy

■ Internal digital switching element (DSE) Ioopback test for all modules

■ B-channel Ioopback test for MLX modules

■ B-channel line or call service states display

■ Error log entries for dual-port RAM errors

1-46 Release Differences


This chapter describes the basic hardware required for the communications system. It includes the control unit, digital or MLX telephones, analog multiline telephones, single-line telephones, system operator consoles, adapters and adjuncts for system telephones, and power-related accessories.

Control Unit

The control unit connects telephone company lines with telephones and adjuncts such as answering machines and fax machines. The control unit includes the following components:

■

Carriers

■ Processor module (one per system)

■ Power supply module (one per carrier)

■

Line/trunk and station modules

■

Covers

Carriers

The basic and expansion carriers each have seven slots to hold modules (see

Figure 2-1 ). The basic carrier contains a power supply, the processor (slot 00),

and line/trunk and station modules (slots 01-05).

Up to two expansion carriers can be added to the right side of the basic carrier to increase the capacity of the system. Like the basic carrier, the leftmost and widest slot of the expansion carrier holds the power supply; the remaining six slots hold the line/trunk and station modules.

Control Unit 2-1

Hardware Components

Besides the slots, both basic and expansion carriers have a backplane with an input/output (I/O) bus that interfaces with the modules.

BACKBOARD

Figure 2-1. Carriers

Processor Module

Placed in slot 00 of the basic carrier, the processor module controls system features and programming, The main component of the processor module is the feature module, This component provides all the system’s release-specific capabilities and features. All system programming is stored in the feature module in non-volatile memory.

The feature module plugs into the main board of the processor module, which contains the 66000 microprocessor, a built-in 1200 bits-per-second (bps) data modem, built-in diagnostics, RAM, a real-time clock, and interrupt circuitry, and interfaces to the other modules through the I/O bus on the carrier backplane.

2-2 Control Unit

Hardware Components

The processor has two modular RS-232 jacks: one for Station Message Detail

Recording (SMDR) and the other for system programming and maintenance via a personal computer (PC) with SPM software (see Figure 2-2).

Figure 2-2, Processor Module

A NiCad battery in the processor provides backup power for the real-time clock and nonvolatile RAM in case of power failure or system shutdown. The battery provides RAM data retention for 12 to 30 days. The trickle-charge circuit can recharge the battery to 50 percent of capacity from a discharged state in 48 hours. The minimum battery life is five years.

Power Supply Module

The power supply provides power to the carrier, to each telephone, and to adjuncts—except for adjuncts such as answering machines and fax machines, which come with their own power supplies. Each carrier requires its own power supply module, which goes into the leftmost slot on each carrier.

The power supply converts 117-VAC line voltage to these outputs: +5 VDC, -5

VDC, and -48 VDC. All modules use +5 VDC and -5 VDC for logic circuits.

Most line/trunk and station modules use -48 VDC for power to the stations. The

Direct Inward Dialing (DID) and off-premises telephone (OPT) line/trunk and station modules also provide -48 VDC on the tip/ring (T/R) interface to the telephone company’s central office (CC)) or OPT station. The 012 module basic telephone module provides 21 VDC to single-line telephones and equipment.


Hardware Components

When the system contains a 012 or 008 OPT module, a 129B Frequency

Generator (ring generator) must be installed in the power supply module of each carrier that houses one or more of these modules.

A green light-emitting diode (LED) on the power supply remains on as long as the module is receiving power. The power supply also has an on/off switch and a modular telephone jack for connecting an auxiliary power unit as needed (see

Figure 2-3).

Figure 2-3. Power Supply

Line/Trunk and Station Modules

The line/trunk and station modules have jacks for connecting the telephone company lines/trunks and the station wires to the control unit. The station wiring connects to individual telephones and to adjuncts such as answering machines and fax machines.

A system with a basic carrier has five slots for modules. Up to two expansion carriers can be added, each one adding six slots for line/trunk or station modules.

2-4 Control Unit

Hardware Components

Different line/trunk modules support different types of telephone company trunks. The types of trunks include the following:

Loop-start trunks— (incoming and outgoing calls) the simplest and most common facilities in the nation-wide telephone network. They provide incoming and outgoing calls and are intended primarily for single-line telephones and older private branch exchanges (PBXS). A potential problem of loop-start trunks is glare, that is, picking up the telephone to make a call and another caller on an incoming call is already on the line.

Glare occurs because of delays in the telephone company’s ringing cycles and disconnects. This is normally not a problem for a residential, single-line telephone, but can be more serious and complex with an automated PBX.

Ground-start trunks—(incoming and outgoing calls) specifically introduced to solve the problems that PBXs encounter on Imp-start trunks, They provide an immediate signal when the trunk is seized and when the call is completed and disconnected. Ground-start trunks can be used only if registered with the FCC. Depending on the mode of operation and the hardware strap in the processor module, the registration number (provided by the account representative or

authorized dealer) includes the letters KF, MF, or PF. See FCC

Registration in Chapter 1 for details.

Tie trunks- a private line that directly connects two communications systems. Thus, a caller on one system can call an extension on another system by dialing an access code and the extension number. In more complex tie trunk configurations, a user can access a facility on the other system that does not exist on their own system.

Direct Inward Dialing (DID) trunks--- (incoming calls only) provide fast access to specific individuals, that is, incoming calls can be routed directly to the called extension, a calling group, or an outgoing trunk without system operator assistance, These trunks are reliable and efficient, but are more complex than loop-start or ground-start trunks.

Therefore, their installation and maintenance must be coordinated with the telephone company,

DS1 connectivity programmed for either T1 or Primary Rate Interface

(PRI) operation-( incoming and outgoing calls) provides two-way connection and high speed transmission of analog and digital signals simultaneously. One trunk provides 24 channels; services provided on the channels can be assigned and subsequently changed by you instead of the telephone company, T1 operation enables the system to transmit and receive voice and analog information; PRI enables the system to transmit and receive voice, analog, and digital data, PRI provides a wide range of benefits not available on any other single type of trunk; for example, access to services on the channels can be on a call-by-call basis, with the system selecting the most efficient or costeffective channel for that call.

The system supports 14 types of line/trunk and station modules. Figure 2-4

shows the line/trunk and station modules. Table 2-1 lists the type and number


Hardware Components of jacks for each type of module. The names of modules are numbers that identify their connectivity and port capacities. The first digit is always the number of trunk jacks, while the third (last) digit is the number of station ports supported. For example, the 408 GS/LS module provides four trunk jacks and eight station jacks and supports ground-start or loop-start trunks.

-1.

4-pair

Jacks

(DS1)

-2.

T i e -

Trunk

Jacks

- 3 .

Trunk

Jacks

( L S )

- 4 .

PFT Jacks

-5.

T r u n k

J a c k s

(GS/LS)

- 4 .

PFT Jacks

- 6 .

Trunk

(DID)

Jacks

-6.

Trunk

(DID)

Jacks

PFT Jack

3 .

T r u n k

Jacks(LS)

- 3 .

T r u n k

Jacks(LS)

-4.

PFT Jacks

- 7 .

Off-Premises

Telephone

(OPT) Jacks

-7.

Off-Premises

Telephone

(OPT) Jacks

- 2 .

T i e -

Trunk

Jacks

100D 400EM 400 400 800 DID 800 008 OPT

GS/LS/TTR

PFT Jack

Line Jacks

(GS/LS)

Line Jacks

(GS/LS)

PFT Jack

Analog

Station

Jacks

Digital

Station

Jacks

Basic

Telephone

J a c k s

PFT Jack PFT Jack PFT Jack

Trunk

Jacks

( L S )

T r u n k

J a c k s

(GS/LS)

Trunk

Jacks

(GS/LS)

Analog

Station

Jacks

Analog

Station

Jacks

Digital

Station

Jacks

8 0 0 0 0 8 0 0 8 M L X 0 1 2 4 0 8 4 0 8 4 0 8

GS/LS GS/LS GS/LS-MLX

Figure 2-4. Line/Trunk and Station Modules

2-6 Control Unit

Hardware Components

Table 2-1. Line/Trunk and Station Modules

008

Module ma MLX

008 OPT 1

012

100D

400 2

Line/Trunk Type

N/A

N/A

Station Type

Analog multiline telephone; CMS

Specifications

Capacity: 8 analog stations

Signaling: analog multiline telephone protocol (40 kbps)

Loop range: 1000 feet (305 meters). For In-Range Out-of-Building service, use analog I ROB protectors, over 1000 feet (305 meters), use Off-Premises Range Extender (OPRE).

MLX telephone; digital Capacity: 8 digital stations, each with 1 or data device (such as

7500B Data Module) following station types:

■ digital voice only

■ digital voice with Voice Announce to Busy feature

■ digital voice and digital data (via the 7500B Data Module)

■ digital voice and Multi-Function Module (MFM)

■ digital data only (7500B Data Module)

Signaling: BRI S/T protocol (two 64-kbps B-channels, one 16-kbps

D-channel) on a passive bus

Power: 48 VDC phantom power to telephone; 48 VDC over a separate pair (7-8) to a Direct Station Selector (DSS) console

N/A

N/A

T1 or PRI

LS and TTR

On-premises or offpremises single-line telephone service, use MLX IROB protectors, over 1000 feet (305 meters), use

OPRE.

Capacity: 8 T/R stations on two-way voice transmission path with support for telephones with message-waiting LEDs, 2 TTRs

Notice to telephone company: meets FCC Class C

Ringing current: 105-Vrms, 30-Hz sinusoidal ringing superimposed on -48 VDC: a ring generator must be installed in the power supply of each earner that has a 008 OPT module.

REN:

≤1.0 per port

Disconnect signal, 900 ms (T/R short for answering machines,

Group Ill Fax, etc.)

Switchhook flash detection: 3001200 ms

Loop resistance: serves 2-wire loops to 1300 ohms, including stations

Single-line telephone;

MERLIN Attendant;

Capacity: 12 T/R stations on 2-way voice transmission path with support for telephones with message-waiting LEDs, 2 TTRs

MERLIN MAIL Voice

Messaging System;

T/R adjunct (such as

Power. 21 VDC, 800-ohm battery source

Ringing current: 105-Vrms, 30-HZ sinusoidal ringing superimposed on -48 VDC; a ring generator must be installed in the power supply answering or fax module of each carrier that has a 012 module.

maachine): Analog data REN:

≤1.0 per port device (such as Disconnect signal: 900 ms (T/R short for answering machines, modem) Group Ill Fax, etc.)

Switchhook flash detection: 300-1200 ms

Loop range: 1000 feet (305 meters), in-building only

Power Failure Transfer

(PFT) telephone

Capacity: 24 lines/trunks for voice and analog data or 23 lines/trunks for voice and data with 1 channel used for signaling

Mode: multiplexes 24 or 23 lines/trunks into one facility and demultiplexes one facility into 23 or 24 lines/trunks speed: up to 64 kbps

Signaling: DS1 over 4-wire; T1 uses Robbed Bit Signaling (RBS) or

Common Channel Signaling (CCS): PRI has 23 B + D

Capacity: 4 lines/trunks, 4 TTRs, 1 PFT telephone

Signaling: loop-start



Hardware Components


Module

400EM

400 GS/LS/TTR

408 2

408 GS/LS

8 0 0 2

800 DID

800 GS/LS

Line/Trunk Type

Tie trunk

LS or GS and TTR

LS

LS or GS

408 GS/LS -MLX 3

LS or GS

LS

DID

LS or GS

Station Type

PFT telephone (GS button needed for PFT telephone)

Analog multiline telephone; CMS; PFT telephone

Specifications

Capacity: 4 tie trunks

Method of Completion: automatic or dial-repeating start; immediate-start, wink-start, or delay-dial-start

Signaling: E&M type 1S, type 1C, type 5

Capacity: 4 lines/trunks, 4 TTRs, 1 PFT telephone

Signaling: loop-start or GS, optioned per port

Analog multiline telephone; CMS; PFT telephone (GS button needed for PFT telephone)

MLX telephone; digital data device (such as

7500B Data Module)

PFT telephone

PFT telephone (GS button needed for PFT telephone)

Capacity: 4 lines/trunks, 8 stations, 1 PFT telephone

Station signaling: analog multiline telephone (40 kbps)

Line/trunk signaling: loop-start line/trunk; analog voice

Loop range: 1000 feet (305 meters), For In-Range Out-of-Building service, use analog IR0B protectors: over 1000 feet (305 meters), use OPRE.

Capacity: 4 lines/trunks, 8 stations, 1 PFT telephone

Station signaling: analog multiline telephone (40 kbps)

Line/trunk signaling: Ioop-start or ground-start line/trunk (optional per port); voice

Loop range: 1000 feet (305 meters). For In-Range Out-of-Building service, use analog IROB protectors; over 1000 feet (305 meters), use OPRE.

Capacity: 4 lines/trunks, 1 PFT telephone, 8 digital stations, each with 1 or 2 endpoints (each endpoint is assigned an individual extension number), including the following station types:

■ digital voice only

■ digital voice with Voice Announce to Busy feature

■ digital voice and digital data (via the 7500B Data Module)

■ digital voice and MFM

■ digital data only (7500B Data Module)

Signaling: BRI S/T protocol (two 64-kbps B-channels, one 16-kbps

D-channel) on a passive bus

Power: 48 VDC phantom power to telephone; 48 VDC over a separate pair (7-8) to a DSS console

Loop range: 1000 feet (305 meters) For In-Range Out-of-Building service, use MLX IROB protectors; over 1000 feet (305 meters), use

OPRE.

Capacity: 8 lines/trunks, 2 PFT telephones

Signaling: loop-start

Capacity. 8 lines/trunks, 2 TTRs

Protocol: incoming calls only, 2-way (one-pair) fixed impedance to

DID trunks; no outgoing calls

Signaling: loop-reverse battery; wink-start or immediate-start: accepts touch-tone dialing

Capacity: 8 lines/trunks, 2 PFT telephones

Signaling: loop-start or ground-start

Notes:

1.

2.

3

The system software recognizes the OPT moduleas a 012 module, Even though the OPT module only has 8 jacks, it uses 12 ports of capacity, thereby decreasing overall station capacity by four stations for every OPT module.

Although these MERLIN II modules are supported, the following are the recommended modules for the system. 400 GS/LS, 408 GS/LS, 800 GS/LS, 408 GSLS-MLX.

This module is not compatible with Releases 1.0 and 1.1: it applies to Release 2.0 and later.

2-8 Control Unit

Hardware Components

408

GS/LS-MLX Module

WARNING:

The 408 GS/LS-MLX module applies to Release 2.0 and later. It is not compatible with Releases 1.0 and 1.1.

The 408 GS/LS-MLX module (Release 2.0 only) is similar in concept to the 408

GS/LS module by providing four line and eight station ports; however, it provides MLX ports instead of analog ports. The MLX port operation is the same as that of the 008 MLX module, and the GS/LS port operation is the same as that of the 400 GS/LS module and the trunk portion of the 408 GS/LS module.

A PFT port is provided for the first CO line on the module. Touch-tone receivers are not provided on the module.

A customer can replace pairs of 400 GS/LS and 008 MLX modules with the 408

GS/LS-MLX module to obtain cost and slot savings. The 408 GS/LS-MLX module can be used in any of the 17 port board slots. All 17 slots can be simultaneously equipped with 408 GS/LS-MLX modules, but the system translates and uses only the number of ports allowed by your specific system configuration, such as number of endpoints and calls per hour.

Cover

The control unit is covered by a plastic cabinet for protection. The size of the

cover increases as expansion carriers are added to the control unit. Figure 2-5

shows how the control unit cover fits around the control unit carrier.


Hardware Components

Figure 2-5. Control Unit Cover

MERLIN II Release 3 Reusable Modules

The following modules used in a MERLIN II Release 3 system can be used in this system:

■

391A1 power supply

NOTE:

While the 391A power supply can be reused in the system, it does not supply as much power as the 391 Al power supply module and should be replaced if anything is added to the station side of the system. The

391 A also has less protection against power surges than the 391 A1.

2-10

Control Unit

Hardware Components

■ 800 line/trunk module

■

400 line/trunk module

■

400 E&M line/trunk module

■ 012 basic telephone module

■

008 analog station module

■ 408 analog line/trunk and station module

Table 2-2 shows the reusable MERLIN II modules and their apparatus codes.

Table 2-3 shows reusable MERLIN II hardware and associated apparatus codes

or PECs.

Table 2-2. Reusable MERLIN II Modules

Type

008

012

100D

Apparatus

Code

517A3

517B3

517A13

517B13

517C13

517D13

517E13

517A15

517B15

Comments

Fully compatible

Fully compatible

Compatible but does not support the downlink disconnect needed for voicemail; does not meet Megacom® transmission requirements

Compatible but does not support the downlink disconnect needed for voicemail; does not meet Megacom transmission requirements

Compatible but can be used for

Megacom only when the customer does not have to meet EIA transmission standards

Compatible but can be used for

Megacom only when the customer does not have to meet EIA transmission standards

Fully compatible

Supports only tie-trunk emulation

Fully compatible


Control Unit

2-11

Hardware Components


Type

400

400EM

408

8 0 0

Apparatus

Code

517A12

517B12

517A14

517A1

517B1

517C1

517A4

517B4

Comments

No lightning protection; 146A surge protector required

Fully compatible

Fully compatible

No lightning protection; 146A surge protector required

Fully compatible

Fully compatible

N

O lightning protection; 146A surge protector required

Fully compatible I

Table 2-3. Reusable MERLIN II Hardware

Type

Apparatus

Code or PEC

Power supply module 391A

Basic carrier

391 AA

391A1

403A

403C

403E

Comments

No surge protection; 147A protector recommended

For Canadian use only; no auxiliary power jack

Fully compatible

Compatible but must order system cover separately (part 16A); required spring clips for the system cover are provided with the upgrade package

For Canadian use only; must order system cover separately (part 16A); required spring clips for the system cover are provided with the upgrade package

Fully compatible


2-12 Control Unit

Hardware Components


Type

Expansion carrier

Apparatus

Code or

PEC

403B

403D

403F

Frequency generator 129B

(ring generator)

Auxiliary power 335A

Comments

Compatible but must order system cover separately (part 17A)

For Canadian use only; must order system cover separately (part 17A); required spring clips for the system cover are provided with the upgrade package

Fully compatible

Fully compatible

Music coupler

9024

61398

Compatible but can be used only when the unit loads do not exceed the 335A’s capacity; an Auxiliary Power Unit 9024 is recommended

Fully compatible

Fully compatible

Telephones and Consoles

Several different analog and single-line telephones can be used with the system; the only digital telephones that can be used with the system are the

MLX telephones,

NOTE:

An analog or digital multiline telephone located in a different building but within 1000 feet (305 meters) of the control unit requires an I ROB protector at each building entrance. If a single-line telephone is located in a different building from the control unit and is in excess of 1000 feet (305 meters), an

OPRE or a 008 OPT module must be used.

Telephones and Consoles 2-13

Hardware Components

MLX Telephones

The following are the four telephones in the MLX telephone line, all of which support the PRI services that can be used with the system:

■ MLX-20L telephone

■

MLX-28D telephone

■

MLX-10D telephone

■ MLX-10 telephone

Each of these telephones is available with factory-imprinted buttons in English,

French, or Spanish, and in black or white.

The following features are common to all MLX telephones:

Programmable line and feature buttons with two associated lights (red and green)

NOTE:

An MLX-20L telephone used as a Queued Call Console (QCC) has no programmable buttons.

Fixed-feature buttons (four of them have a red or a green LED: Feature,

HFAI, Mute, and Speaker)

Red message-waiting LED

Built-in speakerphone

Separate volume controls for speakerphone, handset, and ringer

A card tray under the telephone with frequently used features

Optional internal MFM to connect to T/R equipment and alerting devices

NOTE:

An MLX-20L telephone used as a QCC cannot have an MFM.

Two-position adjustable desk stand

Four-pair modular line cord

MLX telephones with display have the following two additional features:

■ LCD display

■

Display-associated buttons

A list of features specific to each telephone model in the MLX family follows.

2-14 Telephones and Consoles

Hardware Components

Model MLX-28D

The MLX-28D telephone provides the following features:

■

Can be used as a system operator Direct-Line Console (DLC)

■ 28 line buttons

■

Display (2 lines x 24 characters)

■

8 display-associated buttons

■ 8 dedicated feature buttons

■ Accommodates one or two DSSs

This telephone is not wall-mountable.

Figure 2-6. MLX-28D Telephone


Hardware Components

Model MLX-20L

The MLX-20L telephone provides the following features:

■ Can be used for system programming and as a DLC or a QCC system operator console

■

20 line buttons

■ Display (7 lines x 24 characters)

■ 14 display-associated buttons


■ Accommodates one or two DSSs

This telephone is not wall-mountable.

Figure 2-7. MLX-20L Telephone


Hardware Components

Model MLX-10D

The MLX-10D telephone provides the following features:

■

10 line buttons

■

Desktop or wall-mount

■

Display (2 lines x 24 characters)

■

8 display-associated buttons


Figure 2-8, MLX-10D Telephone


Hardware Components

Model MLX-10

The MLX-10 telephone provides the following features:

■

10 line buttons


■

Desktop or wall-mount

Figure 2-9. MLX-10 Telephone


Hardware Components

Direct Station Selector

The DSS is an optional adjunct that can be connected to an MLX-20L or an

MLX-28D telephone, It enhances the capabilities of both DLCs and QCCs and, when connected to the MLX-20L telephone, facilitates programming. The DSS is shown in Figure 2-10.

P a g e

Buttons

Message

S t a t u s

B u t t o n

Figure 2-10. Direct Station Selector

A DSS has 50 buttons with lights that can be programmed with the following:

■ Extension numbers

■ Line/trunk numbers

■ Pool dial-out codes

■ Calling group extension numbers

■ Paging group extension numbers

■ Park zone access codes

■ Automatic Route Selection (ARS) access codes


Hardware Components

■ Remote Access dial code

■ Listed Directory Number (the extension for the QCC queue)

These buttons are used by a system operator for one-touch dialing and call transfer. Ten additional buttons are located at the bottom of the DSS.

The 50 numbers programmed onto the DSS are considered a “page,” Each

DSS can have three pages of numbers, for a total of 150 numbers per DSS.

Two DSSs connected together increase this capability to 300 numbers. (If two

DSSs are connected, page one has the first set of 100 numbers, page two has the second set of 100, and page three has the third set.)

Three of the 10 lower buttons on the DSS are reserved as Page buttons to provide access to the different pages. The range of the numbers on the pages can be programmed to begin with any number up to 9950, but the default for the beginning Page numbers is 0, 50, and 100, respectively, for the three pages. A fourth button is reserved as a Message Status button, which, when pressed, changes the indication of the DSS button lights from telephone use to

Message Status mode. The other six buttons on each DSS are for future use.

See Telephone Power Units and Table 2-9 in this chapter for more information

on connecting DSSs and consoles.

NOTE:

DSSs are shipped without local auxiliary power supplies; if required, these must be ordered separately.


Hardware Components

Analog Multiline Telephones

In addition to the MLX telephones, the analog multiline telephones in Table 2-4.

can be connected to the system.

Table 2-4. Analog Multiline Telephones

5-button*

10-button*

34-button*

34-button Deluxe*

10-button HFAI*

BIS-10

BIS-22

BIS-22D

BIS-34

BIS-34D

Model

34-button BIS*

34-button BIS/DIS*

MLC-5 Cordless

MERLIN PFC Telephone †

Description

5-button telephone with membrane; no adjuncts supported

10-button telephone with membrane

34-button basic telephone with membrane

Deluxe 34-button telephone with membrane

10-button hands-free-answer telephone; no adjuncts

I supported

34-button telephone with built-in speakerphone

34-button telephone with 16-character display and built-in speakerphone


I 22-button telephone with built-in speakerphone




Cordless 5-button telephone (Limitations: Button assignments cannot be changed; cannot use

Conference feature)

Analog multiline phone, fax machine, and copier

* Vintage telephone; no longer available for sale or lease

† Requires two analog multiline ports.


Hardware Components

Single-Line

Telephones

The system supports the single-line analog telephones listed in Table 2-5.

NOTE:

PFT telephones must be selected according to trunk type. If rotary trunks are used, PFT telephones must be rotary telephones (500MM is recommended). If telephones are to be connected to ground-start trunks, a ground-start button (KS23566,L1 ) must be added to each PFT station.

Table 2-5. Single-Line Telephones

Model Description

2500MMGB

2554MMGJ

2500YMGK*

Basic desk telephone

Basic wall telephone

Basic desk telephone with message light and Recall button: Recall button is used instead of the switchhook for features that require a switchhook flash, suet as Transfer and Hold

Basic desk telephone used with 4A speakerphone 2500SM

2514 BMW

2526BMG

7101A*

7102A

Basic desk telephone with built-in headset jack

Outdoor telephone used with weatherproof enclosure

Basic desk telephone with Message light and Recall and Disconnect buttons.

No adjuncts supported.

Basic desk telephone with Message light lamp and Recall button, The 101 and

201 speakerphones and the 500 headsets are supported. Can be used for power-failure transfer (PFT) stations.

CS6402U01A* Basic desk telephone, Feature Phone Model 420. Has built-in speakerphone, memory, and redial.

2500MMGJ Basic desk telephone

2500MMGK

8 1 0 2 †

500MM

554BMPA

500SM

Basic desk telephone with Recall button; Recall button is used instead of the switchhook for features that require a switchhook flash, such as Transfer and

Hold.

Basic desk telephone with jack to support headset adapters and speakerphone adjuncts.

Basic desk telephone with a built-in speakerphone with volume control and

Mute button with LED indicator.

Basic telephone with the following limitation: equipped with rotary dials so no system features requiring * and # can be used. Telephones with neon Message lights are not supported.

* Vintage telephone; no longer available for sale or lease.

† Although the model 8102 can be connected to a speakerphone and the model8110 has a built-in speakerphone, neither can be used for Group Paging, which is not supported on single-line telephones. The Auto Answer function on the model 8110 must be disabled for operation with the system.


Hardware Components

Telephones and Adjuncts Not Supported

CAUTION:

The following telephones and adjuncts cannot be used with the system. Connecting them can damage the telephones, adjuncts, and system.

Table 2-6. Telephones and Adjuncts Not Supported

Model

510D Personal

Terminal

Notes

Uses Digital Communications Protocol (DCP)

DCP telephones 7400 telephones and adjuncts (asynchronous data units and multiple asynchronous data units) that use DCP and that are supported on MERLIN II

MET telephones Multibutton electronic telephones (MET) and adjuncts that are used with the

Dimension ® PBX and Horizon ® communications systems

Single-line telephone with neon Message light

Cannot support voltage required for neon light

Analog telephone adjuncts

Basic telephone modem interface (BTMI and BTMI-2); Off-premises extension

(OPU) unit; System 25 direct extension selector (DXS); DSS attached to a 34button deluxe membrane


Hardware Components

System Operator Consoles

System operator consoles are telephones that are programmed for call handling and other system operator duties. They can be used in two configurations—

QCC and DLC. QCCs are available only in Hybrid/PBX mode.

A system operating in Hybrid/PBX mode can include both QCCs and DLCs. The maximum numbers of both types of system operator positions is shown in Table

2-7.

Table 2-7. Maximum Number of System Operator Positions

Type of

Position Type Telephone

QCC

DLC

MLX-20L

MLX-20L

MLX-28D

DLC Analog multiline telephones

Maximum

Positions

4

8

8

NOTE:

No more than eight system operator positions of any combination (QCCs and DLCs) are allowed; when used in combination, no more than four can be QCCs.

Queued Call Consoles

QCCs are available only in Hybrid/PBX mode. In a QCC configuration, incoming calls are held in a queue and calls are directed to a QCC as a position becomes available. Only one call rings at a time.

QCCs must be connected to a digital station jack on a 008 MLX or 408 GS/LS-

MLX module (Release 2.0 only). The first QCC must be connected to the first station jack in the system.

The MLX-20L telephone is the only telephone that can be assigned as a QCC through system programming. A QCC system operator cannot use feature codes to activate features. The QCC system operator can only use the features that can be selected from the display or that are assigned as fixed features to buttons on the console.

The 7-line, 24-character display also provides the system operator with descriptive information about incoming and outgoing calls. This information includes extension numbers and any programmed labels (such as names), trunk identifiers, reasons for call return and redirection, and the number of unanswered calls waiting in the queue.

2-24 Telephones and Consoles

Hardware Components

The buttons on the QCC are factory-set with fixed features and cannot be programmed by the system operator or through centralized programming. The

QCC fixed-feature buttons areas follows:

Call: Five buttons used to answer incoming calls and make inside and outside calls.

Start:

Initiates the call-extending process by putting a caller on hold at the Source button and providing an internal dial tone to the system operator.

Source: Reconnects the system operator to the original caller while the call is in a split condition.

Release: Releases the system operator from a call and/or completes the call-extending process, making the system operator available for another call.

Destination: Reconnects the system operator to the destination while a call is in a split condition.

Cancel: Cancels call extending and reconnects the system operator with the caller (source).

Join: Connects the system operator with both the caller (source) and the person being called (destination) in a three-way conference. All three parties are connected on one Call button.

Headset Mute (Headset/Handset Mute): Activates or deactivates the headset or handset microphone.

Headset Status: Activates and deactivates the headset operation of the console.

Headset Auto Ans (Headset Auto Answer): Activates or deactivates the

Headset Auto Answer feature when headset operation is activated by pressing the Headset Status button,

Send/Remove Message: Turns on the telephone message LED to indicate a message waiting, and turns off the message LED when all system operator messages are delivered.

Position Busy: Temporarily takes the system operator console out of service.

Night Service: Activates or deactivates Night Service.

Alarm: Provides visible indication of a system alarm. When a system alarm has occurred, the red LED next to the button is on and the system operator can use the Inspect feature to determine the number of alarms present.

Pool Status: Provides the system operator with the status of all trunk pools (a maximum of 11). The information includes the number of trunks and the number of busy trunks in each pool.


Hardware Components

■ Forced Release: Disconnects the system operator from an active call and makes the system operator available to receive another call.

Each QCC can have one or two DSSs attached. The system operator can use the buttons during call handling, for example, to extend a call, make an inside call, park a call, or see the availability of an extension.

No more than four QCCs are allowed on a system. A maximum of two QCCs can be assigned on each 008 MLX or 408 GS/LS-MLX module (Release 2.0

only). No more than eight system operator positions of any combination (QCC and DLC) are allowed; when used in combination, no more than four can be

QCCs.

When a system has QCCs, the first MLX module used for QCCs must be installed in the control unit to the left of any other type of module with station jacks.

A maximum of two QCCs is allowed per 008 or 408 MLX module. A QCC can be connected only on the first and fifth station jack on each module.

The following options must be assigned to a QCC through system programming:

■ QCC operator receiving calls

■ QCC Queue Priority

■ Call Types

■ Elevate Priority

■ Hold Return

■ Automatic Hold or Automatic Release

■ Calls-In-Queue Alert

■ Queue Over Threshold

■ Extended Call Completion

■ Position Busy Backup

■ Return Ring Interval

■ Message Center Operation

Direct-Line Consoles

In a DLC configuration, lines/trunks are assigned to individual buttons and the console can have several calls ringing at the same time.

A DLC operates like other multiline telephones. In all three modes of operation

(Key, Hybrid/PBX, and Behind Switch), outside lines are assigned as personal lines to individual buttons on the console. The lines assigned on an individual

DLC can also be assigned to buttons on other consoles or other telephones.

Incoming calls can ring on any of the line buttons, and several calls can ring simultaneously. The system operator directs calls to other users via the

Transfer button.


Hardware Components

A multiline telephone assigned as a DLC through system programming can use both system operator features and telephone features available for non-operator multiline telephones to increase call-handling efficiency. The system operator features that can be assigned to buttons on the console are Alarm, Night

Service, Missed Reminder, and Send/Remove Message.

On a system with fewer than 29 lines, Alarm, Night Service, and Send/Remove

Message are factory-assigned to analog DLCs with 34 buttons or more. On a system with more than 29 lines, Alarm is replaced with line 30, Night Service is replaced with line 31, and Send/Remove Message is replaced with line 32.

Alarm, Night Service, and Send/Remove Message and the first 1 through 18 lines are factory-assigned on an MLX-28D telephone used as a DLC, regardless of the number of lines/trunks connected to the system.

The following telephones can be used as DLCs:

■ Digital DLC

— MLX-20L telephone

— MLX-28D telephone

■ Analog DLC

— MERLIN II System Display Console with built-in DSS

— BIS-34D telephone

— BIS-34 telephone

— BIS-22D telephone

One or two DSS adjuncts can be added to the MLX-20L or MLX-28D telephone to provide 50x3 or 10OX3 additional extension buttons. The DSS cannot be attached to an analog DLC; however, the MERLIN II System Display Console provides a built-in DSS.

Analog DLCs are connected either to an analog station jack on a 008 or a 408 analog multiline telephone module, or to a digital station jack on a 008 MLX or

408 GS/LS-MLX module (Release 2.0 only).

When both DLCs and QCCs are assigned in the Hybrid/PBX mode, the maximum combined number of system operator positions is eight; no more than four can be QCCs. A maximum of two DLCs can be assigned per MLX or analog module.

An MLX-20 telephone used as a DLC can be used for system programming by connecting it to the first or fifth station jack on the first MLX module and designating the station jack for system programming.

Only multiline telephones connected to the first and fifth station jacks on digital or analog modules can be assigned as DLCs. This includes DLCs assigned as calling group supervisors and Call Management System (CMS) supervision.


Hardware Components

Adapters and Adjuncts

This section describes auxiliary hardware for the communications system. It includes adapters, adjuncts, and other accessories. The following table provides information about the hardware components that can be used with the system.

Table 2-8. Adjunct Summary

Equipment

Alerts (AC)’

Alerts (DC)

Specifications AT&T products

Any audible or visual alert that operates External on 20-30 Hz ringing signals.

Ringer—Loud

External Ringer Associated with a specific station

(supplemental alert) or works on a programmed trunk port (external alert).

Any audible or visual alert that operates Alert bell on 48-WC ringing signals.

Alert horn

Associated with a specific station

(supplemental alert) or works on a

Alert strobe

Alert chime programmed trunk port (external alert) Alert deluxe horn

Alert switch

Answer/Record Industry-standard machine.

machine’

Low ringer equivalence (less than 0.15

or

≤

1.0 total REN for T/R port)

Model 1300 answering machine

Ability to recognize 600-ms disconnect Model 1531 signal or other means of automatic

Remote Answering disconnect (such as voice reset

System telephone disconnect timer, fixed recording time).

Cordless

Telephone’

Must have touch-tone dialing capability 5320 Cordless when connected via MFM; rotary or Telephone touch-tone dialing can be used on T/R 5200 Cordless port.

Telephone

Single line 5500 Cordless

Telephone

Credit Card

Verification

Terminal’

Dial

Dictation’ 1 2

Must have touch-tone dialing capability when connected via MFM: rotary or touch-tone dialing can be used on T/R port.

A device that requires contact closure can be used on LS/GS line jack only with UPAM.

Interface

LS or

GS/LS T/R

012 or

008 OPT MLX

Trunk Station

Jack Jack

Station

Jack

✓ ✓

GPA SAA

Analog Analog

Station Station

Jack Jack

✓ ✓

✓ 2

✓

✓

✓

✓

✓

✓

✓

✓ 7

✓ 8

✓

✓

✓

✓

✓


2-28 Adapters and Adjuncts

Hardware Components


Equipment

Type Specifications AT&T Products

Direct Station A maximum of 2 DSSs can be

Selector connected to an operator console.

(DSS)

A 329A power unit must be added to an operator console having 2 DSSs.

Connects to DSS jack on operator console,

Fax 1

Must have touch-tone dialing capability AT&T 3410D when connected via MFM; rotary or AT&T 3500D touch-tone dialing can be used on T/R AT&T 35100 port.

Industry-standard analog interface

AT&T 3520D

AT&T 3530D

AT&T Fax 4515D

AT&T Fax 4525D

AT&T Fax 9015

AT&T Fax 9020

AT&T Fax 9022

AT&T Fax 9025FX

AT&T Fax 9035FX

Group

Calling Delay

Industry-standard announcement device, Must provide automatic

Announcement’ disconnect.

Each calling group can have its own announcement (maximum 32).

A device can provide delay announcement for more than one group.

Hands-Free

Unit

Headset

For use with analog multiline telephones

Connects directly to telephone.

For use with MLX or analog multiline telephones,

Headset

Adapter

Loudspeaker

Paging

Model 1330

Answering

Machine

DA-5 Digital Voice

Announcer

S202A

StarSet ®

Mirage ®

Supra ®

Supra NC ®

Need to program Auto Answer Ail button for use with 502B, 502C,

502A

502B

Connects directly to telephone OTHER 502C jack

External paging system using DTMF signaling connected to LS or GS line jack. CPE paging systems require an interface unit; if CPE has 2-wire input, the BOGEN UPAM-K (58500) can be used

PagePac 20

PagePac 20 with

Zonemate 9 9

PagePac 20 with

Zonemate 39 9

PagePac 6 10

PagePac 6 Plus

Interface

LS or

GS/LS T/R GPA SAA

012 or

008 OPT MLX Analog Analog

Trunk Station station Station Station

Jack Jack Jack Jack Jack

✓ 9

✓

✓

✓ 8

✓ 8

✓


Adapters and Adjuncts 2-29

Hardware Components

Table 2-8. – Continued

Equipment

Type

Message

W a i t i n g

Indicator

Modem

Specifications

For single-line telephones

Connects directly to telephone

AT&T {roducts

Z34A

If the modem supports touch-tone 4000 Modem

Music-on-

H o l d 4 the keyboard can be used for dailing.

If the modem does not support touchtone dialing, an associated basic

(single-line) telephone can be used for

4024 Modem

2296A Modem

2296 Modem d i a l i n g .

Magic On Hold Any FCC-registered 8-ohm music source or recorded announcement device.

Speakerphone Connect directly to telephone.

For signle-line telephones only.

4 A 6

203A

SMDR Printer Connects to upper RS-232-C jack on proccessor module

Must be located within 50 feet

(15 meters) of contro unit or use ADU to extend distance.

475 Printer

476 Printer

572 Printer

Interface

LS or

GS/LS T/R MFM GPA SAA

Trunk

J a c k

012 or

008 OPT

Station

J a c k

M L X

Staion

Jack

Analog

Staion

J a c k

Analog

Station

J a c k

✓

✓

3

5

✓ ✓ ✓

Notes.

1.

2.

3.

4.

5.

6.

7

8.

9.

10

Cannot be connected to a QCC

Requires UPAM to provide 48 VDC.

For 2224G Modem only.

If you use equipment that rebroadcasts music or other copyrighted materials, you maybe required to obtain a copyright license from and pay license fees to a third party such as American Society of

Composers, Artists, and Producers (ASCAP) or Broadcast M

USIC

Incorporated (BMI ). Or you can purchase a Magic On Hold system, which does not require you to obtain such a license, from AT&T or an authorized dealer

Music Coupler required

Requires 2500SM telephone.

Device originates and receives calls independently of associated telephone when used with an MFM.

When a GPA is used, calls are dialed and received via the analog telephone

Device originates and receives calls independently of associated telephone when used with an MFM.

Bi-directional paging is supported: only one line jack is needed for multizone paging.

Loop-start adapter (53518) is required when connected to Ioop-start line jack.

2-30 Adapters and Adjuncts

Hardware Components

Adapters

Adapters connect adjuncts to the system and stations. They provide access for both voice and data signals.

System Adapters

Three adapters connect directly to the control unit: the channel service unit

(CSU), the Loop-Start Trunk Adapter, and the Universal Paging Access Module

(UPAM).

Channel Service Unit

The CSU is the interface between the 100D module and the DS1 facility provided by the telephone company. This facility contains 24 channels on one

4-pair wire that connects to the back of the CSU. The CSU then connects to the modular jack on the 100D module.

There is an Extended Super Frame (ESF) T1 CSU and a 551 T1 L1 CSU. The

ESF T1 CSU is recommended for this system because it allows the unit to be maintained without interrupting service and provides diagnostic and testing capabilities, and it is the only CSU that provides bipolar 8 zero code substitution

(B8ZS) line coding. The lower-cost 551 T1 L1 CSU performs most of the functions of the ESF T1 CSU but does not provide the B8ZS line coding required for 64-kbps data and for maintenance features. The 551 T1 L1 CSU does not provide diagnostic and testing capabilities; also, it is not recommended for

Video Conferencing applications.

Figures 2-11 and 2-12 detail the CSU connections.


Hardware Components

Control Unit

Maximum distance 440 ft

D8W Cord

RJ45 (Female) to

1 5 p i n

D - S u b m i n i a t u r e

( M a l e ) t

Custom

Adpater

551 ST CSU REAR to CO o r

4 E S S

Figure 2-11. 551 T1 L1 Channel Service Unit Connections


Hardware Components

DTE LOOPBACK

FRONT PANEL

S E N D C O D E

LOCAL

POWER

LOOPED

L O W

DENS

FRAME LOSS

DIAGNOSTIC

INTERFACE

ALARM

N E A R L O O P E D

BPV

CR C ERROR

FRAME LOSS

PULSES

CRITICAL POWER

FAR END LOOPED

D C

F U S E

D C

P O W E R

TERMINAL

B L O C K

A C

FUSE

A C

P O W E R

CONNECTION

WIRE WRAP CONNECTOR

REAR PANEL

Figure 2-12. ESF T1 Channel Service Unit Connections

Universal Paging Access Module or Loop-Start Trunk Adapter

The UPAM and the Loop-Start Trunk Adapter connect paging equipment to a loop-start or ground-start jack on the control unit. A Loop-Start Trunk Adapter should be used only with a PagePac paging system; a UPAM should be used with a customer-provided paging system. A Loop-Start Trunk Adapter must be used with a PagePac paging system and no Zonemate equipment, An acknowledgement tone is not provided when an UPAM is used.

The UPAM is ordered as a D-kit, The kit contains a power source, a microphone matching transformer, and an access module.

Telephone Adapters

The following adapters are used to connect various adjuncts to the telephones.


Hardware Components

Multi-Function Module

The Multi-Function Module (MFM) is an optional adapter for connecting tip/ring

(T/R) or supplemental alert adjuncts to any MLX telephone. The MFM is a circuit board that mounts inside the telephone. Adjuncts plug into a modular jack on the MFM. The MFM is the only T/R adapter used with MLX telephones.

NOTE:

An MFM cannot be used in an MLX-20L telephone serving as a QCC.

Multi-Function Module

Figure 2-13. Multi-Function Module

WARNING:

The MFM can be installed or repaired only by a qualified technician or an authorized dealer representative. To eliminate the risk of electrical shock, the MLX telephone should not be disassembled.

T/R adjuncts operate independently of the MLX telephone. If the telephone is in use, voice or data calls can still be sent and received by the adjunct. The MFM allows the use of the following T/R adjuncts:


Hardware Components

■ Answering machines

■ Fax machines

■ Modems

■ Credit card verification terminals

■ Cordless telephones

■ 2500-type (basic touch-tone) telephones

Supplemental alerts such as bells, chimes, horns, and strobes notify people in noisy areas of incoming calls.

The MFM is shipped with a KS29911 , L2 or L2 power unit that supports one MFM and one DSS. When two DSSs are connected to a telephone, a 329A wall power unit is used instead of the KS22911. With either type of power unit, the total cord length can be no more than 50 feet (15 meters) from the telephone.

The MFM supports only touch-tone dialing and does not detect pulse-dialing.

Also, the MFM does not support features activated by a switchhook flash, so no transfer or conference calls can be made from a station attached to an MFM.

NOTE:

The MFM uses one of the two B-channels when it is active. This means that when an accessory, such as a fax machine, and the MLX telephone are in use at the same time, Voice Announce to Busy and speakerphone paging cannot be used, (When Voice Announce to Busy is being used, a person calling an MFM extension gets a busy signal; a person attempting to call out from a station attached to an MFM will not get a dial tone.)

Two jumper blocks on the MFM configure it for either tip/ring (T/R) or supplemental alert operation. These are preset at the factory for T/R operation.

NOTE:

Only a qualified service technician or an authorized dealer representative should change the jumper settings.

In the T/R mode, the MFM can connect to 20-Hz AC external alerting devices such as a loud external ringer and El CM-type ringer. If several devices are connected to the MFM, only one device can be off-hook at a time and the total ringer equivalent number (REN) < = 2.0.

General Purpose Adapter

A General Purpose Adapter (GPA) permits the attachment of a T/R device such as a single-line telephone, modem, or answering machine to an analog multiline telephone. The device must be touch-tone, not rotary, and calls must be originated on the analog multiline telephone since the GPA has no pulse or touch-tone detectors.


Hardware Components

One end of a 4-pair cord plugs into the V.T. jack on the back of the GPA, and the other end plugs into the OTHER jack on the underside of the telephone.

The 1-or 2-pair cord from the T/R device plugs into the TEL. EQUIP. jack on the GPA (see Figure 2-14).

Control Unit

D8W-87

D4BU

TIP/Ring Device

Genreal Purpose

Adapter

(REAR)

D8AC

L i n e

A n a l o g

Multiline

Telephone

( R E A R )

Other

Figure 2-14. GPA Connections

Sliding the switch on the back of the GPA to the proper setting provides the

GPA service required.

■ Basic. This setting is used to dial and answer calls on an analog multiline telephone or to attach a T/R device such as a single-line telephone or a fax machine. Incoming calls ring only on the analog multiline telephone.


Hardware Components

■ Join. This setting is used to add a recording device or a single-line telephone to a call that is in progress on the analog multiline telephone.

You cannot originate or answer calls on this setting,

■ Automatic. This setting is used in two ways:

— With a device, such as an answering machine or a modem, to answer calls. An Auto Answer All button is needed so that calls can be answered automatically.

— To make and receive calls on the telephone while a computer or modem attached to the GPA is being used. This is called the

Simultaneous Voice and Data feature.

7500B Data Module

The Integrated Services Digital Network (ISDN) 7500B Data Module connects a data terminal to the system on a 008 MLX or 408 GS/LS-MLX module (Release

2.0 only) so that a user can make and receive calls at a digital data station.

Instead of converting digital data signals to analog signals as a modem does, the data module maintains a digital data format that allows transmission to another digital station or over the PRI telephone network, On a data terminal, the keyboard is used to dial the number.

NOTE:

A data module cannot be used with a QCC.

The 7500B Data Module provides a RS-232 interface for asynchronous data terminal equipment operating at speeds of up to 19.2 kbps. The data module also provides a CCITT V.35 interface for synchronous data terminal equipment operating at speeds of up to 64 kbps. (Optional enhancement boards must be ordered separately. )

The data module can be set up to handle a variety of data communications equipment (DCE) and is the only digital adapter approved for use with the system.

The data module’s front panel has the following features (see Figure 2-15):

■ Power/Test LED—lights when power is supplied; flashes when tests are performed.

■ Data LED—flashes to indicate an incoming data call and lights when a call is in progress; flashes when tests are performed.

■ Display—displays status information and option settings.

■ Next, Back, and Enter buttons—used to operate the data module and to adjust the screen’s contrast.


Hardware Components

DCE/DTE Flip Board

Figure 2-15. 7500B Data Module Front Panel

The data module’s back panel has the following features (see Figure 2-1 6):

Phone jack-connects an MLX telephone to the data module.

Line jack—connects the data module to an MLX system module.

Power connector—connects the data module to the DC power supply, which connects to an AC outlet.

Port l—connects the 7500B to a data terminal, computer, or modem.

Port 2—when an enhancement board is installed for synchronous operation, Port 2 connects a second data terminal, an automatic calling device (with a RS-366 interface), or a data terminal with a V.35 interface.

NOTE:

A modem can provide an analog data interface from a MLX telephone has an MFM installed. When an MLX telephone has an MFM, the data module cannot be installed on the same line.

that


Hardware Components

Female 25-pin EIA

Figure 2-16. 7500B Data Module Back Panel

When the 7500B Data Module is used with an MLX telephone, one end of the

D8W cord is plugged into the PHONE jack on the data module, and the other end of the cord is plugged into the LINE jack on the MLX telephone. The maximum cord length from the data module to the telephone cannot exceed 80 feet (24 meters). The MLX telephone cannot be used to dial data calls, and the data terminal equipment connected to the data module cannot be used to dial voice calls. Each device operates independently, and features are assigned to each device independently.

NOTE:

Do not connect two 7500B Data Modules on one line.

The data module can be configured as a stand-alone by ordering a

WP90110,L1 power unit The data module can also be configured in a multiplemount arrangement by ordering a Z77A data mounting, which provides a common power supply for up to eight data modules. Both the power unit and the data mounting require a 115-VAC power outlet. Neither the power unit nor the data mounting is provided with the data module; both must be ordered separately.

The 7500B Data Module does not have the internal 100-ohm line termination that is provided with MLX telephones. Therefore, when it is used without an

MLX telephone, a 100-ohm 440A4 terminating resistor adapter must be installed on the line near the data module.

To provide synchronous operation at speeds up to 64 kbps, the following optional circuit boards must be ordered:


Hardware Components

■ Multipurpose Enhancement Board. Provides an RS-366 Automatic

Calling Unit (ACU) interface and converts the RS-232 interface on the main circuit board from asynchronous to synchronous. A V.35 adapter cable must be ordered separately to operate at the lower data rates and also at data rates of up to 56 and 64 kbps. Without the adapter cable, data rates are limited to 1200, 2400, 4800, 9600, and 19,200 bps.

■ High-Speed Synchronous Interface Enhancement Board. Provides a

V.35 interface at synchronous data rates of 48, 56, or 64 kbps. A V.35

adapter cable that converts the 25-pin male connector on the board to the industry-standard 34-pin V.35 interface is included.

Data Module Features

The 7500B Data Module offers the following features:

Asynchronous Features

RS-232 interface

Asynchronous full-duplex operation

Selected data rates of 300, 1200, 2400, 4800, 9600, and 19,200 bps

Data options set via the data terminal attached to the RS-232 interface

Ability to change options without dropping a data call

Autobaud (also called data metering or speed matching)— the ability to adjust the speed of transmission to match the speed of the data terminal being called

Auto-adjust—the ability to adjust to the speed and parity of the data terminal being used

Call setup (dialing) from the keyboard of an ASCII data terminal by using the local command (CMD) mode or AT mode

Automatic or manual answering of incoming data calls

Synchronous Features with Multipurpose Enhancement Board

■ RS-232 interface

■ Half- or full-duplex operation using the RS-232 interface at data rates of

1200, 2400, 4800, 9600, and 19,200 using data transport Mode 2

■ Half- or full-duplex operation at 56 kbps via the V.35 interface adapter cable

■ Full-duplex operation at 64 kbps via the V.35 interface adapter cable

■ Automatic answering of incoming data calls


Hardware Components

■ Ability to make outgoing data calls manually and select userprogrammable telephone numbers from the data module display on the front panel

■ RS-366 interface to an ACU

Synchronous Features with High-Speed Synchronous Enhancement

Board

■ V.35 interface (the adapter cable is provided when the board is ordered by using PEC 21624)

■ Full-duplex operation at 48, 56, and 64 kbps

■ Half-duplex operation at 56 kbps only

■ Automatic answering of incoming data calls

■ Ability to make data calls manually and select user-programmable telephone numbers from the data module display on the front panel

Modems

A modem is used at an analog data station to make and answer data calls. It converts the digital signals of the data terminal into analog signals for transmission over standard telephone lines. It also converts incoming analog signals to digital signals for acceptance by the data terminal.

Most types of modems can be connected to the system unless the modem is being used in a modem pool. The recommended models are as follows:

■ Modem Model 4000

■ Modem Model 2224G


■ Modem Model 2296A


If a modem is used with an MLX telephone, an MFM must be installed in the telephone to provide a tip/ring interface for the modem. The modem is connected directly to the MFM. If the modem is used with an analog multiline telephone, a GPA is required to provide a tip/ring interface for the modem.

When a modem is connected to an MLX telephone using an MFM, data calls are dialed using the data terminal keyboard and voice calls are dialed using the telephone dialpad. The MLX telephone cannot be used to dial data calls, and the data terminal keyboard cannot be used to dial voice calls. Each device operates independently, and features are assigned to each device independently,


Hardware Components

When a modem is connected to an analog multiline telephone using a GPA, data calls and voice calls are dialed by using the telephone dialpad. The modem and telephone do not operate independently; features assigned to the telephone also apply to the analog data station (modem and associated data terminal).

Modem Features

The modem used in an analog data station (and not in a modem pool) provides the following features:

Dialing or ending asynchronous data calls from the keyboard when connected using a basic telephone station jack on a 012 module or when connected to an MLX telephone using an MFM

Autobaud (also called data metering or speed matching)— the ability to adjust the speed of transmission to match the speed of the data terminal being called

Automatic or manual answering of incoming data calls

Self-test and maintenance procedures

Ability to set data options for the call on the keyboard and, if necessary, change the options without dropping the call

Supplemental Alert Adapter

A supplemental alert adapter (SAA) allows the connection of an alerting device

such as a bell or chime to an analog multiline telephone (see Figure 2-17).

These alerts notify people working in noisy areas of incoming calls.


Hardware Components

SUPPLEMENTAL ALERT ADAPTER

Figure 2-17, SAA Connections

The telephone cord plugs into the jack labeled TELEPHONE; the cord from the external alert device plugs into the jack labeled ALERTER; and the cord from the station jack plugs into the jack labeled CONTROL UNIT.

Adjuncts

An adjunct is an auxiliary piece of equipment, connected to the system or to telephones with an adapter, for example, a fax machine or an answering machine.


Hardware Components

System Adjuncts

System adjuncts are auxiliary pieces of equipment that connect directly to the control unit.

Station Message Detail Recording Printer

A Station Message Detail Recording (SMDR) printer can be connected to the control unit at the SMDR jack on the processor. The SMDR printer must be located within 50 feet (15 meters) of the control unit; otherwise, an asynchronous data unit (ADU) must be used to extend the distance.

The SMDR feature is used to capture detailed usage information on incoming and outgoing voice and data calls. Two SMDR report formats are available: the factory-set basic format or the PRI format. The PRI format is used when the user subscribes to the AT&T INFO2 automatic number identification (ANI). When the

PRI format is selected during system programming, the number identification information is printed in the Called Number field of the call report. The remainder of the fields are identical to the basic format.

This information is sent to the SMDR printer. An SMDR record consists of the following fields:

■ Call Type (Basic or PRI)

■ D a t e

■ T i m e

■ Called Number

■ Dur (Duration)

■ Line (Facility Number)

■ STN (Station Extension)

■ Account (Account Code)

System Programming and Maintenance PC

A PC with DOS version 3.3 or higher and SPM software can be used for the programming and maintenance of the system. The PC is connected to the lower jack on the processor (the system programming/maintenance jack). See

the Applications chapter for additional information.

Loudspeaker Paging Systems

Loudspeaker paging systems use a GS/LS line port. The port should be programmed for loop-start operation and programmed as a paging port. Up to three ports can be programmed as paging ports. When connecting a customer-owned paging system, a UPAM should be used. See the discussion of UPAM earlier in this chapter.


Hardware Components

NOTE:

A ground-start/loop-start (GS/LS) line port should be programmed for loopstart operation for paging equipment. If the loop-start port is programmed for paging, it cannot be used for outside calls unless a PagePac Port Saver is used.

The PagePac 20 with Zonemate 9 or Zonemate 39 is an external paging system using dual-tone multifrequency (DTMF) signaling that can be connected to a line/trunk port programmed for paging operation. Bidirectional paging or

“talkback” is available. For paging systems with multiple zones, only one loopstart or ground-start line/trunk is required.

This system is compatible with the PagePac 20 set to loop-start or ground-start operation, and a PagePac 6 Port Saver. The Port Saver allows the paging unit to be connected to the same loop-start line/trunk used for calls. Calls can be made and received when the paging unit is not active. The Port Saver is not compatible with ground-start lines/trunks.

Music-on-Hold and the Music Coupler

Background music can be provided for outside callers on hold or for selected areas of a building over the loudspeaker system.

NOTE:

If equipment is used that rebroadcasts music or other copyrighted materials, the user may be required to obtain a copyright license from, and pay license fees to, a third party such as the American Society of

Composers, Artists, and Producers (ASCAP) or Broadcast Music

Incorporated (BMI). Or a Magic On Hold system, which does not require the user to obtain such a license, can be purchased from an AT&T or authorized dealer.

Music can be heard by outside callers on hold by connecting a music source or

Magic On Hold unit to a music coupler. The music coupler must connect to a loop-start port that has been programmed for Music-on-Hold operation and must be used with an 8-ohm music source. The loop-start port cannot be assigned to a trunk pool, cannot appear as a programmed line on system telephones, cannot be connected to a loudspeaker paging system, and cannot be on a trunk port assigned to the QCC queue or to calling groups.

When a call is transferred, it is automatically connected to Music-on-Hold if the system provides this feature.

The PagePac 20 also provides a music source for paging and Music-on-Hold without a music coupler.

Dial

Dictation

Dial dictation through a customer-provided dictation unit can be used as either a system or station adjunct. Some dictation units connect directly to the control unit via a T/R jack on the 012 module or 008 OPT module, or to a telephone using an MFM or a GPA. Other dictation units connect to an UPAM that is


Hardware Components connected to a loop-start port programmed for dial dictation (similar to loudspeaker paging). The UPAM does not support dial dictation equipment requiring contact closure.

Fax Machine

A fax machine can be connected to any T/R jack on the control unit or to a

MFM. Using a fax machine with a GPA is not recommended because the fax machine cannot auto dial through the GPA. Instructions are packed with the unit.

A fax machine originates and receives fax calls independently of any associated telephone. Calls are dialed with the fax machine’s dialpad or from an associated single-line telephone.

If the system doesn’t have direct inward dialing (DID) trunks, fax stations should be programmed to personal lines. When the system has DID service, incoming calls can be directed automatically to individual fax stations or to machines in calling groups.

NOTE:

A fax machine can also be considered a telephone adjunct, when used with an MFM.

Delay

Announcements

A delay announcement recording is used when there is a delay before an incoming call is answered (such as with calling groups). Announcements can be made with industry-standard announcement devices connected to a station port on a 012 module or a 008 OPT module, or by an interface such as an MFM.

Telephone Adjuncts

Telephone adjuncts connect to a telephone directly or through an adapter.

Headsets and Headset Amplifiers

For hands-free operation of the telephone, four headsets are available:

■ StarSet Headset— a monaural headset worn without a headband. It uses a one-size-fits-all soft, pliable ear tip that provides high-quality sound yet allows you to hear other conversations or instructions in the workplace.

■ Mirage Headset— a small, almost unnoticeable monaural headset that uses a disk-shaped receiver. It can be worn on either ear, instead of a headband or ear tip.

2-46 Adapters and Adjuncts

Hardware Components

■ Supra Headset— a monaural headset with an adjustable headband. It offers a soft ear, comfortable cushion that reduces surrounding noise, making it easier to understand the caller.

■ Supra NC Headset— a binaural headset with adjustable headband and soft ear cushion for working in noisy environments. Its noise-canceling microphone and voice expansion technology reduces up to 75% of the surrounding noise.

Each headset is light, comfortable and uses a transparent voice tube to eliminate the cumbersome large microphone. Each model comes with a 10-foot

(305-cm) coiled cord and a Quick Disconnect latch.

These headsets work with any telephone connected to the communications system when combined with one of the amplifiers described in the following section.

See Figures 2-18 and 2-19.

The following amplifiers work with the supported headsets:

■ Modular Amplifier— connects the StarSet, Mirage, Supra, and Supra NC headsets to virtually any telephone equipped with a modular handset. It is installed without special tools. You can adjust the incoming volume, switch between the headset and handset as needed, and temporarily mute the line.

■ Plug Prong Amplifier (for non-MLX telephones)—connects the StarSet,

Mirage, Supra, and Supra NC headsets to operator consoles, telephones equipped with a headset adapter, and many automatic call distributors.

It provides switch-hook control for answering calls by pressing a button,

You can also adjust the incoming volume.

Both amplifiers provide safety and comfort features that protect you from unexpected loud signals and provide clear sound. The amplifier stays with the telephone and works interchangeably with individual headsets.


Hardware Components

Supr® Headset Supra NC® Headset

StarSet® Headset

Figure 2-18. Headsets

Mirage® Headset


Hardware Components

Figure 2-19. Analog Multiline Telephone Headset

Speakerphones

Separate speakerphones can be connected to single-line sets

Model S202A Hands-Free Unit (HFU) for analog multiline telephones allows calls to be made and received without using the handset. If an Auto Answer

Intercom button is programmed and activated on the telephone, the HFU goes on automatically when a voice-announced call is received.

The S203A speakerphone allows voice transfer without using the handset. It is compatible with any 2500 or 500 phone and requires its own local power.

Newer units may be shipped with a transformer, but older units require a

KS21239L4 transformer and 248B adapter to be ordered separately.

The 4A speakerphone consists of a 108AA loudspeaker set, a 680-type transmitter, an 85B1 power supply, and a 223C connecting block. The 4A speakerphone must be used with a 2500SM telephone.


Hardware Components

NOTE:

The quality of speakerphone transmission can be affected by equipment in the area of the microphone.

Specialty Handsets

Handsets for users who are hard of hearing (model K6S) are available for use with MLX telephones.

Message

Waiting Indicator

The Z34A message-waiting indicator can be connected to single-line sets that do not have a Message LED.

Additional Telephone Adjuncts

The following adjuncts also can be connected to telephones:

■ Answering machines

■ Credit card verification terminals

■ PCs (connected through modems, 75006 Data Modules, or directly with a built-in modem)

Adapters and Adjuncts Not Supported

CAUTION:

The following analog telephone adjuncts and adapters cannot be used with the system and in some cases will damage the device or the system if connected:

■ Basic Telephone and Modem Interface (BTMI)

■ Basic Telephone and Modem interface-2 (BTMI-2)

■ ATR Interface (ATRI)

■ MTR Interface (MTRI)

■ Off-Premises Extension Unit (OPU)

■ System 25 Direct Extension Selector (DXS)


Hardware Components

Power-Related Hardware

Power-related hardware can be added to the system to provide more power and added protection from power surges. Other accessories apply to specific conditions.

NOTE:

In most cases additional power surge protection is not needed.

Power Accessories

In a power failure, battery backup units can keep the system running for several hours.

When adjuncts and adapters are connected to telephones, the power requirements of the telephones and the communications system increase.

Adding a power accessory to an individual telephone or to the system accommodates these additional needs.

Battery Backup Power

Battery backup for power to the system can be provided by an optional 500 VA uninterruptible power supply (UPS) and reserve UPS units, The basic UPS provides power for 15 minutes. Reserve UPS units can be added to the basic

UPS. Each reserve unit added extends backup power for an additional hour.

The holdover back-up durations for normal system operation of one full carrier at a maximum system load are as follows:

15 minutes basic 500 VA UPS

1 hour one 500 VA reserve cabinet for each UPS

2 hours

4 hours two 500 VA reserve cabinets for each UPS four 500 VA reserve cabinets for each UPS

Telephone Power Units

Connected between the telephone and the wall jack, the KS22911, L2 and 329A power units provide additional power to individual telephones that have adjuncts, adapters, and/or two DSS

S attached, or to telephones far from the control unit. Adding local power to a few telephones can reduce the system load.

Table 2-9 shows local auxiliary power requirements. The KS22911, L2 power

supply must be connected to an unstitched 117-VAC outlet.

Power-Related Hardware 2-51

Hardware Components

Table 2-9. Local Auxiliary Power Requirements

Number of MLX

Telephones

1

1

1

3 or more in one carrier

KS22911,L2

Number of DSSs Number of MFMs

2

—

1 1

2

1 per telephone

1

—

Number of

Number of 329A

Power Supplies

1

1

—

1 for each MI-X telephone after the first 2

Power Supplies

—

—

1

—

CAUTION:

When additional control unit carrier power is required and the system is backed up by an uninterruptible power supply (UPS), the carrier’s

auxiliary power unit (Supplemental Power Unit 9040-2) should also be connected to the UPS.

The total length of wire between the KS22911, L2 or 329A power supply and the MLX telephone can be no more than 50 feet (15 meters).

Do not replace the 2-foot (61-cm) D8AC cord (packaged with the DSS)

with a longer cord. Improper operation may result.

A KS22911 kit for MLX telephones comes complete with a D6AP cord, the

KS22911, L2 power unit, and a 400B or 400B2 adapter. For analog multiline telephones, the KS22911 kit includes the KS22911, L2 power unit, a D6AP cord, and a Z400F adapter.

The 329A power unit does not come in a kit, so the D6AP cord and the 400B or

400B2 adapter must be ordered separately.

The MFM comes complete with a D6AP cord, the KS22911 power unit, and a

400B or 400B2 adapter. DSS

S are shipped without power units. Therefore, when DSSs require local power, the KS22911, L2 or 329A power unit, D6AP cord, and 400B or 400B2 adapter must be ordered separately.

NOTE:

Telephone operation without adjuncts is guaranteed for a wiring run up to

1000 cable feet (305 meters) from the control unit.

Auxiliary Power Units

The power supply provides 54 unit loads to each carrier. If the unit load requirement for a carrier exceeds 54 unit loads, an auxiliary power unit is needed to allow that carrier to support an additional 27 unit loads.

2-52 Power-Related Hardware

Hardware Components

CAUTION:

Running the system with more than 54 unit loads per carrier may not appear to do harm. However, this can cause the system to malfunction, thereby creating “no trouble found” situations, such as malfunctioning LEDs on multiline telephones, or power unit failure.

Any station connected to the modules in the last two slots receives power from the auxiliary power unit instead of from the power supply.

If an auxiliary power unit is required, complete instructions are provided in

Installation.

To determine the number of unit loads for each power supply module on each

carrier, see Unit Loads in Chapter 1.

NOTE:

Only one auxiliary power unit can be connected to the 391A power unit. If additional 48-VDC power is needed, connect some telephones to KS-22911 or 329A telephone power units.

Protection Accessories

Certain accessories are used for grounding and protecting special telephone connections from power surges, electromagnetic interference, and electrostatic discharge.

IROB Protection

Equipment connected to the analog multiline telephone station jacks (on the

008, 408, 408 GS/LS-MLX (Release 2.0 only), and 408 GS/LS/TTR modules) or to the MLX telephone jacks (on the 008 MLX or 408 GS/LS-MLX module) that is located in a different building but within 1000 feet (305 meters) of the control unit requires in-range out-of-building (lROB) protection units. These units protect the equipment and the control unit from lightning strikes and power surges. Two units are required for each piece of equipment—one for the control unit end of the wire run, the other for the equipment end.

NOTE:

012 basic telephone ports may not be used for out-of-building or offpremises telephones.

Use the following IROB protectors:

■ TII Model 343 for analog multiline telephones and equipment

■ Model 505A for MLX telephones and equipment


Hardware Components

WARNING:

The IROB protectors must be installed by a qualified service technician or installer.

See the documentation packaged with the lROB protector for complete installation instructions.

OFF-PREMISES ON-PREMISES

Figure 2-20. Analog IROB Connection


Hardware Components

Figure 2-21. MLX IROB Connection

Off-Premises Range Extender (OPRE)

If a single-line or tip/ring telephone is located 1000 feet (305 meters) or further from the control unit, connect the telephone to the control unit using an offpremises range extender (OPRE).


Hardware Components

Also, if the network interface is greater than 25 feet (7.6 meters) from the control unit, connect the control unit to the network interface using an OPRE.

See the documentation packaged with the OPRE for complete installation instructions.

146A and 147A Surge Protectors

Protection from lightning and power surges is needed to safeguard system functioning.

It is the responsibility of the local telephone company to provide primary protection on the outside lines at the network interface and to ensure that these protectors are properly grounded. If the telephone company line protector is properly grounded and bonded to the AC power ground, most lightning damage will be prevented.

The 391A1 power supply has built-in AC line protection. This built-in protection handles almost all situations.

Occasionally, additional AC line protection maybe needed if the customer is located in a heavy lightning area. A 147A protector an be connected to the system to limit surges from the AC lines and outside lines. One 147A protector provides protection for four outside lines. Up to three 146A protectors can be added to the 147A to provide protection for a maximum of sixteen outside lines.

For more than sixteen lines, additional 147A protectors are required (see Figure

2-22).

NOTE:

The 147A protector is usually not needed with the 391A I power supply. It may be needed with the older 391A power supply in high-risk lightning areas.

Installation instructions are provided with the protectors. See Figure 2-22.

2-56 Power-Related Hardware

Hardware Components

Figure 2-22. Surge Protectors

EMI Filter

The Z200A electromagnetic interference (EMI) filter can be connected to the system between the control unit and a telephone. Instead of a D8W cord, the filter cord is plugged into the telephone LINE jack. The Z200A filter must be installed with the SMDR printer.

ESD Suppression Kits

Electrostatic discharge (ESD) kits can be installed in older analog multiline telephones with membranes to eliminate damage to the telephone that can be caused by a voltage discharge resulting from electrostatic build-up.

Ring Generator

The 129B Frequency Generator (ring generator) must be added to the 391A1 or

391A power supply module when a 012 basic telephone module or a 008 OPT module is installed in the carrier. It provides a 105-VAC, 3(3-Hz ringing current used by the ringers on the single-line telephones connected to these modules.

System Alarms

An alarm condition detected by the system can cause the control unit to activate an alarm device on a loop-start port. When the contacts close, a signal is passed on to a Universal Paging Access Module (UPAM) and then to an external alert. Alerting devices can be a strobe, horn, bell, or chime.


Hardware Components

An UPAM is needed because 48-VDC alerting devices require four contact closures and the ground-start or loop-start ports only have two. The UPAM provides the additional two.

Trouble Alarm

A ground-start or loop-start power failure transfer (PFT) port can be used to

activate an alarm by connecting the port to an UPAM (see Figure 2-23). When

system trouble (software or hardware malfunction) is detected by the system operator console, a signal is sent to that port. The port’s switching contacts close and send the signal on to the UPAM. The UPAM activates an external alert.

Power Failure Alarm

A ground-start or loop-start PFT port can be used to activate an alerting device

during a power failure by connecting the port to an UPAM (see Figure 2-24).

When a power failure occurs, the switching contacts on the PFT port close and send a signal to the UPAM, which activates an external alert.

NOTE:

A PFT telephone cannot be used on this port when the port is connected for a power failure alarm.

Power

Failure

DID Busy-Out

The PFT port on a ground-start or loop-start module can be programmed to automatically short the “busy-out” wire pair associated with a group of DID trunks. Normally a loop-start line/trunk is used as the busy-out pair. When a power failure occurs, shorting this busy-out pair signals the telephone

company’s CO that the DID trunks are out of service. Figure 2-25 shows this

connection.

NOTE:

Before the ground-start or loop-start module containing the PFT port for the

DID busy-out is removed, the busy-out pair must be shorted and then the modular cord must be disconnected from the PFT jack. Otherwise, a false busy-out will occur. The short is removed after the system is powered up.


Hardware Components

Figure 2-23. Trouble Alarm Connections

Figure 2-24. Power Failure Alarm Connections


Hardware Components

Figure 2-25. Power Failure DID Busy-Out Connections

Power Failure Transfer Telephone

A PFT telephone is a single-line telephone connected to a PFT jack on a 400,

400/GS/LS/TTR, 408, 408, GS/LS, 408 GS/LS-MLX (Release 2.0 only), 800, or

800 GS/LS module. In the event of a power failure, the system shuts off and the

PFT telephone automatically connects to the associated outside line for making and receiving calls.

NOTES:

The PFT jack does not operate unless a power outage occurs or the power supply units are turned off.

A single-line telephone connected to an MFM cannot be used as a PFT telephone.

If PFT telephones are to be connected to ground-start lines/trunks, a ground-start button must be added to each PFT telephone. If power fails, this button is used when the number is dialed. If the button is used with modular 2500 sets, the button should be wired from the wall jack.

If rotary lines/trunks are used, rotary telephones must be used as PFT telephones.



Telephone lines and telephone trunks are facilities that carry voice or data communications. They are similar in form and function. The fundamental difference between a line and a trunk is as follows: a line connects a telephone to a switching system, and a trunk connects one switching system to another switching system. Most of the facilities that connect the system to the central office (CO) are properly referred to as trunks. However, a system that is configured for Hybrid/PBX mode supports personal lines. These facilities usually appear on a voice terminal button and pass transparently through the system to the CO. Selecting a personal line button on a voice terminal and lifting the handset brings dial tone directly from the CO.

NOTE:

The system cannot support digital data transmission via dedicated facilities

(DS1/Tie/PRl) connecting two MERLIN LEGEND Communications Systems.

Loop-Start Lines/Trunks

Loop-start lines/trunks are the standard for home and small business Key systems. They are less expensive in some areas but have certain limitations:

■ They do not protect against glare. (Glare occurs when a person tries to make an outside call on a line/trunk at the same time an incoming call is being received on that line/trunk.)

■ They can have higher cable losses and, therefore, transmissions of less quality than ground-start lines/trunks.

■ They cannot assure secure toll restriction.

Loop-Start Lines/Trunks 3-1

Lines and Trunks

Loop-Start Trunk Connection

The system’s control unit can connect to another system’s control unit from an off-premises telephone (OPT) line to a loop-start line/trunk (or vice versa) via analog facilities. A 008 OPT station module on system A’s control unit can be connected to a loop-start port on system B’s control unit. This enables the user on system B to access all the stations and facilities on system A. If System B has remote access, the user on system A can directly access stations and facilities on system B without operator intervention. Conversely, a loop-start port on system A can be connected to an OPT port on system B.

NOTE:

If the systems are on the same premises, the connection can be made to any T/R port.

With the OPT/loop-start connection, glare is more frequent as the volume of calls increases. In addition, if system B (the loop-start interface) does not have remote access, only the stations assigned to the loop-start facility on system B can be accessed by system A.

Ground-Start Lines/Trunks

Ground-start lines/trunks are outside lines/trunks used by some businesses

(such as hotels or motels) where improved signaling is important. The improved signaling of ground- start allows more secure toll restriction. In addition, ground-start lines/trunks prevent glare. Ground-start lines/trunks also provide cable losses

≤

4.5 dB.

The following types of outside lines/trunks can be either ground-start or loopstart:

■ Basic lines/trunks (used for both local and long- distance calls)

■ WATS (wide area telecommunications service)

■ 800 service (in-WATS)

■ Foreign exchange (FX)

Tie Trunks

Tie trunks provide private communication between two systems. Tie trunks “tie” the two systems together, making it seem that all the telephones are on the same system. A tie trunk connection can be either analog or digital.

3-2 Tie Trunks

Lines and Trunks

■ Analog Tie-Trunk Connection

In an analog tie trunk connection, the system’s control unit is connected to the control unit of another system via a 400EM module. if both systems are on the same premises, this module can be connected directly to the other system if the other system has similar tie-trunk facilities

For off-premises connection, the 400EM module can be connected via the telephone company’s facilities to another system.

An analog tie-trunk connection can be administered for two-way traffic or for one-way traffic (incoming or outgoing). The one-way mode prevents blocked calls caused by glare.

■ Digital Emulated Tie-Trunk Connection

In a digital emulated tie-trunk connection, the system’s control unit is connected to the other system’s control unit via a 100D line/trunk module programmed for T1-type transmission. A back-to-back connection from one DS1 facility to the other can be used when the total cable distance is less than 1300 feet (396 meters). To reach a remote system, the DS1 facility connects via a channel service unit (CSU) to the telephone company’s facilities.

Tie trunks provide efficient communication between systems at different locations. These locations can be different floors of the same building, different buildings in the same campus, or different cities or states.

400EM Module Options

Tie trunks can be added to the system via the 400EM module. The 400EM module has four ports that must be programmed individually by selecting trunk options via system programming and setting the DIP switches (located on the front of the module) for E&M-protected, E&M-unprotected, or simplex signaling mode,

The following tie-trunk options need to be programmed via system

programming, See Tables 3-1 and 3-2 for information on DIP switch settings.

■ E&M Signaling Type

— Type 1 Standard (the factory setting)—used when tie trunks are connected to the other system through the local telephone company.

— Type 1 Compatible—used when tie trunks are connected directly to a system that uses type 1 S signaling and is located near this system.

— Type 5—used when tie trunks are connected directly to a system that uses Type 5 signaling and is located near this system.

Tie Trunks 3-3

Lines and Trunks

■ Direction

— Two-way (factory setting)—Calls can be made in either direction.

Outgoing only—Calls can be dialed but not received (no ringing).

— Incoming only—Calls can be received but not dialed (no dialing).

■ Dial Mode

— Rotary (factory setting)

— Touch-Tone

NOTE:

If the 400EM module is programmed for touch-tone dialing and there are no modules in the system that provide touchtone receivers (TTRs) (012, 008 OPT, 400 LS/TTR, 400

GS/LS/TTR, or 800 DID), a 400 GS/LS/TTR module must be installed.

■ Dial Tone determines whether the dial tone originates from the remote or local end of the line:

— Remote (factory setting)—The system sends a dial tone to the remote end.

— Local—The system does not send a dial tone to the remote end.

■ Answer Supervision Time sets a time limit in milliseconds (ins) for the remote station to signal the calling station:

— 300 ms (factory setting)

— 20 to 4800 ms (increments of 20 ms)

■ Disconnect Time sets a time limit in milliseconds for the release of the E or M lead:

300 ms (factory setting)

—

140 to 2400 ms (increments of 10 ms)

3-4 Tie Trunks

Lines and Trunks

■ Signaling Types (also called seizure type)

— Wink (factory setting)—The originating end of the tie trunk transmits an off-hook signal and waits for the remote end to send back a signal (a wink) indicating that it is ready to receive dialing information.

— Immediate—No start signal is necessary, and dialing can begin immediately after the tie trunk is seized.

— Delay—The originating end of the tie trunk transmits an off-hook signal and waits for the remote end to send an off-hook signal followed by an on-hook signal.

— Automatic—incoming calls are routed directly to another station without a start signal. When the user picks up the handset, the signal rings immediately at the other end. This is also called an automatic ringdown tie trunk.

Wink, immediate, and delay types are also called dial-repeating tie trunks.

If you are installing a 400EM module in the control unit, you need Form 3c,

Incoming Trunks—Tie to determine the switch settings prior to installing the module in the control unit. For each 400EM module trunk jack, check the form,

If the E&M Signal column indicates 1C or 5 for a particular logical ID, set the

DIP switches on the front of the 400EM module, as shown in Figure 3-1 and

Table 3-1.

Tie Trunks 3-5

Lines and Trunks

Figure 3-1. Setting the 400EM Module DIP Switches for E&M Signaling

Types lC and 5

3-6 Tie Trunks

Lines and Trunks

Table 3-1, Setting the 400EM Module DIP Switches

2

1

Ports

4

3

E&M Mode

1

4

5

2

3

6

7

8

9

10

Switch

Position

E&M Signaling

Type

DIP

1S (Default)

Unprotected

1C

Protected

E&M Mode Mode

ON OFF NA

ON NA

OFF

OFF

OFF

OFF

OFF

OFF

ON

ON

OFF

OFF

OFF

OFF

OFF

OFF

OFF

OFF

OFF

NA

NA

ON

ON

ON

ON

ON

ON

Table 3-2. Sample DIP Switches for the 400EM Module

Ports E&M Signal Switches

1 and 2 1C Set all switches to OFF

3 and 4 1S Default: no action required

Tie-Trunk Signaling

Tie trunks transmit via three different signaling formats, each made up of a specific mode and a specific type.

The dual in-line packaging (DIP) switches on the 400EM module select the signaling modes needed for tie-trunk transmission; the signaling type is selected during system programming.

Signaling Modes

There are two signaling mode:

■ Simplex mode. Two signaling leads superimposed onto the analog transmission leads provide a 2-pair wire interface for connecting two local systems at minimal cost.

Tie Trunks 3-7

Lines and Trunks

■ E&M mode. This is a standard interface. The E&M signaling leads are isolated from the transmission leads, requiring a 3-pair wire interface.

In the simplex mode, protective resistance is always included in the circuit. The

E&M mode can be either protected or unprotected from high-voltage transients or fluctuations. In the protected mode, a resistance is added to the leads to reduce current peaks, The protected mode is used when there is no network interface to protect the circuit from outside interference.

The unprotected mode must be used for an E&M Type 1 Standard interface

(see below) to meet the specified voltage-drop criteria. This mode is used when there is a network interface.

Signaling Types

Three different signaling types combine with the signaling modes. Together these create the proper signaling format for each system.

■

Type 1 Standard. This is the factory-set type, which is used to connect two systems to the network through two intermediate telephone company

COs. The switches must be set for E&M mode.

■ Type 1 Compatible. This directly connects two systems without intermediate telephone company COs. One system is set to Type 1

Standard, the other to Type 1 Compatible. The switches must be set for

E&M mode.

■ Type 5 simplex or E&M. This type is used to connect similar systems or systems with compatible signaling that are located in the same building or on the same business campus.

The choice of a tie-trunk signaling format to connect two systems depends on the particular application and the systems being connected, including whether or not the tie-trunk signals pass through telephone company lines or over

customer-owned cable. Table 3-3 shows how to determine tie-trunk

compatibility between this system and other systems.

3-8 Tie Trunks

Lines and Trunks

Table 3-3. Tie-Trunk Compatibility

Installation Situation

From system

Preferred Signaling Format

Type 5 Simplex

System

Signaling Mode Protected or and Type Unprotected

N/A

Far End

Signaling Mode and Type

Type 5 Simplex

Protected or

Unprotected

N/A

To

System

MERLIN II

System 25

System 75

System 85 or DEFINITY

Dimension PBX

Location

Same site or interbuilding



Same site

Type 5 Simplex

Type 5 Simplex

N/A

N/A

Unprotected

Type 5 Simplex

Type 5 Simplex

N/A

N/A

Unprotected

Dimension PBX

Other

Other

Network Interface

Interbuilding

Same site

Interbuilding

E&M Type 1

Compatible

E&M Type 1

Compatible

E&M Type 1

Compatible

E&M Type 1

Compatible

E&M Type 1

Standard

Protected

Unprotected

Unprotected

Unprotected

E&M Type 1

Standard

E&M Type 1

Standard

E&M Type 1

E&M Type 1

Standard

N/A

Protected

Unprotected

Requires a protection unit

N/A

E&M Tie-Line Ports

The E&M tie-line circuit module provides four tie-line ports. These ports may be individually configured for Type 1 Standard, Type 1 Compatible, or Type 5

(simplex or E&M).

Type 1 Standard is used to connect to the network. Type 1 compatible is used to co-locate systems that have a Type 1 Standard interface. Type 5 is used to connect co-located systems that have the Type 5 interface.

Table 3-4 shows the E&M option switch settings for the Type 1 Standard and

Type 1 Compatible tie-line modules.

Tie Trunks 3-9

Lines and Trunks

Table 3-4. Type 1 Standard and Type 1 Compatible E&M Switch Settings

Option

Unprotected

I

Switch Switch

Position Setting

3

4

5

1

2

ON

ON

OFF

OFF

OFF

Protected

1

2

3

4

5

OFF

OFF

OFF

OFF

OFF

Tie Trunk Networking

The system supports only nontandem tie-trunk networking. A nontandem tietrunk network is used primarily to connect telephone lines at both ends; it does

not connect to another tie trunk or to other facilities. See Figure 3-2.

3-10 Tie Trunks

Lines and Trunks

Figure 3-2. Nontandem Tie-Trunk Network

Direct Inward Dialing (DID)

Lines/Trunks

Direct Inward Dialing (DID) trunks allow incoming calls to reach specific individuals or facilities in the system without the assistance of a system operator. DID trunks are available only in the Hybrid/PBX mode. They are connected to the system on an 800 DID module or through DID-emulated channels on a 100D DS1 module.

For Release 2.0, DID functionality has been extended to DID-administered Bchannels in the DS1 circuit board in T1 mode. B-channels administered for DID operation behave like tie trunks to the network and like DID trunks to the system switch.

Direct Inward Dialing (DID) Lines/Trunks 3-11

Lines and Trunks

With DID service, the customer reserves blocks of DID numbers from the local telephone company. The DID numbers should correspond to the extension number for an individual, a calling group, or a Remote Access or pool dial-out code.

CAUTION:

DID numbers that correspond to pool dial-out codes (or facility access codes) can be used to evade toll restriction, leading to toll abuse

and/or fraud. (See Customer Support Information in the front of this

book for more information on security.)

The system can receive 1- to 4-digit extension numbers over the DID trunks.

The number of digits received on a specific DID trunk is always the same for that trunk or for the trunks in a particular DID trunk block; however, different DID trunks (or trunk blocks) can receive different numbers of digits.

Because DID trunks allow calls to come directly to a telephone extension, they cannot be pooled. The telephone company’s CO passes the necessary digits to the system, which delivers the call directly to the dialed extension.

If the extension numbers used in the system are fewer than four digits but the

CO sends four, the system can be programmed to ignore the leading digit(s).

For example, if the DID number sent by the CO is 2157, the extension numbers the system can access are 57, 157, or 2157. System programming determines the proper extension number to connect.

The system also can be programmed to match more digits than are received from the CO. For example, if the system is set up to match three digits and the

CO sends the number 24, the system might insert a 9 in front of the 24 (resulting in the number 924) to complete the match and connect the call.

No routing of calls is made until the designated number of digits is received.

Incoming DID numbers that don’t match a valid extension are directed to a predesignated extension, such as the system console, or the system can be programmed to send back a reorder tone.

Options are assigned to blocks of DID trunks. A maximum of two blocks of DID trunks is allowed. Each block can be configured to match the system numbering plan. For example, the system could have both 3- and 4-digit extension numbers. Trunk block 1 could contain the options needed to reach the 3-digit numbers and trunk block 2 could contain the options needed to reach the 4-digit numbers,

3-12 Direct Inward Dialing (DID) Lines/Trunks

Lines and Trunks

The following items must be programmed for each trunk group:

■ Type of DID Trunk

— Wink-start (factory setting)—The preferred setting if the local telephone company can support it. It allows a greater probability of call completion during heavy calling periods,

— Immediate-start—The setting used when the local telephone company can support only immediate-start.

■ Signaling sent from the local telephone company.

— Rotary (factory setting)

— Touch-tone

■ Expected Digits sent by the local telephone company.

— 3 (factory setting)

— 1 to 4

■ Delete Digits The number of leading digits that must be deleted from the digits sent by the local telephone company when the number of digits sent is more than in the chosen system numbering plan.

— 0 (factory setting)—Used when the number of digits sent by the telephone company matches the number of digits in the chosen system numbering plan.

— 0 to 4

■ Add Digits. The specific leading digits that must be added to the digits sent by the local telephone company when the number of digits sent is fewer than the number of digits in the chosen system numbering plan.

— 0 (factory setting)—Used when the number of digits sent by the telephone company matches the number of digits in the chosen system numbering plan,

— 1-to 4-digit number (1 to 9999)


The following system programming is needed in Release 2.0 to administer DID

B-channels on the DS1 circuit module:

■ A menu item to program a single B-channel on the circuit module as DID

■ A menu item to program all B-channels on the circuit module as DID


Lines and Trunks

■ Expand the Inspect feature to show DID channels as well as

LS/GS/TIE/Unassigned

■ A new Inspect screen for DID channels

Programming a DID B-channel automatically adds it to the first DID trunk block.

The following considerations apply:

■ The maximum number of DID trunks in a DID trunk block is the system maximum number of trunks (24 or 80, depending on system configuration).

■ Only trunks on the 800 DID circuit module and B-channels programmed as DID on the DS1 facility are allowed in DID trunk blocks.

■ A DID trunk cannot appear in more than one DID trunk block.

■ If the administrator wants to move a DID trunk from one block to another, the trunk simply can be assigned to the new trunk block. The trunk is automatically removed from the previous trunk block and assigned to the new trunk block. The trunk then takes on the programmed attributes of the new trunk block.

■ A DID trunk always appears in one of the two DID trunk blocks, even if a physical channel is not present between the circuit module and the CO.

The following options need to be programmed via system programming:

■ DID Trunk Dialing Protocol Type describes the dialing protocol used for determining when address digits are sent from the CO to the Release

1.0 system.

— Wink Start (factory setting)—the Release 1.0 system signals the

CO when it is ready to receive incoming address digits.

— Immediate Start—the CO sends digits about 65 ms after line seizure. (This is a criterion required of the serving CO.)

Programming of the Immediate Start type for a trunk block that uses dual-tone multifrequency (DTMF) for passing address signs is not allowed.

■ DID Trunk Address Signaling Type describes the method by which address signals are transmitted from the serving CO to the Release 1.0

system.

— Dial Pulse (factory setting)

—

DTMF (not allowed for Immediate Start dialing protocol trunks)

■ Expected Number of Digits (a number from 1 to 4, factory setting = 3) indicates the number of address digits expected from the CO on DID calls in this trunk block. Assignment of a value greater than 4 or less than

1 will be blocked.

3-14 Direct Inward Dialing (DID) Lines/Trunks

Lines and Trunks

■ Number of Digits to Delete (a number from 0 to 4 digits, factory setting

= 0) describes the number of digits to delete from the incoming address digits. A values greater than 4 will be blocked.

■ Digits to Add (a number from 0 to 9999, factory setting = 0) describes the digits to prepend to the collected digits in order to determine a routing number,

The following options can be programmed on a DID trunk-by-trunk basis:

■ Trunk Disconnect Timing (a number from 10 ms to 2550 ms, factory setting = 500 ms) is used by the DID circuit module to determine the time needed before a disconnect from the CO is considered valid.

■ Trunk Number is the “line number, ” assigned to every trunk or line in the system on startup, that serves as the trunk number. This trunk number can be changed by System Numbering.

■ Alphanumeric Label is an ASCII string with up to seven characters

(factory setting = OUTSIDE) that can be assigned to an individual DID trunk.

DID trunks can be programmed in a pool but on system resets these trunks are not placed in a default pool. The board renumbering mechanism automatically removes DID trunks from pools.

DID trunks can be assigned via system line programming as DFTs on any station in the system, but they will never be automatically assigned to any station in the system, including the operator position. The intent of having a DFT for a DID trunk is not for receiving or originating calls, but to allow monitoring the facility by observing the lamp.

DID trunks cannot be assigned as Music On Hold ports, page ports, or given direct access to calling groups.

You can program the DID feature in modes other than Hybrid/PBX; note, however, that the DID feature works on/y in Hybrid/PBX mode.

In Release 2.0, if the B-channel already has been programmed as a non-DID type trunk, the following items will block programming of the B-channel for DID operation:

■ Trunk is programmed to ring into a DGC group

■ Trunk is programmed for Remote Access

■ Trunk is owned by a station for Coverage/Call Forwarding


Lines and Trunks

DS1 Facilities

A Digital Signal 1 (DSl) facility is a transmission system that transports digital signals in the DS1 format. The interface that allows the connection of DS1 facilities to the system is the 1000 module. Through this module, voice and data calls can be made or received using a DS1 facility.

Twenty-four digital signal 0 (DS0) channels, each operating at 64 kbps, plus framing bits, are multiplexed, forming a DS1 signal of 1.544 Mbps. Each DS0 channel within the DS1 signal corresponds to a logical endpoint. Even though there is only one physical jack, the 10013 module supports up to 24 logical endpoints or ports (one for each channel).

In DS1 format, calls to other digital private branch exchanges (PBXs) or telephone company CO remain digital, and signals do not need to be converted to analog for acceptance by the connecting trunk. In addition, the 100D module can be configured to work with T1 or Primary Rate Interface (PRI) service.

To connect the 100D module to an outside DS1 facility, a CSU is used. The

CSU regulates the transmission into and out of the 100D module so that the module matches the transmission of the outside facility.

Both ends of the DS1 facility must be able to communicate. To ensure this, the following options are set during system programming to match the transmission of the outside DS1 facility:

■ Type of service (T1 or PRI)

■ Framing format

■ Line code

■ Line compensation

■ Clock synchronization

■ Signaling mode (for T1 service only)

The appropriate setting for each option is determined by the transmission facility to which the module is connected. Each option is discussed below.

Type of Service (T1 or PRI)

The system supports two types of service for DS1 facilities: T1 and PRI. The

100D module can be programmed to operate in either type of service. T1 service transmits and receives voice and analog data; PRI transmits and receives voice, and analog and digital data.

Any combination of the following AT&T Services Network (ASN) Services can be provided through a T1 or PRI line/trunk:

3-16 DS1 Facilities

Lines and Trunks

■ Megacom WATS service for domestic long-distance outward voice calls;

PRI on the system does not support access to international Megacom

WATS service.

■ Megacom 800 for domestic toll-free incoming voice calls. PRI on the system does not support access to international Megacom 800 service.

T1 and PRI services support Megacom 800 with or without Dialed

Number Identification Service (DNIS), also called Routing by Dial Plan.

Dialed Number Identification Service (DNIS) is a service provided by the

AT&T Switch Network that routes incoming 800 or 900 calls according to customer-selected parameters, such as area code, state, or time of call.

For example, a customer can specify that calls received from a particular area code should be routed to a specific individual or group responsible for accounts in the area.

■ Software Defined Network (SDN)—ASN service; for voice and circuitswitched data calls. SDN lets businesses use portions of the ASN in concert with their dedicated private line networks. However, the system does not support “uniform dialing plan,” which is necessary for complete integration with SDN. PRI on the system not support access to global

SDN service.

■ MultiQuest for domestic toll incoming voice calls (900 number). T1 and

PRI support MultiQuest with or without DNIS.

In addition to these ASN services, T1 and PRI also support Shared Access for

Switched Services (SASS) and Call-by-Call Service Selection. SASS allows both

Megacom and Megacom 800 services to be offered over the same Iine/trunk facilities, eliminating the need to have separate incoming and outgoing line/trunk groups.

Like PRI, T1 also supports Megacom WATS and Megacom 800 on a shared line/trunk, but on a call-by-call basis. Call-by-Call Service Selection provides more than one outgoing PRI service, such as Megacom WATS, Accunet ®

Switched Digital 56/64, SDN, OUTWATS, Virtual Private Network Access, and

Long Distance. PRI also provides Accunet switched digital service for 56-kbps,

64-kbps restricted, and 64-kbps clear circuit-switched data calls.

T1

T1 is the factory setting for DS1 facilities, allowing each of the 24 channels to be programmed to emulate tie, loop-start, ground-start, and DID lines/trunks in any combination. This means that a single 100D module can take the place of 24 regular outside lines.

If common-channel signaling (CCS) is selected, 23 channels are available for emulation and the 24th channel carries formatting signals.

The system’s control unit can be connected to another system’s control unit via a digital emulated tie trunk on a DS1 facility connected to a 100D module programmed for T1 -type transmission. A back-to-back connection from one

DS1 facility to the other can be used when the total cable distance is less than

1300 feet (396 meters).

DS1 Facilities 3-17

Lines and Trunks

Primary Rate Interface (PRI)

The Primary Rate Interface (PRI) is a standard access arrangement that can be used to connect the system to a network providing voice and digital data services through a 4ESS™ Generic 16, a 5ESS Generic 6, and a 5ESS serving the FTS2000 network.

A PRI line consists of 24 channels, sometimes referred to as DS0 channels, each with a capacity of 64 kbps. Each channel can be designated as either a

B-channel (bearer channel) or a D-channel (data channel). DS1, then, refers to the twenty-four 64-kbps channels plus framing and signaling bits multiplexed together to form a 1,544-Mbps signal.

A B-channel is used to carry end-to-end user information, such as the voice or data content of a call, between the system and the far-end switch. Each Bchannel provides access to one or more network services. Release 1.0 of the system supports access to only one network service per B-channel. For

Release 2.0, Call-by-Call Service Selection allows multiple network services over the same B-channels. The D-channel conveys signaling required to set up, control, and clear calls made over all of the B-channels.

The most common configuration of a PRI consists of 23 B-channels and 1 Dchannel, although other combinations are possible. Each PRI must include a

D-channel, but may include fewer than 23 B-channels. The remaining channels cannot be used for any other purpose.

Up to three PRIs can be connected to the system through separate 100D circuit modules, each of which may occupy a slot in the system carrier. In terms of system capacity, each PRI line counts as a trunk endpoint, so the maximum number of B-channels supported by the system is 69. Their signaling is provided over three separate D-channels.

Framing Format

To identify the DS0 channels, the DS1 signal is segmented into blocks of 193 bits called frames. A frame consists of 24 eight-bit words (one for each channel) plus a framing bit at the beginning of each frame: 24 words x 8 bits =

192 bits. Thus, a framing bit appears in every 193rd bit position of the 1.544-

Mbps DS1 signal.

Frames repeat at a rate of 8000 per second, with each frame repeating DS0 channels 1 through 24 sequentially.

Two methods of framing can be used by a 100D module (T1 service): D4 or

Extended Super Frame (ESF). The framing method chosen must match the framing at the far end, and must be programmed to the format selected when service was ordered.

To identify the DS0 channels, the DS1 signal is segmented

■ D4 Framing Format. The system is factory set for the most common framing format, D4 framing. A D4 frame consists of 24 eight-bit time slots and one framing bit. To perform synchronization, the receiving equipment uses the framing information to identify the start of each frame

3-18 DS1 Facilities

Lines and Trunks and to identify which frames contain signaling information. The framing information repeats once every 12 frames; these 12 frames form the 134 superframe. This framing format is used by most DS1 equipment.

■ ESF Framing Format. The ESF format extends the 12-frame D4 superframe to a 24-frame superframe, hence its name. The 24 framing bits include a cyclic redundancy check (CRC) for the entire extended superframe and a facility data link for maintenance. The ESF can detect more errors than D4 framing; however, ESF is not used universally by

DS1 equipment.

Line Coding

The DS1 signal consists of a continuous bit stream of ones and zeros, encoded into bipolar pulses for transmission. Only the ones create a pulse; the zeros represent the absence of a pulse. The pulses of the ones alternate between positive and negative. This type of line coding is called bipolar or alternate

mark inversion (AMI) zero code suppression (ZCS). The line-coding formats guarantee that the “ones-density” requirement is met to achieve clock recovery.

To meet the ones-density requirement, either AMI-ZCS or bipolar 8 zero substitution (B8ZS) line coding can be chosen.

AMI-ZCS

AMI-ZCS line coding monitors each DS0 channel and prevents strings of eight or more zeros. Upon detecting an all-zero channel octet, AMI-ZCS line coding forcibly changes the seventh zero (second least significant bit) to a one. The factory-set line coding is AMI-ZCS.

With AMI-ZCS line coding, any bit that is overwritten has no noticeable effect on voice and voice-grade data, However, the AMI-ZCS line-coding format can cause errors in digital data transmission.

B8ZS

B8ZS line coding inserts eight consecutive zero bits into a unique binary sequence with a “bipolar violation” in bit positions 4 and 7. Normally for bipolar transmission, ones are encoded alternately as a positive then negative, or negative then positive, pulse. If two positive or two negative pulses are received in succession, a bipolar violation occurs,

Ordinarily, bipolar violations are caused by noise hits to the signal, However,

B8ZS line coding allows the 8-bit strings to be detected at the receiving end and converted back into the original sequence.

B8ZS line coding is preferred over ZCS because it does not cause errors in data transmission.

B8ZS violations are passed by the ESF T1 CSU but not by other CSUs. The

CSU is a hardware component needed when two endpoints are located in different buildings or when the distance between the two endpoints makes office or line repeaters necessary. The CSU is located on the customer’s


Lines and Trunks premises and is used to connect the system to DS1 network facilities. The CSU has three functions:

■ It terminates an outside DS1 facility on the 100D module.

■ It ensures that the signals entering the public network comply with the requirements of the DS1 facility as specified by the FCC.

■ It includes maintenance, diagnostic, and testing capabilities.

Line Compensation

Line compensation adjusts for the amount of cable loss in decibels (dBs), based on the length of cable between the 100D module and the CSU or other far-end connection point. The factory setting is a value of 1, which allows a maximum loss of 0.6 dB. The other possible settings are shown in Table 3-5.

Table 3-5. Line Compensation Settings

Cable Length

Setting dB Loss (22-Gauge Wire)

1

2

0.6

1.2

0—133

133—266

3

4

1.8

2.4

266—399

399—533

5 3.0

533—655

NOTE:

Cable length in Table 3-5 is the distance between the 100D module and the

CSU. If no CSU is used, the distance between 100D modules is twice these numbers.

Clock Synchronization

Clock synchronization is an arrangement where digital facilities operate from a common clock. Whenever digital signals are transmitted over a communications link, the receiving end must be synchronized with the transmitting end to receive the digital signals without errors.

The system synchronizes itself to the network by extracting the timing signal from the incoming digital stream. If the system has more than one 100D module, the module that provides the primary synchronization for the other

100D modules and for the time-division multiplexing (TDM) bus must be identified during system programming.

3-20 DS1 Facilities

Lines and Trunks

Backup synchronization in the event of a maintenance failure can be provided by programming the second and third installed modules as secondary and tertiary synchronization.

In addition, the source of synchronization can be factory set to “loop clock reference source” (the clock is synchronized to the external endpoint—the factory setting) or set to “local clock reference source”

Signaling Mode

Signaling is the process of communicating channel-state information (such as dialing) from endpoint to endpoint. Two types of signaling can be used in T1 transmission: robbed-bit signaling (RBS) and common-channel signaling (CCS).

Choosing a signaling mode pertains only to T1 service; PRI always uses CCS

(23 B-channels and 1 D-channel).

Robbed-Bit Signaling

RBS replaces (“robs”) the least significant bit of every sixth frame of each DS0 channel with signaling information. (RBS is also called in-band signaling. since signaling information is embedded in the least significant bit of every sixth 8-bit word.)

RBS is appropriate for voice and voice-grade data (up to 19.2 bps), but facilities using RBS cannot transmit digital data at 64 kbps because this bit-robbing corrupts data. Digital data at 56 kbps may be possible in certain applications by using 7-bit words.

Common-Channel Signaling

CCS is an format that places the signaling bits for channels 1 through 23 into the 8-bit word of the 24th channel. This restricts DS1 from using the 24th channel for voice or data transmissions. D4 framing does not preclude the use of CCS, but CCS is not compatible with D4 channel banks because the D4 channel banks only recognize RBS. Coupled with B8ZS coding, CCS can support digital data up to 64 kbps per channel.

Recommended Framing Formats and Signaling

Modes

ESF framing should be used to take advantage of its improved maintenance, diagnostic, and testing capabilities (the ESF T1 CSU is required to interface with the network). If the transmission between two systems is voice-only, RBS should be used for all 24 communication paths. For voice transmission, both

ZCS and B8ZS line coding can be used to satisfy the ones-density requirement; the preferred line-coding format is B8ZS, which is needed for 64 kbps digital data.


Lines and Trunks

The framing and signaling formats depend on the network and interconnection devices (CSUs) used. For example, many CSUs only support ZCS line coding.

NOTE:

■ Digital data up to 64 kbps is possible only in PRI mode.

■ An ESF-T1 CSU must be used for interbuilding DS1 connections.

3-22 DS1 Facilities

Applications

CAUTION:

This chapter is intended solely as an overview of the applications that can be connected to the system. For information about the use of any application listed here, see the documentation for that product.

The following applications can be connected to the system for enhanced callhandling and system management capabilities:

■ Standalone voice messaging applications

— MERLIN MAIL Voice Messaging System

— MERLIN Attendant

■ Standalone call accounting and management applications

— Call Accounting System (CAS)

– Call Accounting Terminal (CAT)

– Call Management System (CMS)

— InnManager Guest Management System

■ Standalone system management application

— System Programming and Maintenance (SPM)

■ Integrated applications

Integrated Solution II (IS II) incorporates the following applications:

— AUDIX Voice Power

— CAS

— SPM

Applications 4-1

Applications

Integrated Solution III (IS Ill) incorporates the following applications:

— AUDIX Voice Power

— CAS

— SPM

— Fax Attendant

— CONVERSANT Intro

■ Primary Rate Interface (PRI) applications

— Group IV (G4) Fax

— Video Conferencing

■ Optional telephone service

— Centrex operation

■ Fax services

— MERLIN PFC Telephone

— Automated Document Delivery System (ADDS)

■ Voice response system

— CONVERSANT Intro

This chapter provides a brief description of each of these applications, services, and systems. The descriptions are organized under the following subheadings; any subheading not applicable to a given application is omitted.

■ Mode Differences—differences or limitations of the application in the

Key, Hybrid/PBX, or Behind Switch modes of operation.

■ Considerations and Constraints—restrictions, capacities, and other information to be considered before installing or using the application.

■ Feature Interactions—system and telephone features that affect how the application works, and any features that do not work with the application.

■ System Programming—an outline of the communications system programming required to set up the application. See System P/arming for planning instructions and System Programming for system programming instructions. Also see the documentation provided with the application for instructions to program the application.

■ Platform Requirements—additional hardware and software required to connect the application to the system. (See Installation and the documentation provided with the application for connection diagrams and installation instructions. )

Table 4-1 summarizes the capacity of the system to support each application

and the modes of operation in which it can be used.

4-2 Applications

Applications

Table 4-1. Application Capacities and Modes of Operation

Application

MERLIN MAIL Voice Messaging System

Number of mailboxes

MERLIN Attendant

CAS

CAT

CMS

Number of lines/trunks (each)

Number of agents (each)

Number of external alerts (each)

InnManager Guest Management System

SPM (standalone)

IS II

AUDIX Voice Power

Number of mailboxes

Integrated Voice Power

Automated Attendant

CAS—IS II

SPM—IS II

IS Ill

AUDIX Voice Power

Number of mailboxes

CAS—IS Ill

SPM—IS Ill

Fax Attendant

PRI

Group IV (G4) Fax

Video Conferencing

Centrex

ADDS

Capacity

1 *

(2 or 4 ports)

40

4*

1

1

2

28

28

4

1

1

300

1

1

1

1

1

1

1

1

300

1

1

1

CONVERSANT Intro 1

1 voice port and

1 fax port

Hybrid/ Behind

Key PBX Switch

*

These attendant applications are mutually exclusive.

Applications 4-3

Applications

Voice Messaging Systems

CAUTION:

This section is intended solely as an overview of the application. For comprehensive information about the use of the application, see the documentation for the product.

A voice messaging system (VMS) is an application that provides call answering services and may provide voice mail services as well. Each of the following

VMS applications is connected to an enhanced tip/ring port, called a voice

messaging interface (VMI) port.

■ MERLIN MAIL Voice Messaging System

■ MERLIN Attendant

■ AUDIX Voice Power (IS Ill)

■ Integrated Voice Power (lVP) Automated Attendant (IS Ill)

4-4 Voice Messaging Systems

Applications

CAUTION:

Your Voice Mail system permits callers to leave verbal messages for system users or gain access to the back-up position in an emergency as well as create and distribute voice messages among system users.

The Voice Mail system, through proper administration, can help you reduce the risk of unauthorized persons gaining access to the network.

However, phone numbers and authorization codes can be compromised when overheard in a public location, are lost through theft of a wallet or purse containing access information, or through carelessness (writing codes on a piece of paper and improperly discarding them). Additionally, hackers may use a computer to dial an access code and then publish the information to other hackers.

Substantial charges can accumulate quickly. It is your responsibility to take appropriate steps to implement the features properly, evaluate and administer the various restriction levels, protect and carefully distribute access codes.

Under applicable tariffs, you will be responsible for payment of toll charges. AT& T cannot be responsible for such charges and will not make any allowance or give any credit resulting from unauthorized access.

To reduce the risk of unauthorized access through your Voice Mail system, please observe the following procedures:

■ Employees who have voice mailboxes should be required to use the pass words to protect their mailboxes.

—— Have them use random sequence passwords.

— Impress upon them the importance of keeping their passwords a secret.

— Encourage them to change their passwords regularly.

■ The administrator should remove any unneeded voice mailboxes from the system immediately.

■ AUDIX Voice Power has the ability to limit transfers to subscribers only. You are strongly urged to limit transfers in this manner.

■ Use the system programming capability to do the following:

— Block direct access to outgoing lines and force the use of account codes/authorization codes.

— Disallow trunk-to-trunk transfer unless required.

— Assign toll restriction levels to all AUDIX Voice Power ports.

— If you do not need to use the Outcalling feature, completely restrict the outward calling capability of the

AUDIX Voice Power ports.

■ Monitor SMDR reports or Call Accounting System reports for outgoing calls that might be originated by AUDIX Voice Power ports.

Voice Messaging Systems 4-5

Applications

Tip/ring ports on an 012 module can be programmed either as generic VMI

ports or integrated VMI ports. The MERLIN MAIL Voice Messaging System and

AUDIX Voice Power use streams of touch-tone codes, called mode codes, to

communicate with the system’s control unit. Because they use mode codes, these applications must be connected to integrated VMI ports, MERLIN

Attendant and IVP Automated Attendant, which do not use mode codes, connect to generic VMI ports.

Mode codes are categorized in two classes: Call Information and Other. Table

4-2 lists the types of mode codes in each class.

Table 4-2. Mode Codes

C a l l I n f o r m a t i o n

Direct inside access

Direct dial access—line/trunk

Call coverage—internal station

Call coverage—external (line/trunk)

Call coverage-other

O t h e r

Leave word calling

Refresh message-waiting LEDs

Port taken out of service

Port restored to service

Day service

Night service

A VMS requires touch-tone receivers (TTRs); the number required depends on

the number of VMI ports, as shown in Table 4-3. Note that these TTR

requirements are only for a VMS and do not include the TTR needs of tip/ring sets.

TTRs are supplied by the following modules: 012, 400, and 400 GS/LS/TTR.

(The 008 OPT module also supplies TTRs, but does not support VMS applications. )

The following symptoms indicate that the system needs more TTRs:

■ Single-line telephone users do not get dial tone when trying to dial out.

■ The voice messaging system fails to transfer calls.

■ Calls fail to ring or go to coverage prematurely.

4-6 Voice Messaging Systems

Applications

Table 4-3. TTRs Required by Voice Messaging Systems

<

No. of No. of No. of 012

VMI Ports TTRs Required Modules

No. of 400 or

400 GS/LS/TTR

Modules

4

6

2

3

1

8

12

1

2

2

3

4

6

1

1

1

1

3 or

2

2

1 or

1

2

1 or

1

0

0

0

0

0

0

1

0

1

VMI Port Capabilities

VMI ports use switchhook flashes in the same way single-line telephones do for

Hold, Transfer, Conference, and Drop. VMI ports also have the ability to perform transfer redirection, respond to far-end disconnect, and, for integrated

VMI ports only, mark a port in service or out of service. These capabilities are described in the following sections.

Transfer Redirect

If unanswered by the end of the transfer redirect time interval (programmable for

0 to 9 rings), a call transferred from a VMI port will alert at the VMS transfer redirect extension, rather than return to the VMI port that originated the transfer.

For example, suppose station port 15 is programmed as a VMI port connected to a MERLIN Attendant, and the programmed transfer redirect time interval is

4 rings. A call comes in on port 15, and after listening to the recorded prompt, the caller dials a request for extension 24 (station port 24). The call rings at station 24 for four rings without being answered. The call is then redirected to station port 10, the system operator. It is not redirected back to port 15.

On an unsupervised transfer (described later in this chapter in the section

“Automated Attendant” ), when the transfer destination is busy or is an invalid

extension, the transfer redirect is immediate (no time interval). If the transfer redirect station cannot be alerted (all buttons are being used), the VMS will keep trying to alert the transfer redirect station every 20 seconds until the alert is delivered or the caller hangs up.

Voice Messaging Systems 4-7

Applications

Far-End Disconnect

When a far-end disconnect signal is detected on an outside line/trunk on which a call is made to or received from a VMI port, the system sends the disconnect signal to the VMI port, whether or not that port is the only party left on the call. If another party is still on the call, the VMS decides whether to continue or disconnect the party. (The far-end disconnect signal occurs only if the

LS/GS/DID/Tie VMI port is programmed for reliable disconnect.)

Ports In/Out of Service

When a calling group call to a VMI port is not answered within 30 seconds, the call is sent to another available VMI port in the calling group or is queued back to wait for an available port in the calling group.

For an integrated VMI port, the control unit sends mode codes to inform the

VMS that the port is out of service. Both the VMS and the calling group software mark the unavailable port out-of-service. If all VMI ports go out of service, a programming logic inconsistency (PLI) is generated.

Every 10 minutes the system tests each out-of-service VMI port. If the port responds to the test, the VMS and the calling group software mark it in-service.

For an integrated VMI port, the control unit informs the VMS by sending port-inservice mode codes.

4-8 Voice Messaging Systems

Applications

MERLIN MAIL Voice Messaging

System

CAUTION:


The MERLIN MAIL Voice Messaging System is a standalone application that provides the following integrated call-handling services:

■ Automated Attendant Service

■ Call Answer Service

■ Voice Mail Service

Automated Attendant Service

The Automated Attendant Service answers incoming calls and plays a menu of recorded prompts, A caller can respond to the prompts by dialing touch-tones, and the Automated Attendant routes the call to an internal extension accordingly. If there is no answer or the extension is busy, the caller can be given the option to leave a message or try another extension.

A caller without a touch-tone telephone is transferred to the system operator for further call handling and routing.

The system manager can record multiple levels of menus and announcements, including separate menus for day and night service.

Calls can be answered immediately (Immediate Call Handling) or after a delay

(Delayed Call Handling), for example, if the call goes unanswered by the system operator after a specified number of rings.

The Automated Attendant Service can recognize fax tone on an incoming call and direct the call either to a single fax extension or to a calling group serving multiple fax machines. If the fax machine is busy or does not answer within four rings, the call is automatically disconnected.

The Automated Attendant Service can be programmed to transfer calls in either of three ways:

■ Unsupervised transfer— in combination with the system’s Coverage feature, the Automated Attendant dials the extension or department requested by the caller and disconnects. If the call is not answered or the extension is busy, the call is routed to the system operator or, if the user is a registered subscriber, returns to the Automated Attendant,

MERLIN MAIL Voice Messaging System 4-9

Applications

■ Supervised transfer— the Automated Attendant transfers the call and can retrieve it if the transfer is unsuccessful. If the called party is not a subscriber, the caller can opt to be transferred to another location.

■ No transfer— the Automated Attendant transfers the call, and the caller is prompted to leave a message.

Call Answer Service

When a caller reaches a busy or unanswered extension, the Call Answer

Service connects him or her to the personal mailbox of the subscriber associated with that extension, where the caller can leave a message. If the subscriber has recorded a personal greeting, the caller hears it; otherwise, a general greeting including the subscriber’s name is played.

If the subscriber’s personal mailbox is full, the Call Answer Service connects the caller to a general mailbox and plays a message including the subscriber’s name. The caller can leave a message in the general mailbox; the system operator is responsible for forwarding the message to the appropriate subscriber,

If the general mailbox is full, the Call Answer Service informs the caller and allows him or her to transfer to another extension.

When a message is left in a subscriber’s personal mailbox, the system lights the message-waiting LED on his or her telephone. When a message is left in the general mailbox, the general mailbox owner’s (typically the system operator) message LED goes on.

With Outcalling, when a user, or subscriber, receives a new message, the system can automatically call a number that he or she has programmed, for example, a beeper or a home telephone. The subscriber can then log in to the

Voice Mail Service to retrieve messages.

Voice Mail Service

The Voice Mail Service allows subscribers to send messages to other extensions in the system, forward messages received with comments, and return a call to an extension that has left a message. A subscriber can also record a personal greeting and program a password to prevent others from retrieving messages from his or her personal mailbox. The system manager can broadcast a message to every system subscriber. A broadcast message does not light message LEDs and does not cause outcalling.

In addition, the system manager can create group lists of subscribers. Any subscriber can send a message to a group list.

Additional features include the following:

■ Outcalling automatically calls the user at a number the user has programmed when a new message is received. The user can then log in to the VMS to retrieve messages.

4-10 MERLIN MAIL Voice Messaging System

Applications

■ Pager Notification calls the user at a designated number when a new message is received. However, the user cannot log in to the VMS.

■ Broadcast Lists allows the system administrator to send a message to every user on the communications system. However, this message does not light message LEDs and does not cause outcalling.

■ Fax Transfer directs an incoming fax call to a designated fax station.

This fax station can be a single machine or a calling group with several machines.

■ Announcement Service allows a caller to enter a code to hear information about specific subjects, such as new product information or marketing programs.

Mode Differences

The system must operate in Hybrid/PBX or Key mode. The MERLIN MAIL Voice

Messaging System cannot be connected to a system operating in Behind

Switch mode.

Considerations and Constraints

■ Only one MERLIN MAIL Voice Messaging System can be connected to the system.

■ The MERLIN MAIL Voice Messaging System is available in two-port and four-port configurations. Both configurations have four hours of message storage capacity.

■ The size of a subscriber’s mailbox—that is, the total length of the messages it can hold—can be set up to match individual needs, up to a maximum size of 60 minutes. Available options are 5, 10, or 15 minutes;

60-minute storage is available for special mailboxes.

■ Callers who dial from rotary telephones cannot use the features of the

MERLIN MAIL Voice Messaging System and should be directed to the system operator during business hours.

■ The Automated Attendant Service can answer calls immediately

(Immediate Call Handling) or after a delay (Delayed Call Handling), for example, when a call remains unanswered by the system operator after a certain number of rings.

■ Programming is done via a touch-tone telephone. The MERLIN MAIL

Voice Messaging System is equipped with an RS-232 serial port and an external modem to support remote diagnostics.

■ Call restrictions should be assigned to the VMI ports that connect the

MERLIN MAIL Voice Messaging System to the system so that toll calls cannot be dialed through this application and so that MERLIN MAIL is not prohibited from outcalling.

MERLIN MAIL Voice Messaging System 4-11

Applications

■ The MERLIN MAIL Voice Messaging System cannot be used with

MERLIN Attendant.

Feature Interactions

C o v e r a g e

■ All extensions that need coverage are assigned to a coverage group through system programming The MERLIN MAIL Voice Messaging

System ports are assigned to a calling group designated as the coverage receiver for the coverage group.

■

■

■

An internal call on a VMI port that transfers to an internal extension will not go to coverage, but will continue to ring at that extension.

If a sender’s telephone is programmed so that only outside calls are sent to coverage, calls received on ICOM or System Access buttons are not sent to voice mail.

For Release 2.0, outside calls that would normally proceed to the

MERLIN MAIL Voice Messaging System as coverage do not do so if the telephone that sends the call to group coverage has activated Coverage

VMS. No special action is needed on MERLIN MAIL administration to activate this feature.

Group Calling

All VMI ports to which the MERLIN MAIL Voice Messaging System is connected are assigned to the same calling group through system programming.

Calls that overflow from one calling group to another calling group with integrated VMI ports are identified as coverage calls via mode codes.

As a result, the overflow calling group’s number appears in the calledparty field of the mode code.

For Release 2.0, when the MERLIN MAIL Voice Messaging System sends a Leave Word Calling message to an extension, the system identifies the voice mail system as the sender of the message. As a result, when the voice mail subscriber uses the Return Call feature, the call goes to any available voice mail port, not just to the specific port that generated the message. This improves access by reducing the chance of getting a busy port.

Leave Word Calling

■ Leave Word Calling sends mode codes to the MERLIN MAIL Voice

Messaging System to deposit a canned message if the target telephone does not have display capabilities.


Applications

Night Service

■ The MERLIN MAIL Voice Messaging System Automated Attendant

Service works with the Night Service feature to provide specialized afterhours service. The Automated Attendant can answer calls on lines it does not handle during business hours or can direct calls to ring at a specific night extension or department, such as Building Security. A special night announcement can greet after-hours callers.

Privacy

■ Privacy is automatically programmed for each VMI port connected to the

MERLIN MAIL Voice Messaging System.

Ringing Options

■ If lines set for answering by the Automated Attendant Service also appear on telephones other than the system operator console or a backup extension, they should be programmed for no ring.

Transfer

■ Integrated VMI ports can transfer an incoming call to an outgoing line/trunk.

■ If a call received on a line/trunk is transferred to a VMI port, the direct inside access mode code is sent. The call is treated as a transferred call, and the caller hears the internal greeting.

■ If a caller incorrectly specifies the answering VMI port as the desired transfer destination station, the VMI port may park the call.

■ Any calling group, calling group member, or station can be programmed to be a VMS transfer redirect extension. If the station is a Queued Call

Console (QCC), the transfer redirect call is delivered to the QCC as a returning call and is not placed in the QCC queue.

■ If a transferred caller gets no answer and returns to the system operator, the operator has no indication of the origin of the call.


■ Assign all the MERLIN MAIL Voice Messaging System ports to a calling group, set the group type to VMI Integrated, and set the hunt type to linear.

■ Program VMI loop-start ports for the MERLIN MAIL Voice Messaging

System for reliable disconnect.

● Specify the touch-tone duration and interval between digits in codes sent between the MERLIN MAIL Voice Messaging System and the communications system.


Applications

■ Specify the VMS Transfer Return Interval, that is, the number of rings before a call transferred by the MERLIN MAIL Voice Messaging System is sent to the backup position (system operator).

■ Set Inside (intercom) Dial Tone to outside.

■ Assign call restrictions to each VMI port used to connect the MERLIN

MAIL Voice Messaging System.

■ When the Automated Attendant is used only for Night Service:

—

If the lines/trunks set for answering by the Automated Attendant

Service appear at other extensions, set the no ring option for the other extensions.

— Specify immediate answer (one ring) for the VMI ports.

— Specify the VMS calling group as the Night Service operator

Platform Requirements

The following equipment is required to connect the MERLIN MAIL Voice

Messaging System to the system:

■ MERLIN MAIL Voice Messaging System unit and power cords

■ Remote maintenance device (a modem with a wall-mounted transformer)

■ Modem cable with a 9-pin connector at one end and a 25-pin connector at the other, for connecting the modem to the serial pod on the MERLIN

MAIL Voice Messaging System unit

■ D4BU modular cords (two for a two-port system or four for a four-port system, plus one for the modem)

■ 012 module (a ring generator is required)

NOTE:

Additional TTRs may be needed to allow the 012 module to handle a large number of voice connections.

If the MERLIN MAIL Voice Messaging System is to be used only for backup call handling or night service, only one VMI port may be required. For other uses with heavier call traffic, the number of VMI ports required depends on the number of incoming lines/trunks, the number of subscribers programmed for

Automated Attendant service, and the number of busy-hour calls. Table 4-4

shows these requirements.


Applications

Table 4-4. MERLIN MAIL Voice Messaging System Ports Required

No. of No. of

VMI Ports Incoming

Required Lines/Trunks

Subscribers or

Busy-hour Calls

2

4

1 to 6

7 to 18

1 to 20

21 to 60


Applications

MERLIN Attendant

CAUTION:


The MERLIN Attendant is a standalone application that answers incoming calls and plays a menu of recorded prompts. A caller can respond to the prompts by dialing touch-tones, and the MERLIN Attendant routes the call to an internal extension accordingly. A caller without a touch-tone telephone is transferred to the system operator for further call handling and routing.

The MERLIN Attendant can be programmed to transfer calls in either of two ways:

■ Unsupervised transfer— the MERLIN Attendant dials the extension or department requested by the caller and disconnects. If the call is not answered or the extension is busy, the call is routed to the system operator or goes to the redirect extension.

■ Supervised transfer— the MERLIN Attendant transfers the call and can retrieve it if the transfer is unsuccessful. The MERLIN Attendant then directs the call to another telephone, allows the caller a second route choice, or plays a failed-transfer announcement, depending on how the application is programmed.

Calls can be answered immediately (Primary Call Handling) or after a delay

(Secondary Call Handling), for example, if the call goes unanswered by the system operator after a specified number of rings.


The system must operate in Hybrid/PBX or Key mode. The MERLIN Attendant cannot be connected to a system operating in Behind Switch mode.


■ The MERLIN Attendant cannot be connected to a communications system that has an AUDIX Voice Power or MERLIN MAIL Application installed.

■ A maximum of four MERLIN Attendants can be connected to the system.

■ The MERLIN Attendant can be programmed to answer every incoming call or in or only calls on certain lines/trunks.

■ Unanswered transferred calls do not return to the MERLIN Attendant, but are redirected to a designated extension, such as the system operator,

4-16 MERLIN Attendant

Applications

■ If the extension called is busy or unanswered, or after business hours, calls can be directed to an answering machine to allow callers to leave messages.

■ The MERLIN Attendant can transfer calls to fax machines, if the fax extension number is specified and the caller dials it.

■ The MERLIN Attendant provides 64 seconds for recording up to five standard messages, including the caller greetings used during and after business hours, a hold announcement for a caller who is being transferred, a connect announcement for the department or extension receiving a transferred call, and an announcement explaining that a call cannot be completed.

■ When the MERLIN Attendant is set up for after-hours operation, the time on its clock must match the system clock.


Coverage

■ An internal call on a VMI port that transfers to an internal extension will not go to coverage, but will continue to ring at the internal extension.

■ For Release 2,0, outside calls that would normally proceed to the

MERLIN Attendant as coverage do not do so if the telephone that sends the call to group coverage has activated Coverage VMS. No special action is needed on MERLIN Attendant administration to activate this feature.

Group Calling

■ All MERLIN Attendants connected to the system must be assigned to the same calling group through system programming.

Night Service

■ The MERLIN Attendant works with the communications system’s Night

Service feature to provide specialized after-hours service. The MERLIN

Attendant can answer calls on lines it does not handle during business hours or can direct calls to ring at a specific night extension or department, such as Building Security. A special night announcement can greet after-hours callers.

Privacy

■ Privacy must be programmed for each VMI port connected to the

MERLIN Attendant.

MERLIN Attendant 4-17

Applications

Transfer


■ Calls on generic VMI ports cannot be transferred to telephones that have

Remote Call Forward activated.


■ Assign all the MERLIN Attendant ports to a calling group and set the group type to VMI Generic.

■ Set Inside (intercom) Dial Tone to outside.

■ Designate a transfer redirect extension, such as the system operator, to receive calls that were originally transferred to unanswered or busy extensions, or to receive calls when a caller fails to respond to the announcement.

■ Program all calling groups as Auto-logout. (Auto-logout is the default.)

■ Assign Privacy to each VMI port used to connect the MERLIN Attendant.


The following equipment is required to connect the MERLIN Attendant to the system:

■ MERLIN Attendant unit

■ 6-wire modular telephone cord

■ 012 module (a ring generator is required)

NOTE:

Additional TTRs may be needed to allow the 012 module to handle a large number of voice connections.

The number of MERLIN Attendants required depends on the number of incoming lines/trunks and the number of busy-hour calls. One is normally

sufficient for handling after-hours calls only and for delayed call handling. Table

4-5 shows these requirements when the MERLIN Attendant is programmed for

Primary Call Handling.

4-18 MERLIN Attendant

Applications

Table 4-5. MERLIN Attendants Required

No. of

Attendants

Required

Incoming Busy-hour

Lines/Trunks calls

2 1 to 6 1 to 25

3

4

7 to 9

10 to 12

25 to 50

50 to 100

MERLIN Attendant 4-19

Applications

Call Accounting

System (CAS)

CAUTION:


CAS is a software application for businesses that need to manage telephone usage and control costs by tracking, sorting, and recovering telephone charges. CAS provides a menu-driven user interface and on-line help.

There are three versions of CAS:

CAS integrated with IS Ill

CAS Plus V3 for general business use, a standalone application that runs on an approved AT&T DOS personal computer (PC)

CAS/Hospitality (CAS/H) for hotels and health care facilities, a standalone application that runs on an approved AT&T DOS PC

All three versions allow businesses to calculate the cost of calls by using the rates charged by long-distance carriers in one of 11 major metropolitan areas.

In addition, CAS Plus V3 and CAS—IS Ill can be customized by programming additional rate tables.

CAS Plus V3, CAS/H, and CAS—IS Ill provide the following services and features:

■ Call Record Processing— records of calls are collected and stored, and costs are calculated using the rate table selected. The system can be programmed to process all calls or only calls that exceed a specified cost threshold. It can also add a service charge to calls before billing them to clients, departments, projects, or (with CAS/H) rooms.

In addition, CAS-IS III collects and processes automatic number identification (ANI) information (caller identification), and can provide detailed information on incoming calls by point of origin. However, the availability of this information may be limited, depending on the legal jurisdiction and the equipment at the telephone company central office

(CO) serving the caller.

■ Report Generation— stored call record information can be organized and printed in the following kinds of reports:

—

Summary reports provide consolidated information on call activity. A wide variety of summary reports is available, based on all the types of data available about the application: for example, by department, by extension, by area code, by cost, by time of day, or by trunk facility used.

4-20 Call Accounting System (CAS)

Applications

— Detail reports provide detailed, call-by-call information for each extension or (with CAS/H) by room.

— Selection reports organize information on the basis of userspecified criteria, allowing trends and problems to be highlighted.

— Account Code Detail Report lists every call associated with each account code entered by users.

— Facility and Cost Center Reports show the distribution of line/trunk usage over organizations or cost centers.

— Preselected Reports provide a choice of up to five reports from any of the other report categories and can be set to print on demand or at a specified time and date.

■ System Management— the system manager can perform a variety of customization and maintenance activities, such as editing tables, setting up reports, and keeping call rate information up to date.

■ Directory Lookup and Message Center— callers can look up anyone in the organization by name or extension, leave a message, and print or display messages.


■ Only one CAS can be connected to the system.

■ The system does not provide Station Message Detail Recording (SMDR) for calls within the system.

■ The number of calls about which CAS can store information depends on the amount of available disk space. In its largest configuration, CAS records data for up to 5,000 extensions and 15,000 account codes.

■ When an industry-standard T/R device, such as a fax machine with a built-in telephone, is attached to an MLX telephone via a Multi-Function

Module (MFM), the connected T/R device cannot enter an account code,

When the T/R device is connected to an analog multiline telephone via a

GPA, the same is true. However, an Account Code Entry button can be programmed in the telephone. A user can then place a call from the telephone, enter an account code, and then go off-hook on the associated T/R device. Or a user might place a call from the built-in telephone on the fax machine, press the Account Code Entry button on the analog multiline telephone, enter the account code, and press the

Account Code Entry button again. The account code is then captured along with the other call information.

Call Accounting System (CAS) 4-21

Applications


Account Code Entry

■ CAS uses the account codes entered by users before or during calls to provide reports by account code.

SMDR

■ CAS collects call information from the SMDR output of the system


The following equipment is required to connect Standalone CAS software (CAS

Plus V3 or CAS/H) to the system:

■ An AT&T 286/386 PC, configured as follows:

— MS-DOS 3.1 or higher

— 64K RAM

— 20 MB hard disk

— 1 parallel port

— 2 serial ports

— CGA/VGA/EGA, Super VGA, or Hercules monochrome monitor

— Real-time clock card

— 3.5-or 5.25-inch floppy disk drive

■ 132-or 80-column lBM-compatible graphics parallel printer

■ D8W modular cord and 355AF adapter connecting the SMDR port on the system to the COM1 serial port on the PC (CAS Plus V3 only; CAS-IS Ill connects to the COM2 port).

4-22 Call Accounting System (CAS)

Applications

Call Accounting Terminal (CAT)

CAUTION:

This section is intended solely as an overview of the application, For comprehensive information about [he use of the application, see the documentation for the product.

CAT, a standalone application, is a dedicated terminal and printer designed to track, sort, and print reports on telephone charges.

Three versions of CAT are available:

■ CAT Basic is an entry-level system for small businesses.

■ CAT Plus/Business, for larger businesses, includes a two-line display.

■ CAT Plus/Hospitality, for hotels and health care facilities, also includes a two-line display.

CAT can be set up to calculate the cost of calls by using toll rates or charging by the minute. Service charges and discounts can be applied to calls made to local and long-distance numbers and to directory assistance, Calls to specified area codes (such as 900) can be singled out for special treatment,

CAT is customized with current local and long-distance rates for a company’s location. As rates change or a new area code or exchange is added, the rate information can be updated simply by exchanging a chip inside the terminal.

When a new telephone line or account code is added to the system, the CAT adds this information to its memory automatically the first time the new line or code is used.

The CAT provides a variety of reports that can be printed on a regular schedule or automatically when stored call information reaches 90% of the terminal’s capacity. The available reports include the following, depending on the version of CAT:

■ A variety of summary and detail reports. For example, reports can be printed on all extensions or rooms, a single extension or room, account codes, time of day, duration, and trunk facility.

■ Management analyses organize call information by time of day, cost and duration of calls, area codes and exchanges called, and trunk facilities.

CAT can receive and process ANI information from the SMDR. The system gets such information from the AT&T Megacom 800 service and puts it into the

SMDR.

CAT features an LCD display instead of a printed menu.

Call Accounting Terminal (CAT) 4-23

Applications

NOTE:

The availability of the caller identification may be limited by local-serving

(caller’s) jurisdiction, availability, or telephone company equipment.


■ Only one CAT can be connected to the system.

■ CAT Basic can store information on up to 1200 calls for 100 extensions and 49 lines.

■ CAT Plus/Business can store information on 6500 calls made from up to 200 telephones that share up to 49 lines. When 90% of this capacity

(5040 calls) is reached, When 5850 of these calls have been processed, reports are printed and memory is cleared. Any calls that come in during this process are held until reports are printed again.


Set SMDR options:

■ Select basic or PRI call report format.

■ Specify the minimum call length to be recorded (10 seconds is recommended).

■ Specify whether information is to be recorded for both incoming and outgoing calls or only for outgoing calls.



■ CAT uses the 9-digit account codes users enter before or during calls to associate calls with accounts and individuals; these codes appear on

CAT reports.

SMDR

■ CAT collects call information from the SMDR output of the system.


The following equipment is required to connect the CAT to the system:

■ One of the following CAT units:

4-24 Call Accounting Terminal (CAT)

Applications

— CAT Basic

— CAT Plus/Business

— CAT Plus/Hospitality

■ CAT Printer

■ D8W modular cord and 355AF adapter connecting the SMDR port on the control unit to the CAT

Call Accounting Terminal (CAT) 4-25

Applications

Call Management System (CMS)

CAUTION:


CMS is a standalone DOS-based application that simulates the actions of a system operator by answering calls and distributing them to individual agent extensions. If no agents are available, CMS puts calls on hold and, if programmed, plays a recorded announcement to the callers. When agents become available, CMS searches the system for the appropriate agent—usually the one who has been idle the longest—and transfers the call to that person’s extension.

CMS is appropriate for businesses with large groups of personnel who perform a common function, such as airline ticketing, filling catalog orders, or providing customer service. Agents within these groups can be divided into splits, or subgroups, to handle different kinds of calls or customers. For example, the agents in a travel agency might be divided into three splits: one that handles personal vacations, one that handles business trips, and one that handles group charters. Another split can be designated to provide support when call traffic is particularly heavy in the other splits. Calls come in to each split on a group of lines designated to ring into that split.

Agents make themselves available and unavailable to take calls by logging in and out. In addition, agents can enter the After-Call-Work (ACW) state, which allows them to complete work on their last call without being interrupted by new

CMS calls. The system can be setup so that agents are automatically in the

ACW state whenever they complete a CMS call or so that they must dial a feature code or press a programmed button to enter ACW.

CMS provides the following additional features:

Management reports that analyze call volume and patterns and agent activity. Summary reports can span from 1 to 93 days.

The Answer Delay option, which determines how long a call rings before it is designated as unanswered and connects to the recorded announcement.

The Forced Delay option, which connects all calls to the recorded delay announcement regardless of whether all agents are busy.

Designation of priority lines to ensure that calls coming in on those lines are answered first.

Display of current agent activity on system status screens to allow monitoring, tracking, and analyzing of short- and long-term performance.

4-26 Call Management System (CMS)

Applications

■ Ability to connect Music-on-Hold to callers waiting for available agents.

■ Ability to connect up to four external alerts to indicate an exception, for example, an LED that lights when the oldest call has waited longer than

30 seconds. Exception thresholds are programmed.

■ Real-time dynamic reconfiguration, allowing the user to modify the call flow on-line.


The system must operate in Hybrid/PBX or Key mode. CMS cannot be connected to a system operating in Behind Switch mode.


A maximum of two CMSs can be connected to the system.

CMS must be installed on an approved AT&T DOS PC. The PC must be dedicated to CMS. The two CMS interface card ports on the PC must be connected to two analog multiline extension jacks on the same module in the control unit (an 008 or 408). These jacks must be system operator positions. If two system operator position jacks are not available on the same module, another of these modules must be installed in the control unit to provide them.

Each CMS can handle calls for up to 28 agents on up to 28 lines, and it can answer calls on two lines at the same time with the same announcement.

Up to six agent splits can be designated for each CMS, with 28 agents per split.

The CMS supervisor’s console is any Direct-Line Console (DLC). CMS agents can have any MLX telephone or any analog multiline telephone that can be used with the system. CMS agent telephones must be connected to the first 58 extension jacks on the control unit.

Lines/trunks ringing in to CMS can be loop-start, ground-start, T1 emulated ground-start, or PRI.

Up to four external alerts can be used to alert agents and supervisors when the number of calls waiting to be answered reaches the programmed threshold.

A MERLIN Attendant can be used to direct callers to the appropriate

CMS group by use of Ioop-arounds.

To play music for waiting callers, a Music-on-Hold product must be used that is compatible with a Music-on-Hold coupler.

Call Management System (CMS) 4-27

Applications

NOTE:

If such equipment is used to rebroadcast music or other copyrighted materials, it may be necessary to obtain a copyright license from and pay license fees to a third party, such as ASCAP or BMI. A Magic

On Hold system does not require such a license.


Extension Status

■ A CMS supervisor uses the Extension Status feature to control and monitor when agents are in the available, unavailable, or ACW state. A

CMS agent does not have to be a member of a calling group to be available or unavailable. The system can be programmed for CMS or for

Hotel/Motel Extension Status, but not for both.

Group Calling

■ CMS agents log in and out by using the same buttons or codes as calling group members.

System Numbering

■ CMS agent telephones can use any extension. However, CMS refers to telephones using the 2-digit default numbering plan.


■ Set basic system operating conditions:

— Select the 2-digit System Renumbering plan (2-digit is the factory setting) or set-up space, with CMS agents numbered for two digits.

— Set Transfer Return Time for 3 to 5 rings.

— Set Transfer Audible to Ringback.

— Select the Group Calling/CMS option for the Extension Status feature.

■ Remove CMS lines from all telephones (Key mode only) or from trunk pools (Hybrid/PBX mode only).

■ Set up three DLC system operator positions—two for CMS PC positions and one for the CMS supervisor position (if a CMS supervisor telephone is required):

4-28 Call Management System (CMS)

Applications

— Assign the positions.

— Assign CMS lines and external alerts to the CMS supervisor’s console, and copy the assignments to the CMS PC ports.

■ Set up a CMS fallback plan:

— Designate the CMS supervisor console as a Group Coverage sender.

— Assign the agent telephones to a calling group and assign

Group Coverage to the calling group.

■ Set up optional equipment and features, including headsets and paging groups.

■ Set the ringing options for lines assigned to CMS ports to No Ring.


The following equipment is required to connect CMS to the system:

An approved AT&T DOS PC, configured as follows:

— 640 kbytes RAM

— 3.5-inch floppy disk drive

— 20-Mbyte hard disk drive

— Monochrome or color monitor

CMS interface card with two 14-foot (430-cm), 4-pair modular extension cords

CMS software for the system

DA-5 Digital Voice Announcement Unit with one 14-foot (430-cm) DIN connector cord parallel printer with cable to connect to the PC parallel port

Supervisor console—any DLC position

Agent telephones—any MLX or analog multiline telephones supported by the system

One analog multiline module (008 or 408) to connect the two PC ports to the extension jacks assigned as DLC ports.

Call Management System (CMS) 4-29

Applications

InnManager Guest Management

System

CAUTION:


The InnManager Guest Management System is a standalone application that provides a complete management package for hotels with up to 100 extensions.

It is a turnkey system, consisting of bundled hardware and the following menudriven software modules. The system manager can customize each of these modules to suit the needs of the business.

■ Call Accounting and Rating System (CARS) provides detailed call record accounting and control over call billing.

■ Guest Management System provides front desk functions for management and tracking of room reservations, occupancy, and billing; hotel management functions such as housekeeping and bookkeeping; and general office utilities such as form letters, spreadsheet export, calculation and tracking of travel agency commissions, and credit limit reports.


■ If the PC is more than 50 feet (15 meters) from the control unit or the PC does not share the same AC power supply as the control unit, ADUs must be used.

■ If the single-user version of the InnManager Guest Management System is run on a network with multiple terminals, the data files will be corrupted and unusable.


■ An approved AT&T DOS PC with at least a 40-Megabyte fixed disk, one

3 1/2-inch floppy disk drive, and at least 640K of available memory.

■ MS-DOS 3.3 or higher

■ 80-column printer capable of printing in elongated, bold-face, underline, and condensed modes.

■ Internal or external modem

■ A 355AF adapter and cords to connect the PC to the control unit.

4-30 InnManager Guest Management System

Applications

System Programming and

Maintenance (SPM)

CAUTION:


SPM is a software application used for programming and maintenance of the system. It performs the same functions as an MLX-20L telephone used as a system programming console, providing a display that emulates the console display. SPM also has additional features, such as the ability to back up and restore system programming and to print reports.

Two versions of SPM are available:

■ SPM standalone, running on an approved AT&T DOS PC

■ SPM Integrated with IS Ill

A PC with DOS-based SPM can be connected directly to the control unit or can access the system remotely in one of the following ways:

■ The system programmer dials the system directly. A password can be set up to prevent unauthorized access.

■ The system programmer dials the system operator and asks to be transferred to the system’s built-in modem (Dial Code *IO).

SPM—IS Ill can be used only through a direct local connection,

SPM can be programmed to operate in English, French, or Spanish for communication with the control unit. Independent of this language setting, an on-screen option allows the programmer to select from the same three languages for the console-simulation window only for the duration of the current session.


■ SPM must be upgraded to version 2.

XX to function with Release 2.0 of the communications system.

■ Unless the system is being backed up or restored, a remote SPM connection takes priority over a local user. If the local user is programming when a remote user connects to the system, the system sends a warning message to the local user and disconnects him or her.

■ A PC running DOS-based SPM connects to the lower RS-232 jack on the processor module of the control unit. This connection runs at 1200 or

2400 bps with autobaud.

System Programming and Maintenance (SPM) 4-31

Applications

A UNIX* system-based version of SPM is available. See IS II and IS Ill.

SPM reports can be printed out or can be written to the PC’s hard or floppy disk drive. At the same time, the report is displayed on the screen together with prompts for browsing.

SPM reports should not be printed while the system is handling more than 100 calls per hour.

A printer connected to the computer running SPM can be used to print system programming reports. Reports can also be sent to a printer connected to the SMDR port on the control unit. However, SMDR information may be lost while system programming reports are being printed through the SMDR port.


Standalone SPM requires an approved AT&T DOS PC, configured as follows:

MS-DOS 3.3 or higher.

At least 128 kbytes of RAM.

A double-sided floppy diskette drive, either 5¼-inch or 3½-inch. (A hard disk is optional, but recommended.)

A serial port assigned to COM1 or COM2. The serial port can use either a DB-9 or DB-25 connector. If a DB-9 connector is used, a 9-pin to

25-pin adapter is also required. The 9-pin side must be female.

A monochrome or color monitor.

A D8W modular cord and a 355AF modular adapter if the PC is less than

50 feet (15 meters) from the control unit. Distances of greater than 50 feet (15 meters) require back-to-back ADUs.

4-32 System Programming and Maintenance (SPM)

Applications

Integrated Solution II (IS II)

CAUTION:


Integrated Solution II (IS II) is a complete package of UNIX System-based voice processing and call analysis software applications. IS II offers a single interface to any of the following applications:

■ Integrated Voice Power Automated Attendant (IVP AA) answers telephones automatically and transfers callers to the appropriate departments or extensions. Callers are provided with a menu of recorded prompts that they respond to by dialing numbers on a touchtone telephone.

Callers without touch-tone telephones can be transferred to the system operator, who then handles their calls. Separate menus for day and night service as well as multilevel menus and corresponding announcements can be set up to ensure that callers reach the right person or department as quickly as possible.

IVP AA can operate in touch-tone gate mode or in no-gate mode. To speed handling of calls from touch-tone telephones, gate mode prompts callers to dial 1 to continue to the main menu. If a 1 is not dialed within a programmed interval, calls are automatically transferred to the system operator. In the no-gate mode, callers hear the main menu immediately and, if no response is received after the main menu is played, calls are transferred to the system operator.

IVP AA is a low-cost alternative for businesses that need enhanced call handling without the added voice messaging capabilities of AUDIX Voice

Power—lS II.

■ AUDIX Voice Power—lS II (AVP) offers all the features of the IVP AA combined with the following services:

— Call Answer Service, which allows callers who reach a busy or unanswered extension to leave a message, transfer to another extension, or transfer to a system operator. Individual extension users can program a personal greeting or select a standard greeting; users can also program a password to prevent others from retrieving their messages.

— Voice Mail Service, which allows users to send messages to other extensions in the system, forward messages received with comments, and reply to messages received. The system manager can send general messages to everyone in the system.

Integrated Solution II (IS II) 4-33

Applications

— Information Service, which provides a customer-oriented, callin information service that plays a recorded message and then disconnects the caller.

— Message Drop, which offers an answering service, similar to an answering machine, that plays a message to the caller and then allows the caller to “drop off” a message, such as a request for service or an order. Callers cannot direct their messages to specific extensions.

Call Accounting System—lS II (CAS) collects and analyzes call information, calculates the prices of calls by using rates selected by the business, organizes calls by client or project, and prints reports on a daily or as-needed basis. CAS—IS II provides all of the functionality of

CAS along with ANI. For more information on the features of CAS, see

Call Accounting System in this chapter.

System Programming and Maintenance-lS II (SPM) is a programming package built into IS II that allows the system manager or a systems technician to upgrade and maintain the system and its features and to add, change, or rearrange telephones. Programming can be done on site or remotely.

Additional IS II features include the following:

Dial by Name permits AVP users to call subscribers by dialing the last name of the subscriber instead of dialing the extension number.

Alternate Personal Greetings allows a user to record a second personal greeting in addition to the primary call-answer greeting.

Fax Transfer directs incoming fax calls to a designated fax machine.

Class of Service allows the system manager to assign one of 16 predefine parameters to a subscriber. These parameters define the size of the mailbox, the type of coverage service, and the activation of the outcalling feature.

General Mailbox Options are two special mailboxes that have reserve extensions associated with them. Callers using rotary telephones or needing assistance can be transferred to leave messages in a general mailbox. Subscribers having problems with the system can report problems to the trouble mailbox.

The number of incoming lines and subscribers programmed for AVP or IVP AA and the number of busy-hour calls determine how many voice channels are

required for the user’s system. See Table 4-6.

4-34 Integrated Solution II (IS II)

Applications

Table 4-6. Voice Channels Required

No. of Channels

Required

2

8

12

4

6

Lines

1 to 6

7 to 18

19 to 24

25 to 42

Subscribers Busy-Hour Calls

1 to 20 1 to 20

21 to 60

61 to 80

81 to 200

Over 42 201 to 300

21 to 60

61 to 80

81 to 200

201 to 300

Mode

Differences

Only the CAS—IS II and the SPM—IS II applications can be connected to a system that operates in the Behind Switch mode.


IS II uses UNIX System V Release 3.2.2.

IS II stores up to 12 hours of voice-mail messages when IS II includes

AVP and over 200,000 call accounting records when IS II includes CAS.

Either IVP AA or AVP can be installed on the system, but not both.

The system supports up to 12 IVP AA ports (on three circuit boards).

If IS II includes AVP, when users receive voice-mail messages, the message LEDs on their telephones turn on, provided that a mailbox has been assigned to each of those telephones.

For AVP or IVP AA, the following symptoms indicate that the system needs more TTRs:

— Single-line telephone users do not get dial tone when trying to dial out.

— AVP or IVP AA fails to transfer calls.

— Calls fail to ring, or calls go to coverage prematurely.

SPM—IS II reports can be printed out or written to a disk (floppy or fixed drive). At the same time, the report is displayed on the screen together with prompts for browsing.

SPM—IS II reports should not be printed while the system is handling more than 100 calls per hour.


Applications



■ The account code entered by users before or during calls are used by

CAS-IS II to associate calls with accounts and individuals; they appear on CAS-IS II reports.

Coverage

■ An internal call on a VMI port that transfers to an internal extension will not go to coverage. It will continue to ring at the internal extension.

■ If a sender programs his or her telephone to that only outside calls are sent to coverage, calls received in ICOM or System Access buttons will not be sent to voice mail.

■ For Release 2.0, outside calls that would normally proceed to AUDIX

Voice Power as coverage do not do so if the telephone that sends the call to group coverage has activated Coverage VMS. No special action is needed on AUDIX Voice Power administration to activate this feature.

Group Calling

■ Calls answered by an overflow calling group will get coverage mode codes; the overflow calling group’s number appears in the Called Party field of the mode code.

■ For Release 2.0, when AUDIX Voice Power sends a Leave Word Calling message to an extension, the system identifies the voice mail system as the sender of the message. As a result, when the voice mail subscriber uses the Return Call feature, the call goes to any available voice mail port, not just to the specific port that generated the message. This improves access by reducing the chance of getting a busy port.

Leave Word Calling

■ If a Leave Word Calling message is left in a mailbox in a system with heavy VMI traffic, the user may have to dial out manually for messages.

Night Service

■ If the AVP Automated Attendant handles only after-hours calls, a phantom station (an unused telephone jack) must be programmed as a member of a Night Service group associated with the system operator.

In turn, this phantom station is covered by a calling group with integrated

VMI ports as members. If an incoming call is not answered in the programmed number of rings, the control unit sends the call to the calling group with the VMI ports. Because of prior programming, AVP recognizes the call to be from the phantom station and provides

Automated Attendant service rather than the usual Call Answer service.


Applications

SMDR

■ CAS-IS II uses the call information provided by the system’s built-in

SMDR feature to process calls. There are two system formats for

SMDR—basic and PRI.

Transfer

VMI ports can transfer an incoming call to an outgoing line/trunk.

If a caller incorrectly specifies the answering VMI port as the desired transfer destination telephone, the VMI port can inadvertently park the call.

Any calling group, calling group member, or telephone can be programmed to be a voice messaging system (VMS) transfer redirect extension. If a QCC is programmed as such, the transfer redirect call is delivered to the QCC as a returning call and is not placed in the QCC queue.

If a transferred caller gets no answer and returns via voice mail to the system operator, the system operator has no indication of the origin of the call.


The following must be programmed when IS II includes IVP AA:

■ Designate Inside (Intercom) Dial Tone to be the same as the outside line/trunk dial tone.

■ Assign all Automated Attendants connected to the system to the same calling group and set the group type to VMI Generic.

■ Program each VMI loop-start port for reliable far-end disconnect.

■ Designate a backup position, such as the system operator, to receive calls that were originally transferred to unanswered or busy extensions or when a caller fails to respond to a message.

■ Specify the number of rings before a call transferred by the voice messaging system is sent to the backup position.

The following must be programmed when IS II includes AVP:

■ Assign AVP ports to a calling group and specify the group type as VMI

Integrated.


■ Specify the touch-tone duration and interval between digits in codes sent between the AVP and the system.


Applications

■ Specify the number of rings before a call transferred by the AVP is sent to the backup position (system operator).

■ When the AVP Automated Attendant is used for Night Service only, do the following:

—

—

If the lines/trunks set for answering the Automated Attendant appear at other stations, set the No Ring option for the other telephones.

Assign the phantom station to a Night Service group for each system operator position.

Assign the phantom station to a coverage group, and assign the

VMI calling group to cover that coverage group.

— Specify the VMI ports that provide Automated Attendant to be

Automated Attendant ports.

— Specify the business schedule for AVP.


■ An 80-Mbyte or 200-Mbyte fixed disk drive is required if IS II includes either IVP AA or AVP.

■ IS II uses an AT&T Master Controller based on a 6386/SX WGS processor with UNIX System V/386 Release 3.2,2, which includes the following:

Master Controller II processor (with a 40-Mbyte, 80-Mbyte, or

200-Mbyte fixed disk and a 3.5-inch floppy disk drive)

—

Video monitor (monochrome or color)

— Keyboard

— Optional tape drive (required for systems with a 200-Mbyte fixed disk for saving UNIX files, application program files, administration files, and voice system files during backup)

■ A 355 AF adapter for connecting the Master Controller to the serial port on the control unit if they are within 50 feet (15 meters) of each other and are on the same AC branch circuit

■ ADUs for connecting the Master Controller to the serial port on the control unit if they are not within 50 feet (15 meters) of each other and are not on the same AC branch unit

■ Any additional hardware required by the individual applications included in IS II, including the cables and adapters for connecting the applications to the system. See the instruction booklet that comes with each application.


Applications

■ IVP4 boards

■ 012 basic telephone module to provide the tip/ring interface for IVP AA or

AVP


Applications

Integrated Solution III (IS III)

CAUTION:


IS Ill is a complete package of UNIX System-based voice processing and call management software applications. It provides a single integrated interface to any of the following applications:

■ AUDIX Voice Power 2.1.1 (AVP) combines the following voice messaging services:

— Call Answer Service, which allows callers who reach a busy or unanswered extension to leave a message, transfer to another extension, or transfer to a system operator. Individual subscribers can program a personal greeting or select a standard greeting and can program a password to prevent others from retrieving their messages.

Voice Mail Service, which allows subscribers to send messages to other extensions in the system, forward messages received with comments, and reply to messages received. The system manager can broadcast messages to all subscribers.

—

Information Service, which provides a call-in information service that plays a recorded message and then disconnects the caller.

— Message Drop, which provides an answering service, similar to an answering machine, that plays a message to callers and then allows a caller to “drop off” a message, such as a request for service or an order. Callers cannot direct their messages to specific extensions.

— Automated Attendant Service, which answers incoming calls and plays a menu of recorded prompts. A caller can respond to the prompts by dialing touch-tones, and the Automated

Attendant routes the call to an internal extension accordingly. If there is no answer or the extension is busy, the caller can be given the option to leave a message or try another extension.

A caller without a touch-tone telephone is transferred to the system operator for further call handling and routing.

The system manager can record multiple levels of menus and announcements, including separate menus for day and night service.

4-40 Integrated Solution III (IS Ill)

Applications

— With Outcalling, when a user or subscriber receives a new message, the system can automatically call a number that he or she has programmed, for example, a beeper or a home telephone number. The subscriber can then log in to the VMS to retrieve messages.

■ Fax Attendant provides an integrated voice/fax mailbox, fax broadcasting, fax bulletin board, and coverage for busy or off-line fax machines, It must be installed with AUDIX Voice Power. Fax Attendant includes the following services:

— Fax Call Answer, which allows Fax Attendant to receive fax messages for subscribers whose fax machines are busy or out of paper. This feature also enables subscribers who have personal fax numbers but do not have fax machines to receive fax messages. In such a case, Fax Call Answer gives the appearance of a personal fax machine by automatically answering and receiving fax messages for the specified phone number.

— Fax Mail, which allows subscribers to send fax messages, get fax messages, record personal greetings, administer outcalling

(standalone configuration only, create fax distribution lists, and change their account passwords, delivery report settings, and autoprint setting.

— Fax Response, which allows the user to dedicate a phone number from which callers can retrieve information. This feature directs callers through a series of prompts to retrieve information on their fax machines. Callers are greeted with spoken prompts that guide them in pressing touch-tone buttons to access the information and to receive their information within minutes by fax transmission.

■ Call Accounting System collects and analyzes call record information, calculates costs using rate tables selected by the customer, organizes calls by client or project, and prints reports daily or as needed,

CAS-IS Ill provides the same functionality as the standalone CAS

Plus V3 application, and in addition collects and processes ANI information (if available from the originating CO).

■ System Programming and Maintenance provides a maintenance and programming interface to the system. SPM—IS Ill provides the same functionality as the standalone SPM application, except for remote connection to the control unit.

In addition to these applications, IS Ill provides Extension Directory services.

Extension Directory allows integrated programming of extension and subscriber information for both AUDIX Voice Power and the system from a single interface.

This integration eliminates duplication of effort. When a change is made to either database, the two databases are automatically reconciled, ensuring that they remain in agreement.

Integrated Solution Ill (IS Ill) 4-41

Applications

Integrated Administration

Integrated Administration is the integration of AUDIX Voice Power and Fax

Attendant administration with the switch parameters that are used by those two applications. Integrated Administration consists of three cooperating parts accessed by menu selection:

■ Extension Directory

■ Extension Directory Setup

■ Integrated AUDIX Voice Power and Fax Attendant Administration

(System Programming/Switch Administration)

Integrated Administration is intended to be used primarily by system technicians who are responsible for administering the applications and the switch through

IS Ill. Users can also use Integrated Administration to make changes to their system. All three parts of Integrated Administration are available to system technicians; only the Extension Directory and System Programming/Switch

Administration are available to users via the IS Ill menu.

Initial Installation

The system technician performing initial installation of Integrated Administration logs in to IS Ill and selects System programming and Maintenance

(SPM) from the menu, The SPM screen appears. The technician performs basic system administration, such as dial plan, mode, attendant, phantom stations, and lines in pools, for the switch and exits from SPM.

Through the Technician Maintenance menu item on the Integrated Solution

Ill Maintenance menu, the technician selects AUDIX Voice Power Switch

Defaults and changes the defaults for Calling Group if necessary.

Through the same menu item, the technician performs an Extension Directory

Setup, which downloads the switch dial plan and directory labels into the

Extension Directory Database.

Through the Integrated Solution Ill Maintenance menu, the technician performs an Extension Directory update, that is, steps through each extension and attaches a name label and other information.

The technician then selects System Programming/Switch Admin from the

AVP or AVP/FA main menu and performs Integrated AVP or AVP/FA

Administration. (The technician is guided through a series of choices and forms that direct the administration. ) The technician then administers the remaining

AVP and FA administration, and performs the rest of switch administration.

Installation on Existing Switch

The system technician installing Integrated Administration on an existing switch, that is, a switch that is already functioning, selects AUDIX Voice Power

Switch Defaults from alll the Technician Maintenance item on the

Integrated Solution Ill Maintenance menu, and changes the defaults for calling Group if necessary. Through the same menu item, the technician performs an Extension Directory Setup. This downloads the switch dial plan and directory labels into the Extension Directory Database.

4-42 Integrated Solution Ill (IS Ill)

Applications

Through the Integrated Solution Ill Maintenance menu, the technician performs an Extension Directory update, that is, steps through each extension and adds them as AVP subscribers if necessary.

The technician then selects System Programming/ Switch Admin from the

AVP or AVP/FA main menu and performs Integrated AVP or AVP/FA

Administration. (The technician is guided through a series of choices and forms that direct the administration.) During the administration, the technician presses the Save key (without entering any information) whenever prompted to enter lines/pools, This allows the flow of information to proceed without sending any outside calls directly to the AVP ports.

The technician completes the remaining AVP and FA administration, including administration of greetings and other voice prompts, and through the System

Programming/Switch Admin menu, goes back to each installed service and adds the appropriate lines.

Remote Operation

Remote initial installation can be provided by equipping the remote location with a surrogate switch and IS Ill. Using the remote location’s switch and Master

Controller II+ or Ill, the technician programs the customer’s configuration, as specified earlier in Initial Installation.

Through SPM, the technician backs up the switch configuration and, through the Technician Maintenance menu, backs up the Extension Directory Database files.

The technician then dials up the customer location and accesses the internal modem, and, via remote SPM, restores the customer’s switch from the translations made at the remote location.

After requesting Pass Through to the customer’s Master Controller, the technician, through the Technician Maintenance menu, restores the customer’s database files from the database files backed up at the remote location.

NOTE:

The Extension Directory and Integrated Administration screens can be accessed remotely, but the information is stored in a file and run after the remote caller hangs up, Also, a change made to System Renumbering is not reflected immediately in the Extension Directory; reconciliation is run automatically at 3:00 a.m. or can be invoked manually via the technician menus. This reconciliation synchronizes the information in the database and the switch.


■ The system must operate in Hybrid/PBX or Key mode for all IS Ill applications except CAS and SPM. Those are the only two applications that can be connected to a system operating in Behind Switch mode.

Integrated Solution Ill (IS Ill) 4-43

Applications


■ IS Ill can store up to 36 hours of voice-mail messages for AUDIX Voice

Power with the 500-Mbyte fixed disk.

IS III can store over 200,000 call records for CAS with the200-Mbyte fixed disk.

Fax Attendant cannot be installed without AUDIX Voice Power.

■

Automated Attendant cannot be installed as a standalone application, but only in conjunction with AUDIX Voice Power.

When an AUDIX Voice Power subscriber receives a Voice Mail message, the message-waiting LED is lit on his or her telephone.

If an AUDIX Voice Power mailbox is needed for a person with no telephone, a phantom station must be assigned in the system switch.

AUDIX Voice Power time should be synchronized with the time on the system switch.

Updating of Message Waiting lights on users’ telephones is most efficiently obtained by properly linking users’ AVP mailboxes with telephones in the system switch. If an AVP mailbox is desired for a person who does not have a telephone, a phantom extension must be assigned in the system switch.

Integrated Administration is not supported for Standalone Automated

Attendant.

For Integrated Administration, the Master Controller II+ or III provides a separate Backup and Restore capability that saves the Directory information on the fixed disk. This is provided through the Backup and

Restore menu options under Maintenance.

Assigning the AVP application to a user via the Integrated Administration

Extension Directory automatically sets that user to be covered by AUDIX

Voice Power, if specified, and produces the AVP Subscriber Screen.

The AVP Subscriber menu uses the name, extension, and coverage information contained in the Integrated Administration Extension

Directory. This information cannot be changed on the AVP Subscriber

Screen.

No AUDIX Voice Power subscriber who does not have AVP in the

Applications field of the Integrated Administration Extension Directory can be added to or deleted from AUDIX Voice Power.

All stations and lines being programmed via Integrated Administration must be idle.


Applications


Account Code

Entry

■ CAS uses the account codes users enter before or during calls to associate calls with accounts and individuals; these codes appear on

CAS reports.

Coverage

■ An internal call on a VMI port that transfers to an internal extension will not go to coverage, but will continue to ring at the internal extension,

■ If a sender programs his or her telephone so that only outside calls are sent to coverage, calls received on ICOM or System Access buttons will not be sent to voice mail,

■ For Release 2.0, outside calls that would normally proceed to AUDIX

Voice Power as coverage do not do so if the telephone that sends the call to group coverage has activated Coverage VMS. No special action is needed on AUDIX Voice Power administration to activate this feature.

Group Calling

■ Calls that overflow from one calling group to another calling group with integrated VMI ports are identified as coverage calls via mode codes.

As a result, the overflow calling group’s number appears in the calledparty field of the mode code.

■ For Release 2.0, when AUDIX Voice Power sends a Leave Word Calling message to an extension, the system identifies the voice mail system as the sender of the message. As a result, when the voice mail subscriber uses the Return Call feature, the call goes to any available voice mail port, not just to the specific port that generated the message. This improves access by reducing the chance of getting a busy port.

Labeling

■ Names entered via the Integrated Administration Extension Directory are sent to the switch and are available through Switch Labeling screens.

Leave Word

Calling

■ If a Leave Word Calling message is left in a mailbox in a system with heavy VMI traffic, the subscriber may have to dial out manually to retrieve the message.

Integrated Solution Ill (IS Ill) 4-45

Applications

Night Service

■ If the Automated Attendant handles only after-hours calls, a phantom station (an unused station jack) must be programmed as a member of a

Night Service group associated with a system operator. In turn, this phantom station is covered by a calling group with integrated VMI ports as members. If an incoming call is not answered within the programmed number of rings, the control unit sends it to the calling group with the VMI ports. AUDIX Voice Power must be programmed to recognize the call to be from the phantom station, and provides Automated Attendant service rather than the usual Call Answer service.

SMDR

■ CAS collects call information from the SMDR output of the system.

System Renumbering

■ System Renumbering can be done only via SPM or MLX-20L system programming. Integrated Administration uses System Renumbering to read extension numbers and adjuncts.

Transfer

■ Integrated VMI ports can transfer an incoming call to an outgoing line/trunk.


■ Any calling group, calling group member, or extension can be programmed to be a VMS transfer redirect extension. If a QCC is so programmed, the transfer redirect call is delivered to the QCC as a returning call and is not placed in the QCC queue.

■ If a transferred caller gets no answer and returns via Voice Mail to the system operator, the system operator has no indication of the origin of the call.


The following system programming is required for AUDIX Voice Power:

■ Assign AUDIX Voice Power ports to a calling group and specify the group type as VMI Integrated.


■ Specify the touch-tone duration and interval between digits in codes sent between AUDIX Voice Power and the system.

■ Specify the number of rings before a call transferred by AUDIX Voice

Power is sent to the backup position (system operator).


Applications

The following system programming is required for AUDIX Voice Power with

Automated Attendant:

■ Set Inside (Intercom) Dial Tone to outside.

■ Assign Automated Attendants to a calling group and specify the group type as VMI Integrated.

■ Designate a backup position, such as the system operator, to receive calls that were originally transferred to unanswered or busy extensions or when a caller fails to respond to the announcement.

■ Specify the number of rings before a call transferred by the VMS is sent to the backup position.

When the AUDIX Voice Power Automated Attendant is used only for Night

Service:

■ Set the no ring option for the other extensions if the lines/trunks set for answering by the Automated Attendant Service appear at other extensions.

■ Specify the VMI ports that provide Automated Attendant Service to be

Automated Attendant ports.

The following system programming is required for Integrated Administration with

Fax Attendant:

IVP 4/6 Board Ports. For Fax Response service, the following items must be programmed:

Put the tip/ring ports dedicated to Fax Response into a calling group for

Integrated Administration.

■ Set the calling group type to VMI Integrated—Automatic.

■ Assign outside lines to the calling group.

■ Give the lines appropriate labels.

Fax Board Ports. Each fax port on the TR112 or TR114 board is connected to a tip/ring port on the switch. These ports are regular tip/ring ports and do not have to be identified as fax ports on the communications system. The following items must be programmed:

■ Give the tip/ring connections appropriate labels.

■ Put the tip/ring extensions on the Night Service Exclusion list to enable

Off-Site Fax Delivery to function at night.

■ Identify the ports as fax ports on the switch using SPM if the user wants to use the Fax Message Waiting Light feature.

Integrated Solution III (IS Ill) 4-47

Applications

Private Fax Extensions. A Private Fax Extension is either an extension connected to an actual fax machine used by an individual or a phantom station associated with an individual’s voice extension. Programming for Private Fax

Extension depends on whether or not the communications system’s configuration supports DID lines. In systems with DID, unique DID extension numbers are sufficient for Private Fax Extension because outside calls placed to that DID number ring the fax machine or phantom station. Systems without DID must rely on personal line appearance.

The following item must be programmed for Private Fax Extensions in DID configurations:

■ Assign the DID extension of a phantom station or actual fax machine as a Private Fax Extension.

The following items must be programmed for Private Fax Extensions in non-13113 configurations:

■ Assign a personal line to a phantom station or to the extension connected to an actual fax machine.

■ Assign the phantom station or fax machine as the owner of that line using

SPM.

The following remaining items must be programmed for Private Fax Extensions in both configurations:

■ Set the individual as a subscriber to Fax Attendant service. Specifying

AUDIX Voice Power as an application automatically subscribes the user to both AUDIX Voice Power and Fax Attendant.

■ Assign the extension of the phantom station or fax machine as a Private

Fax Extension.

■ Place the Private Fax Extension in a coverage group.

■ Set the label of the Private Fax Extension appropriately.

■ Assign the specified coverage group previously to be covered by the calling group of Automated Attendant, Call Answer, or Voice Mail—

Automatic.

One Private Fax Extension can also be used by a group of individuals through parameters set using Fax Attendant setup screens on the Master Controller. To facilitate these configurations, a Group Fax Administrator is selected. The following items must be programmed:

■ Assign a non-valid extension number as a Special Purpose extension.

This will be the Group Fax Administrator.

■ Assign a Private Fax Extension to the Special Purpose extension.

■ Set group members as Fax Attendant subscribers. Do not program any of these users for Private Fax Extension.


Applications


IS Ill is delivered already installed and configured with the applications ordered.

The system consists of an AT&T Master Controller II+ or Master Controller Ill running UNIX System V Release 3.2.2. Various hardware configurations are available; see the AT& T Integrated Solution Ill Installation and Maintenance

Guide for details.

If AUDIX Voice Power is installed, an 012 module (with a ring generator) is required in the system to provide the tip/ring interface.

The number of voice channels required for AUDIX Voice Power depends on the number of incoming lines/trunks, the number of subscribers programmed for the system, and the number of busy-hour calls. Table 4-7 shows these requirements,

Table 4-7, Voice Channels Required

No. of

Channels

Required Lines

2 1 to 6

4

6

8

12

Busy-hour

Subscribers Calls

1 to 20 1 to 20

7 to 18

19 to 24

21 to 60

61 to 80

21 to 60

61 to 80

25 to 42 81 to 200 81 to 200

Over 42 201 to 300 201 to 300

Integrated Solution Ill (IS Ill) 4-49

Applications

Primary Rate Interface (PRI)

Applications

CAUTION:


Group IV (G4) Fax

Group IV (G4) Fax is an application that enables the system to use the advanced Group IV (G4) Fax equipment-one of the new services accessible with the PRI trunks. Group IV (G4) Fax equipment provides several advantages:

■ High speed transmission

■ High quality laser reproductions

■ High speed, high capacity printing

■ Virtually error-free transmission

■ Fax machine can double as an office copy machine.

Documents received using Group IV (G4) Fax equipment are virtually perfect reproductions of the original document. Therefore, any company involved in graphic media (such as detailed engineering or architectural drawings or advertising graphic layouts) are ideal candidates for this application.

Depending on the interface the fax machine has, the Group IV (G4) application can be connected in three configurations:

■ Direct RS-232 (the recommended method)

■ V.35 interface connecting to a UDM RS-232D interface

■ V.35 interface connecting to a 7500B Data Module

Each of these configurations requires additional equipment.

See Chapter 5, Data, for additional information about Group IV (G4) Fax.

Video Conferencing

Video Conferencing, available with PRI service, enables groups of people in geographically dispersed locations to meet face to face. They may exchange information, documents, ideas, and data while employing a variety of visual aids to support the exchange of this information. Visual aids can include interactive writing and drawing, prepared text and graphic materials, and precorded audio and motion video material. Improved technology and superior camera optics and digital audio signals result in video pictures that are equal to commercial broadcast quality

4-50 Primary Rate Interface (PRI) Applications

Applications

Video conferences can be started from an easy-to-use control console and can be as easy to use as a telephone. No special technical expertise is required to operate the system.

The basic components include

The conference control subsystem—to establish and terminate connections and allows camera control including pan, tilt, and zoom; and audio control for volume and privacy.

The video subsystem—includes a full-motion video camera, a video monitor, and various video switching circuits, May also include an auxiliary room camera, a document camera, and a video cassette player/recorder.

The audio subsystem—allows video participants to hear and speak at the same time. Includes microphones, microphone mixer, an echo canceller. Microphones may include table top microphones, wired lapel microphones, and/or wireless hand-held or lapel microphones.

The video codec subsystem—a signal processing computer that digitizes, merges, and compresses audio and video signal input from the camera and microphone mixer for transmission to the far-end conference unit.

These basic components can be integrated into a mobile roll-about console that can be easily wheeled into a conference room or executive office prior to a scheduled video conference call. Alternatively, the components can be built-in to a video conference room.

Additional equipment may be required, for example, an interface converter for the video codec.

Optional applications can be added to the basic components to enhance the information being sent back and forth during an interactive broadcast.

See Chapter 5, Data, for additional information about Video Conferencing.

Primary Rate Interface (PRI) Applications 4-51

Applications

Centrex Operation

CAUTION


Centrex is an optional telephone service for business customers. It provides an array of telephone features from the CO that formerly were available only from a

PBX located on the customer’s premises.

Basic Centrex features include the following:

■ Transfer

■ Three-Way Conference

Drop

Hold

Recall

Call Forwarding

Call Waiting

Call Pickup

Group Pickup

Automatic Callback

Additional features such as Speed Dialing and Night Service can also be added.

The system can be used with either full or limited Centrex service. Full Centrex service requires that telephones have a Centrex line and that the users depend primarily on Centrex features for their communications needs. Full Centrex can also be used when only some telephones have direct lines, while others share lines or have no direct line assigned. Limited Centrex service is for customers who use the system features for most of their communications needs.

Timed Flash

In Releases 1.0 and 1.1, a timed flash can be generated on a Direct Facility

Termination (DFT) button or a Direct Pool Termination (DPT) button only, as long as the call is not a conference call and the facility is a loop-start line. A timed flash cannot be generated for an external call that terminates on a System

Access button.

4-52 Centrex Operation

Applications

In Release 2.0, a timed flash can be generated on a call terminating on a

System Access button, as well as on a DFT button or a DPT button. This includes, among others, transferred calls, group calling calls, and forwarded calls.

The following apply to timed flash (Release 2.0):

■ Dial Access to Pools: If the Recall button is pressed during dialing while connected to a trunk or when end-of-dial is reached, a timed flash is generated, the accessed trunk is kept, and restrictions are applied.

■ Automatic Route Selection (ARS): While an ARS call is being dialed, a timed flash cannot be generated. When dialing is complete, pressing the Recall button generates a timed flash, the accessed trunk is kept, and restrictions are applied,

■ Rotary trunks: A timed flash cannot be generated during dialing. When dialing is completed, pressing the Recall button generates a timed flash, the accessed trunk is kept, and restrictions are applied.

Full Centrex Service

With full Centrex service, each telephone has a direct line to the CO Centrex.

The direct line allows users to dial outside numbers directly after dialing an access code (usually a 9). The direct line is also used to call other four-digit

Centrex extension numbers. System intercom lines are used to dial other telephones on the system.

Users with full Centrex service can send a switchhook flash via the Recall button without the system intercepting or responding to the signals,

A full Centrex service requires that the planning form for each MLX telephone using a direct line be marked for central administration of a Recall button (code

*775).

Limited Centrex Service

Limited Centrex service is for customers who will primarily use the system features, but wish to retain access to the network or other Centrex locations by use of a limited number of lines.

With limited Centrex service, some telephones have direct Centrex lines, while others do not. Some telephones may be assigned POTS, Tie, or DID lines.

Others will use System Access buttons to access pooled facilities. Generally, users will rely heavily on the features of the system.

In a limited Centrex configuration, the system provides the primary connection to the CO, serving as a “local” switch between the telephones and the CO, A switchhook flash, feature access code, or feature button signal is interpreted by the system to be a system command, not a Centrex command.

Centrex Operation 4-53

Applications


The system must be configured for Behind Switch mode for full Centrex operation.

For limited Centrex operation, if Centrex features are dominant, the system should be configured for Behind Switch mode; if system features are dominant, the system should be configured for Key or Hybrid/PBX mode.

The system can be used as a Hybrid/PBX behind a host switch by combining the features of a Behind Switch system with the ground-start capabilities of a

Hybrid/PBX system. If a ground-start line is connected directly to the control unit, the FCC considers the system a Hybrid. Any Behind Switch system with full or limited Centrex service using a ground-start line must be registered as a

PBX system even if it operates in the Key mode.


■ With full Centrex service, during periods of high traffic, users may experience delays in obtaining dial tone from the host. Should a user begin dialing too rapidly, the first and second digits could be lost and the call would be misdialed. This situation is more probable when the host is another PBX, not a CO Centrex. With full Centrex service, the delay in dial tone could cause misdialing when using System Speed Dialing or

Personal Speed Dialing.

■ With full Centrex service, dependence on loop-start lines during a hightraffic period can cause a glare condition when calls grab the same line simultaneously. The loop-start lines normally used in Centrex service do not protect against glare.

■ With full Centrex service, some Centrex features require 2-or 3-digit codes for access. These must be obtained from the telephone company and provided to the customer at installation.

■ With full Centrex service, loop-start lines have higher cable losses than ground-start lines and cannot assure secure toll restrictions.

■ With full Centrex service, single-line telephones have limited functionality when connected directly to the CO Centrex host. They cannot access system features or make inside calls. The can, however, use all of the

Centrex features via code numbers.

If the single-line telephone has the idle line preference programmed for in intercom ring line, it cannot be used for internal conference, transfer, or drop because it would result in a Centrex dial tone.

■ With limited Centrex service, touching a hard feature button to call a local feature will cause misdialed calls.

■ If the limited Centrex configuration is programmed for Hybrid/PBX mode, the use of calling groups, Shared System Access buttons, pools, and other features is possible, and applications such as MERLIN MAIL, CAS, and CMS can be used. If the mode is Behind Switch, the applications cannot be used.

4-54 Centrex Operation

Applications

■ If a Key system is being converted to a Hybrid/PBX system for limited

Centrex service, the conversion from loop-start lines to ground-start lines can cause a long delay before the system can be installed.

■ Extension numbers should reflect the last four digits of the Centrex telephone line number. A brief ring delay occurs when calling a Centrex or PBX host extension number because the call is being processed through two systems. No delay occurs when making a system intercom call.

Centrex Operation 4-55

Applications

MERLIN PFC Telephone

CAUTION:


The MERLIN PFC (Phone-Fax-Copier) Telephone is a BIS-34D (34-button) display telephone with a built-in fax machine and personal copier that provides the convenience of a fax machine and personal copier in one compact unit.

Using the MERLIN PFC Telephone allows the user to do the following:

■ Make and receive inside and outside calls using the built-in speakerphone as well as use the BIS-34D telephone features provided by the system

■

■

Send and receive fax transmissions while using the telephone

Make quick copies while using the telephone

The system must have two analog ports available on the control unit. In Behind

Switch mode, a dedicated fax line for incoming fax calls is also required; in

Hybrid/PBX or Key mode, the system can have either a dedicated fax line or direct inward dialing (DID).

NOTE:

The fax machine component of the MERLIN PFC Telephone does not transmit date, time, and fax number.


Hybrid/PBX and Key Modes

■ The dedicated fax line for incoming fax calls from the CO must be connected to a line port on the control unit, and the line cannot be assigned to any pool.

■ If DID is used, a DID number must be assigned to the fax extension.

■ If a dedicated private line is used, assign a fax line to the voice station.

■ No lines or line pools can be administered to the fax extension.

■ The dedicated fax line should be administered to Immediate Ring and any other lines to No Ring at the fax extension.

4-56 MERLIN PFC Telephone

Applications

Behind Switch Mode

■ The dedicated fax line can be administered only to the MERLIN PFC

Telephone fax extension,

■ The dedicated fax line cannot be assigned to a pool.

■ The dedicated fax line should be assigned as the secondary line on the

MERLIN PFC Telephone.


■ The MERLIN PFC Telephone requires two analog ports: one for the voice line and one for the fax line.

■ The telephone wiring between the system controller or control unit and the MERLIN PFC Telephone must be installed in the same building.

■ The MERLIN PFC Telephone cannot be installed outside of building.

■ All button assignments except the one for the fax line must be removed from the fax extension.

■ The Voice Announcement feature should be removed from the fax extension.


Ringing Options

■ If the dedicated fax line is shared for outgoing calls only, the Ringing

Option must be administered to No Ring at any station except the

MERLIN PFC Telephone fax extension.

MERLIN PFC Telephone 4-57

Applications

Automated Document Delivery

System (ADDS)

CAUTION:


The AT&T Automated Document Delivery System (ADDS) is a computer-based system that stores documents in a database and automatically faxes them on request 24 hours a day.

This type of fax application is called fax response or fax-on-demand. ADDS has one voice port for handling incoming calls and one fax port for fax delivery of documents, Using a touch-tone telephone, a caller accesses the system and is guided by prompts through the process of selecting a document and indicating the fax number to which the information is to be sent. The caller then receives the requested information in minutes by fax transmission.

Callers may be required to enter a password to gain access to ADDS. Access to system administration also requires the use of a password.

The application can be configured to allow callers to request more than one document per call. Also, callers can leave any message or no message after requesting a document.

A record is maintained by ADDS, including the file name of documents the system has transmitted or attempted to transmit, the phone number of the destination fax machine, the time and date of the transmission attempt, and whether the transmission succeeded or failed.


■ Using one line for fax transmission limits ADDS to approximately 100 calls per day. Businesses anticipating more than 100 calls per day may need more than one system to handle the call volume efficiently.

■ ADDS should be used in a two-line configuration to maximize performance and minimize busy signals.

■ ADDS can be used behind an Automated Attendant.

■ ADDS does not function using a BTMI, GPA, or a tip/ring adapter.

4-58 Automated Document Delivery System (ADDS)

Applications


To set up ADDS, a business must have the following:

■ The Automated Document Delivery System software

■ A touch-tone telephone

■ A Group Ill (G3) Fax machine with an integrated handset

To request and receive information, a caller must have a touch-tone telephone and a Group Ill (G3) Fax machine.

For backup of stored data, one of the following is required:

■ AT&T 705 MT Multi-tasking Terminal

■ AT&T 6386/SX WGS (or compatible) with ProComm Plus software

Automated Document Delivery System (ADDS) 4-59

Applications

CONVERSANT Intro

CAUTION:


CONVERSANT Intro is an entry-level voice response system that enables the user to run integrated voice response (IVR) applications. CONVERSANT Intro can automatically answer and route calls and execute telephone transactions.

The CONVERSANT Intro software can be configured in either of the following ways:

■ As an application development environment in which all the tools to create an application are available

■ As a platform to run applications already developed

CONVERSANT Intro consists of the hardware and software that supports transaction processing, data retrieval, and data entry using a touch-tone telephone connected to a public telephone network. When a telephone connection is made to CONVERSANT Intro, the application running on

CONVERSANT Intro prompts the caller with synthesized voice in an application-dependent dialogue. The caller enters the appropriate responses by using the touch-tone keys on the telephone. This interaction continues until the caller ends the call.

Applications can be developed that allow the call to be transferred to an attendant telephone during some part of the dialogue. Calls also can be transferred to an attendant telephone automatically if the application determines that an attendant is required. CONVERSANT Intro also supports scripts that allow callers to record and play back information.

CONVERSANT Intro offers the following capabilities:

Customized inbound call management or call routing

Functions that are performed by choosing options in windows displayed on the screen

Multiple script configuration possibilities that allow for different paths within the same script for handling calls during normal business hours, after hours, and on holidays

Simple prompt recording using a telephone

Optional seasonal greetings to be played during set time intervals

Repeatable prompts

4-60 CONVERSANT Intro

Applications

■ Interaction of applications with voice mailboxes, with the ability to leave and retrieve messages, execute voice mail script, or get subscriber information

■ Creation of tables and retrieve and update data using database tables

■ Logging and displaying error messages

■ Management reports and a system monitor for monitoring daily and ongoing system progress


■ CONVERSANT Intro supports a maximum of 24 channels of analog ports, or up to 6 IVP4 boards. In a co-resident environment, such as

CONVERSANT Intro and AUDIX Voice Power, the system supports a maximum of 16 channels. The number of channels assigned to AUDIX

Voice Power can never exceed 12.


The platform for’ CONVERSANT Intro is the Master ControIler Ill, a highperformance 32-bit computer built around an Intel® 486® SX microprocessor.

It has 8 MB of random-access memory (RAM) and a 500 MB fixed disk drive.

The system unit has a 250-MB tape drive and a 3.5-inch floppy disk drive. Two serial ports and one parallel port are integrated on the main board with connectors on the back panel of the system unit. A diskette drive controller and fixed disk drive interface also are integrated on the main board. A Video

Graphics Array (VGA) video display controller and a tape drive controller are provided on separate add-in boards. Six additional Extended Industry Standard

Architecture (EISA) slots are available for other input/output (I/O) cards.

The Master Controller Ill uses AT&T UNIX System V version 3.2.2, It includes a system unit, a monitor, and a keyboard.

CONVERSANT Intro 4-61

Applications

Applications Printers

The following table shows the printers that can be used with the communications system for applications connected to the system.

Table 4-8. Applications Printers

Printer

Applications Printer

Applications Printer

(Wide Carriage)

Call Accounting

Terminal

(CAT) Printer

Document No.

582-421-105

582-421-106

582-421-100

Description

9-pin dot matrix printer that provides choice of print quality and speed. Uses parallel connection to the computer.

9-pin dot matrix printer that provides choice of print quality and speed. Has wide carriage that accommodates pinfeed paper up to

14 7/8 inches (37.8

cm) wide. Uses parallel connection to the computer.

9-pin dot matrix printer that provides choice of print quality and speed. Uses serial connection to the computer.

4-62 Applications Printers


Using its circuit-switched connections, the communications system can establish a dedicated communications path between two data endpoints for the transfer of data. This connectivity enables the communication system to be used to share resources as well as to establish and manage connections between computers and other data input and output devices. In addition, the communications system can support advanced network services that integrate voice and data, such as Video Conferencing. The communications system’s features used for voice service, such as Automatic Route Selection (ARS), Call

Restriction, Idle Line Preference and others, can be used to enhance the usage of data facilities.

This chapter describes the system’s data communications capabilities, the configurations and features that support those capabilities, and typical data communications applications. For instructions on making and answering data calls, see the Data User’s Guide.

Data Communications Support 5-1

Data Communications Support

Data Communications Configuration

Overview

Figure 5-1 shows how data and voice equipment connects to the

communications system to provide the support for data communications. The communications system’s control unit (hardware and software), in conjunction with other external hardware devices, provides data connectivity for the following:

Internal analog data stations

Connection to external analog data stations via analog facilities (GS, LS,

Tie, DID) or via dedicated analog facilities, or via a DS1 Digital Service

Link (DSL) providing emulated GS, LS, Tie, DID, or PRI

Internal digital data stations

Connection to external digital data stations via a Primary Rate Interface

(PRI) facility

Circuit-switched connections between two similar type data stations

Circuit-switched connection between a digital data station (on a Bchannel) and an analog data station via two-stage dialing through a conversion resource (modem pool)

Conversion resources, such as 7500B 7500B Data Module/modem pools for making connections between Analog-to-Digital and Digital-to-Analog data stations

Data Hunt Groups (DHGs)

Simultaneous Voice and Data on analog and MLX ports

On-premises host computer access

Local area network (LAN) access via a modem or 75009 Data Module connected to an RS-232 port on a workstation on the LAN

This section describes the various equipment configurations and connectivity

arrangements shown in Figure 5-1.

5-2 Data Communications Configuration Overview


Figure 5-1. Individual Use Data Station Configurations

Data Communications Configuration Overview 5-3


Data Stations

A data station consists of data terminal equipment (DTE), such as a personal computer (PC), data terminal, printer, optical scanner or video system, and data communications equipment (DCE), such as an internal or external modem or an external 7500B Data Module.

The DTE connects to the communications system via the DCE, which has capabilities similar to a telephone. The DCE places the data call, maintains its connection, and terminates the data call.

The DCE and DTE may have hardware and/or software options that can be set for transferring and receiving data, such as parity and bit rate. Other options may differ, depending on whether the transfer of data is synchronous or asynchronous. See the DTE and DCE documentation for configuration compatibility requirements, the options for transferring and receiving data, and guidelines for changing options.

Data stations (PC-based or workstation-based) require a communications software or terminal emulation package to transfer and receive data. The communications setup for each data station depends on specific configuration requirements and equipment limitations. Reference the communications

or

terminal emulation software documentation for instructions.

Data stations can be either analog or digital. Some analog data stations and all digital data stations can include a telephone for users who need simultaneous voice and data transmission.

Analog Data Station

An analog data station uses a modem as its DCE. The modem converts digital signals from the DTE into analog signals. It transmits these signals as continuously varying electrical voltages in the voice frequency band. It converts incoming analog signals into digital signals, passing them to the DTE. Most

types of modems can be connected to the system. (See Chapter 2 for modem

hardware descriptions. )

From the standpoint of the communications switching system, the analog data endpoint for an analog data station is the Tip/Ring (T/R) interface provided by one of the following:

■ A General Purpose Adapter (GPA), connected to an analog multiline telephone

■ A port on a T/R module (012T/R or 008 OPT)

■ A Multi-Function Module (MFM) adjunct on an MLX telephone

From a physical viewpoint, the connected modem provides a T/R analog signal interface to the telephone network or switching system and an EIA RS-232 digital interface (or other type) to the data equipment.

5-4

Data Communications Configuration Overview


Analog Data Station Configurations

A

S

Figure 5-1 shows, the T/R interfaces support the following analog data

stations:

■ Analog Voice and Analog Data (Station A): Includes a data terminal

(with a keyboard and display) or a computer connected to a modem via an ElA-type RS-232 interface. The GPA connects the modem to the analog multiline telephone to provide the T/R interface for the modem.

To provide the simultaneous voice and data, the port configuration requires two adjacent odd/even station jacks on a 408, 408 GS/LS, or

008 module in the control unit. The even jack is for voice and the odd jack is for data. The bridging adapter joins the odd/even jack pair for connection to the analog multiline telephone. The telephone provides the dialing capability for the data station.

■ Off-Premises Analog Data-Only (Station B): Includes a data terminal

(with a keyboard and display) or a computer connected to a modem via an ElA-type RS-232 interface. The modem connects to a port on a

008

OPT module in the control unit. A telephone mayor may not be connected depending on modem capabilities; simultaneous voice and data is not supported, The terminal keyboard provides the dialing capability for the data station.

NOTE:

This distance between the station jack and the modem, as shown in

Figure 5-1, is acceptable for short distances up to 5000 feet ( 1524

meters). For longer distances, this configuration may need ‘{data grade” facilities with 4.5 dB maximum loss.

■ Analog Data-Only (Station C): Includes a data terminal (with a keyboard and display) or a computer connected to a modem via an EIA RS-232 interface. The modem connects to a port on an 012 module in the control unit. A telephone mayor may not be connected, depending on modem capabilities. Simultaneous voice and data transmissions not supported. The terminal keyboard provides the dialing capability for the data station.

■ Digital Voice and Analog Data (Station D): Includes a data terminal

(with a keyboard and display) or a computer connected to a modem via an RS-232 interface. The modem connects to an MLX telephone configured with an MFM. The MFM provides the T/R interface for the

modem. (See Chapter 2 for the hardware description of the MFM. ) The

MLX telephone voice capabilities operate independently from the data station data capabilities. The terminal keyboard provides the dialing capability for the data station.

Data Communications Configuration Overview 5-5


NOTE

If the MLX telephone is voice-signaled while active on a call, both Bchannels are required to accommodate the signaling. This may conflict with the data station.

Other Supported Analog Data Endpoints

Other data equipment that may be connected to an analog data endpoint (T/R interface) via a modem includes the following:

■ A local host computer (described later in this chapter)

■ Group Ill (G3) Fax terminal

■ An output-only device (optical scanner)

■ An input-only device (printer or display)

A personal computer containing an internal modem card can also be connected to a T/R interface.

NOTE:

A few restrictions or special conditions should be considered when configuring ports to serve a certain class of equipment or a special class of service. Port modules, station sets, and station adjuncts specifically designed for the communications system provide equipment type information to the system software automatically via the port signature or terminal/adjunct classmark. However, there are types of devices that can connect to a T/R interface that provide no classmark or other means of automatic identification to the system. This equipment includes Group III

(G3) FAX terminals, automatic answering machines, modems, automatic dial-announce alarm-sending equipment, external alerting device, music sources, and paging equipment,

Digital Data Station

The 7500B Data Module adapts the DTE, such as a PC or data terminal, to the

MLX environment. It is configured between the EIA type RS-232 interface to the data equipment and the MLX port on the control unit. The 7500B Data Module does not convert the digital signal from the DTE to an analog signal, but sends it

as a sequence of separate electrical impulses. (See Chapter 2 for the 75006

Data Module hardware specifications and requirements.)

Along with the feature descriptions in Chapter 2, the communications system

also supports the 7500B Data Module’s capability for circuit-switched data connections on the B-channel.



NOTE:

To communicate with the switch, the digital data endpoint uses D-channel messages during call setup and termination. When the call is set up, the switch establishes a connection between the calling and called endpoints on a B-channel. The switch sends the appropriate messages to drive the endpoints into the data mode. In the data mode, the 7500B Data Module transmits and receives data over a B-channel using transport modes as defined in the Digital Multiplexed Interface.

This capability uses data transport mode 3/2 adaptive or mode 2 only for asynchronous transmission, and modes 0, 1 and 2 for synchronous transmission. The transport modes are defined in the Digital Multiplexed

Interface (DMI), and the communications system does not interact with modes of data transport The 7500B Data Module can share the MLX port with an MLX telephone; however, the two units operate independently of each other.

NOTE:


The 7500B Data Module may provide dialing and answering capabilities to the data station. The terminal keyboard can also provide the dialing capability for the data station.

NOTE:

Although the 7500B Data Module supports packet-switched data on the Dchannel, this data mode is not supported by the communications system.

MLX Port Connection Requirements

When configuring digital data equipment connection to an MLX port, the following requirements and/or restrictions must be applied:

■ Only one 7500B Data Module should be connected to an MLX port.

When two 7500B Data Modules are connected, the system cannot address a specific 7500B Data Module for incoming calls. Although outgoing calls are not a problem, incoming calls may not always be answered by the intended party.

■ If a 7500B Data Module is the only digital data endpoint on the MLX port

(no MLX telephone is connected), a 440A4 Terminating Resistor (TR)

Adapter must be configured to provide 100-ohm termination for each transmission pair. The 7500B Data Module does not provide termination.

■ An MLX telephone is independent from the 7500B Data Module; however, the telephone may cause B-channel conflict between the telephone and the 7500B Data Module when it is voice signaled while active on a call. If a slight chance of data call blocking is unacceptable, an MLX telephone should not be connected to a 7500B Data Module used in a data station configuration.

Data Communications Configuration Overview 5-7


■ The maximum cord length from an MLX telephone to a the 7500B Data

Module is 80 feet (24 meters). This should be considered if there are plans to use the voice capability of a port by connecting an MLX telephone located some distance away from the 75006 Data Module.

Digital Data Station Configurations

As Figure 5-1 shows, the 7500B Data Module interface supports the following

digital data stations:

■ Digital Voice and Digital Data (Station E): Includes a data terminal (with a keyboard and display) or a computer connected to a 7500B Data

Module via an RS-232 or V.35 interface. The 7500B Data Module connects to a jack on a 008 MLX, 408 MLX, or 408 GS/LS-MLX (Release

2.0 only) module on the control unit. The MLX port is shared by the telephone and the 75006 Data Module but, they operate independently of each other. The terminal keyboard provides the dialing capability for the data station.

NOTE:

For a digital voice and digital data station, the MLX telephone cannot contain an MFM. The MFM interferes with communication to the switch in this data station configuration.


■ Digital Data-Only (Station F): Includes a data terminal (with a keyboard and display) or a computer connected to a 7500B Data Module via an

RS-232 or V.35 interface. The configuration does not include an MLX telephone, so a 440A4 terminating resistor is required. The 7500B Data

Module connects to a 008 MLX, 408 MLX, or 408 GS/LS-MLX (Release

2.0 only) module in the control unit. The keyboard provides the data call dialing capability.

NOTE:

There is no analog voice and digital data station configuration.

Data Station Configurations Summary

Table 5-1 summaries the hardware requirements and port assignments needed

to support the various data station configurations.



Table 5-1, Configurations of Data Stations

Equipment Configuration

Computer 7500B or Data Analog MLX

Type of Data Station Terminal Modem Module Telephone Telephone

Analog voice and analog data

✓ ✓ ✓ *

Type of

Module

008, 408,

408 GS/LS

Analog data only ✓ ✓ ✓

Digital voice and analog data

✓ ✓

‡

✓ †

012, 008 OPT

008

MLX, 408 MLX,

408 GS/LS-MLX

Digital voice and digital data

Digital data only

✓

✓

✓

✓

✓ §

008 MLX, 408 MLX,

408 GS/LS-MLX

008 MLX, 408 MLX,

408 GS/LS-MLX

* Requires a GPA

† Requires an MFM

‡ Telephone may be connected depending on modem capabilities.

§ Telephone cannot contain an MFM.

Other Supported Digital

Data Endpoints

Other data equipment that may be connected as a digital data endpoint via a

7500B Data Module includes the following:

■ Video Conferencing system

■ Group IV (G4) Fax machine

See PRI Applications later in this chapter.

Data

Hunt Group

A data hunt group (DHG) consists of a specified group of station ports of the

same type (i. e., all analog or all digital) assigned a Calling Group number,

When a call is placed to a DHG, the switch performs a circular search (starting with the station listed after the one that received the last call) to find the first idle station. The idle station is alerted and when it answers, it is connected to the originator. If all the stations in the DHG are busy, the originator hears ringback.

Data Communications Configuration Overview 5-9


DHGs support the following:

■ modem pools

■ dedicated lines for data service (for example, DHG #4 in Figure 5-1)

■ a host computer with multiple ports

■ a workstation (gateway) on a LAN

NOTE:

One important aspect of a modem pool is the specific direction of call origination. All the units (for example, modems) on one side of the modem pool can be grouped into a DHG, so users only need to dial one number for any pair.

Figure 5-1 shows the following examples of DHG assignments for the various

shared resources:

■ DHG #5 is assigned to the data modules used to communicate with the local host computer

■ DHG #6 is assigned to the modems used to communicate with the local host computer

■ DHG #7 is assigned to the 7500B Data Modules used to communicate with the workstation (gateway) on a LAN

■ DHG#8 is assigned to the modems used to communicate with the workstation (gateway) on a LAN

The communications system accommodates up to 32 calling groups (or DCG).

Each calling group can have a maximum of

20

members. A data station can be a member of only one calling group.

Modem Pool

A modem pool acts as a conversion resource to accommodate communication between analog and digital data endpoints. The modem pool can be one or more pairs of DCEs; a pair consists of one 7500B Data Module connected via an RS-232 interface to a modem.

NOTE:

It is possible for a communications system user to unintentionally or inadvertently corrupt the modem or 7500B Data Module settings by changing options at the modem/7500B Data Module interface. If a problem occurs with a modem pool, the settings on each device should be checked.

5-1o



Data Call Direction

An important aspect of modem pools is that call origination is in one direction only:

■ Analog-to-digital, which enables an analog data endpoint to originate a call to a digital data endpoint

■ Digital-to-analog, which enables a digital data endpoint to originate a call to an analog data endpoint

Using a DHG extension to access a modem pool ensures that the intended incoming service unit (modem or 7500B Data Module) receives the data call first. Once communication is established between the data endpoints, the communication is full duplex.

NOTE:

Modem pools should not be used to originate calls in both directions because it is possible for a modem/7500B Data Module pair to be called at both ends at the same time. This could result in connecting the wrong data endpoints or not completing either or both calls.

Modem Pool Dialing

Calls through a modem pool are placed using two-stage dialing. In the first stage, users enter the modem pool extension or DHG group extension, if available, for the analog-to-digital or digital-to-analog modem pool being called.

In the second stage, users enter the extension of the analog or digital data endpoint for inside calls. For outside calls, users enter the dial-out code and telephone number of the outside analog or digital data station.

This two-stage dialing should be considered before providing modem pools dialing access to outside lines.

For example, it is possible for a data call (originated from an external data endpoint) to come into a modem/7500B Data Module pair and, through second stage dialing, access another external data endpoint.

CAUTION:

Because the communications system originates the second data call in

the two stage dialing process, a call to an external data endpoint in the second stage would be charged to the communications system. If this

is a concern, separate analog-to-digital and digital-to-analog modem pools should be created that service incoming data calls but have no access to outside lines for second stage dialing.

Data Communications Configuration Overview 5-11


Modem Pool Configurations

Figure 5-1 shows three modem pool configurations:

1. Digital-to-analog modem pool, which enables a digital data endpoint to originate a call to an analog data endpoint. In this configuration, the

7500B Data Modules are assigned to a DHG.

2. Analog-to-digital modem pool, which enables an analog data endpoint to originate a call to a digital data endpoint. In this configuration, the modems are assigned to a DHG.

3. Digital-to-analog modem pool on dedicated outside lines, which converts digital signals to analog signals for data calls from a digital data station to an outside analog data station. It does this over analog dedicated outside lines used solely for data communications. In this configuration, the 7500B Data Modules are assigned to a DHG.

Configurations 1 and 2 (digital-to-analog and analog-to-digital) use system lines; therefore, the modems and 7500B Data Modules are connected directly to the control unit. Each modem is connected to a jack on a 012 or 008 OPT module, and each 7500B Data Module is connected to a jack on a 008 MLX, 408 MLX, or 408 GS/LS-MLX module (Release 2.0 only).

In configuration 3 (digital-to-analog on dedicated outside lines), the 75006 Data

Module connects to the control unit via jacks on a 008 MLX or 408 GS/LS-MLX module (Release 2.0 only).

In all three configurations, a 440A4 terminating resistor is required because the

7500B Data Module connects directly to the control unit. Also, each modem/7500B Data Module pair requires a null modem connector to interconnect the RS-232 interfaces of the modem and 7500B Data Module.

The 7500B Data Modules or modems in modem pools must not share a port with an MLX telephone. Otherwise, the telephone could use both B-channels and block use by the 7500B Data Module or modem.

Other Resource Pools

Users can share a limited number of modems or 7500B Data Modules connected to a shared data endpoint. To do this, the communications system can be configured for modem-only or 7500B Data Module-only resource pools.

The modem-only pool can provide access to multiport data equipment, such as a local host computer, by assigning the T/R ports interfacing with the modems in the pool to a DHG.

Like the modem-only pool, the 7500B Data Module-only pool can provide access to multiport data equipment, such as a host computer, by assigning the

MLX ports interfacing with the 7500B Data Modules in the pool to a DHG.

5-12



Connectivity to a Local Host Computer

Figure 5-1 shows DHGs assigned to modem-only and 7500B Data Module-only

modem pools that provide access to a local host computer.

These modem pools and the host computer are connected by EIA type RS-232 interfaces. Each modem provides the T/R interface to the 012 module on the control unit, and each 7500B Data Module connects to the 008 MLX or 408

GS/LS-MLX module (Release 2.0 only) on the control unit. Terminating resistors are required for 7500B Data Module connection.

Connectivity to a LAN

A LAN is an interconnected chain of terminals or PCS that pass data to and from a mainframe computer or interconnected workstations using some topology arrangement.

The communications system connects to the LAN through a workstation that functions as a gateway, The gateway provides the ports for the modem and data module connections to the communications system and the connection to the LAN. It also accommodates the protocols needed for the transfer of data between a data endpoint on communications system and a workstation (or data terminal) on the LAN.

The modems connect to the gateway via ElA-type RS-232 (or other type) interfaces and provide the T/R interface to the 012 module on the control unit.

The 7500B Data Modules connect to the gateway via ElA-type RS-232 interfaces (or other type) and on the communications system end connect to the

008 MLX, 408 MLX, or 408 GS/LS-MLX module (Release 2.0 only) on the control unit. Terminating resistors are required for the 7500B Data Module connection.

Once a connection is established between a data endpoint on the communications system side and a workstation within the LAN, all of the features and capabilities of the LAN environment are available to the originating data endpoint on the communications system side. However, limitations or hardware requirements may restrict the usage of some LAN facilities.

Outside Trunks

Figure 5-1 shows the types of outside trunks that can be used to make and

receive data calls to and from data stations outside the system. (In the figure, the trunk types are displayed to the left of the control unit and are labeled 1-4.)

1. Ground-start (GS) trunks and loop-start (LS) trunks are used to communicate with outside analog data stations. A loop-start trunk is the standard for home and small businesses is the least expensive trunks in some areas LS trunks have the following disadvantages:

Outside Trunks 5-13


■ They do not protect against glare, a condition that occurs when an outside call is made at the same time that an incoming call arrives on the same trunk.

■ They cannot provide reliable far-end disconnect for toll restriction.

A ground-start (GS) trunk is preferred for communication with outside analog data stations. Ground-start trunks provide improved signaling and reliable far-end disconnect for secure toll restriction.

The following kinds of outside ground-start/loop-start trunks can be used for data communications:

■ Basic trunks

■ Wide Area Telecommunications Service (WATS)

■ 800 service (inbound WATS)

■ Foreign exchange (FX)

Ground-start/loop-start trunks connect to ground-start/loop-start jacks on the following types of modules in the control unit:

■ 800 GS/LS

■ 400 GS/LS/TTR

■ 408 (LS trunks only)

■ 408 GS/LS



■ 408 GS/LS-MLX (Release 2.0 only)

2. A tie trunk provides communication between two telephone switching systems. A tie trunk “ties” the two systems together, providing access to all telephones or data stations on each system. Tie trunks are usually used for data communication with analog data stations connected to a system at a different location, such as a different floor of a building, a different building, or a different city or state.

A tie trunk connects to a jack on a 400EM module in the control unit.

3. A Direct Inward Dialing (DID) trunk allows incoming calls to reach specific individuals or facilities in the system without the help of a system operator. DID trunks are available only in the Hybrid/PBX mode. A DID trunk is used to receive incoming calls from outside analog data stations; it is not used for outgoing calls.

A DID trunk connects to a jack on an 800 DID module in the control unit.

4. A DS1 trunk carries digital signals in the Digital Signal 1 (DS1 ) format.

The DS1 format multiplexes 24 Digital Signal 0 (DS0) channels of 64 kbps each and one 8-kbps framing signal, for a total of 1.5444 Mbps. A DS1 trunk can be used for communication with outside digital or analog data stations.

5-14 Outside Trunks


A DS1 trunk connects to the jack on a 100D module in the control unit.

Even though there is only one physical jack, the 100D module supports up to 24 logical endpoints or ports for voice and data calls. Each DS0 channel in the DS1 signal corresponds to a trunk or logical ID.

A DS1 trunk provides either T1 or PRI access.

■ T1 is the factory setting.

A T1 facility is used for communication with outside analog data stations. The 24 channels on a T1 facility can be programmed individually in any combination to emulate a loop-start, groundstart, E&M tie, or DID trunk, so a single 100D module can replace

24 outside trunks. Digital data calls cannot be placed through this trunk.

■ PRI is the standard format provided by connection to a 5ESS central office (CO) switch or a 4ESS toll switch. (PRI must be used for digital data calls).

The 100D module supports any combination of the following AT&T

Switched Network services:

— Accunet switched digital service for 56-kbps and 64-kbps restricted and 64-kbps clear circuit-switched data calls

— Megacom 800 for incoming domestic toll-free voice calls

Megacom WATS service for outgoing domestic longdistance voice calls

— Software Defined Network (SDN) for circuit-switched voice and data calls at up to 56 kbps

PRI service provides the following benefits:

■ Speed. Data calls to outside destinations can be made on the same B-channels used for voice calls if the service allows.

Modems and dedicated, conditioned trunks are not required.

■ lNFO-2 automatic number identification (AN I) service. Customers who subscribe to this service can identify the caller on an incoming call on a PRI trunk by either telephone number or billing number.

NOTE

If your system has automatic number identification (AN I), the display shows the number for the outside caller. Availability of the caller identification information may be limited by localserving (caller’s) jurisdiction, availability, or

CO

equipment.

Outside Trunks 5-15


Dynamic B-channel assignment. An individual B-channel can be removed from service without blocking calls to or from any other

B-channels.

Improved toll restriction. PRI trunks severely limit the potential for bypassing of toll restrictions.

Reliable indication of far-end disconnect. Blocking of incoming calls is prevented because a trunk is not immediately released; instead, there is a delayed indication of disconnect.

Improved Station Message Detail Recording (SMDR). Call records provide more accurate duration information,

Shared use of B-channels for Megacom WATS and Megacom 800 on a call-by-call basis for more efficient use of facilities.

Supports digital data transmission at speeds up to 19.2 kbps for asynchronous and 64 kbps for synchronous.

System Features Used For Data

Some communications system features provided for voice service may also be used to enhance the usage of data facilities:

Account Code Entry allows tracking of outgoing data calls for billing, forecasting, or budget reports.

Auto Answer All allows a modem with automatic answering capability to answer data calls when the user is away from the station. This feature is used for Analog Voice and Analog Data configurations only.

ARS (Hybrid/PBX mode only) routes calls over outside trunks according to the number dialed and the trunks available. Therefore, the system can be programmed to select the least expensive route for each data call.

When using ARS with digital data calls, make sure that the calls use PRI.

Calling Restrictions, such as Allowed Lists, Disallowed Lists and others, inhibits line access. These features enable companies to control and manage communications costs for outgoing data calls.

A Data Status button monitors station activity (busy, not busy) of any data station. The green LED on the button lights up to indicate a “busy” status. Unlike an Auto Dial or Signaling button, which is programmed to dial a specific number when pressed, the Data Status button does not perform any dialing function. Pressing it, then, has no adverse effect on data calls in progress. The Data Status button is the only button which should be used to monitor station activity.

Dial Access to a Direct Facility Termination (DFT) provides access from a digital data station to outside lines supporting the PRI interface.

5-16

System Features Used For Data


Idle Line Preference automatically selects the first available line for data calls.

Last Number Dialed automatically places a call to the last number dialed from that station. Dialing sequence must include dial-out code for outside calls.

Personal Speed Dial allows quick dialing (a 2-digit code) of frequently used numbers on 10-button phones. The dialing sequence requires a dial-out code for outside calls.

NOTE:

Use this feature on telephones that have only 10 or fewer buttons. If you have an MLX-20L telephone, you can program Personal

Directory entries instead of Personal Speed Dial codes. If you have any other MLX or analog telephone, you can program Auto Dial buttons. If you program Personal Speed Dial codes on telephones with more than 10 buttons, you may delete features you have already programmed onto buttons.

Pool Access to external transmission facilities (Hybrid/PBX Mode only) allows data endpoint dialing to seize access to pool numbers servicing outside lines and trunk.

Privacy prevents loss of data by ensuring that data transmission is not interrupted accidentally. The Privacy feature is automatic for data calls on digital data stations and on analog data stations with analog multiline telephones. It is activated manually on all other analog data stations.

System Speed Dial allows quick dialing of numbers that are used often and provide for data security. Dialing sequence requires a dial-out code for outside calls.

NOTE:

Certain system (voice) features interfere with data connections. These features should be disabled and include the following:

■ Voice Announce

■ Call Waiting

■ Automatic Callback

System Features Used For Data 5-17


Endpoint Communications Features

The communications system supports the use of data equipment or data software. These features are provided by the data station hardware/software in the personal computer, data terminal, or communicating device, such as the

7500B Data Module or modem.

■ Data transport mode selection (for example, 7500B Data Modules support DMI modes 0, 1, 2, and 3)

■ Data metering (speed matching) of bit rates between digital data endpoints

■ Data terminal dialing

■ Automatic answering of data calls

PRI Applications

The communications system provides an interface between PRI services and the small business customer. Some advanced digital applications are already supported by the communications system, which is unique because it is the first

Key system to allow customers to use PRI services. These applications include high-speed fax transmissions and Video Conferencing.

This section provides the configurations supported for fax transmissions on the

Group IV (G4) Fax machine and a general description of video conferencing connections.

Fax Transmissions Application

The Group IV (G4) Fax machine is a fax unit, offering 400x 100 dots per inch

(DPI) in fine mode. It can operate at any speed for communications with a

Group Ill (G3) Fax machine or another Group IV (G4) Fax machine. When speed is essential, it can transmit at 64 kbps and achieve speeds as fast as 3 seconds per page.

Supported Configurations

There are three ways to connect a Group IV (G4) Fax machine to the communications system for transmitting and receiving data:

■ A Direct RS-232 Interface

■ An RS-232 to V.35 Interface Conversion

■ A Direct V.35 Interface

These connections are only a guide, not an assurance that different fax machines with other proprietary interface connections also operate properly.

The configurations are based on compatibility testing that used Canon and

5-18 PRI Applications


Ricoh fax models on the communications system. The communicating adjunct

is the 7500B Data Module, (see Chapter 2 for a description of the 7500 Data

Module).

Group IV (G4) Fax machine are available from a number of manufacturers, each of whom uses proprietary interface connections. One manufacturer may even use different interfaces as standard from one model to another. Some require the buyer to specify various CCITT versions for the standard EIA-RS-232 interface, for example, V. 11, V.28, or V.35. In most cases, these interfaces are simple plug-in connections and are off-the-shelf items. It is important to know before buying the fax machine what interfaces are required. This also means knowing the transmission type the machine normally operates in, asynchronous or synchronous. (When operating behind the communications system, it is operates synch ronously.)

Fax Configuration 1: Direct RS-232 Interface

This is the recommended connection. The other two methods, described later, are shown as alternatives when the EIA interface on the fax machine requires that they be used. The communications system does support the three methods, but there is virtually no advantage of using one method over another; it is only a matter of which interface the fax machine uses.

Figure 5-2 shows the configuration for a direct RS-232 interface, a synchronous

DTE-to-synchronous DCE configuration that includes communication between a

7500B Data Module and a Group IV (G4) fax machine.

PRI Applications 5-19


Figure 5-2. Direct RS-232 Interface

To use this method, it is important that the fax machine be ordered with an EIA-

RS-232 connection, with V.28 or V.35 interface.

In this method, there is a direct connection between the Group IV (G4) Fax machine and a 7500B Data Module. The standard RS-232 interface on the back panel of the Group IV (G4) Fax machine is connected from the jack by an

EIA-232D cable to the RS-232D interface jack (port 1 ) on the 7500B Data

Module. This connection is a simple jack-to-jack plug-in operation. It provides the imagery transmission path.

The 7500B Data Module is a DCE terminal adapter that connects unattended

DTE or other DCE to the Basic Rate Interface (BRI), for example, a 008 MLX or

408 GS/LS-MLX (Release 2.0 only) module in the control unit. This application describes the Group IV (G4) Fax machine as synchronous DTE sending and receiving imagery transmission through the digital network.

5-20 PRI Applications


The dialing path in this configuration requires a Multipurpose Enhancement

Board that is installed in port 2 of the Data Module. The dialing path is then established by connecting the RS-366 jack on the Group IV (G4) Fax machine to the RS-366 jack in port 2 on the back panel of the 7500B Data Module. An

RS-366 cable is required for the connection.

The 7500B Data Module is then connected from the Line jack on the back panel to an MLX port on an 008 MLX or 408 GS/LS-MLX module (Release 2.0 only) in the control unit, Completing the link is the PRI trunk connection that is plugged

into the jack on the channel service unit (CSU) shown in the Figure 5-2. The

CSU is then plugged into the 100 DSI Module in the control unit.

This connection is recommended when manual dial is used to dial the fax transmission telephone numbers from the fax dial pad.

It is recommended that the 75006 Data Module be located in a safe location, away from accidental tampering with the configuration settings, which could result in misdialed calls.

For the 7500B Data Module to operate properly, the DCE/DTE Flipboard circuit card in the unit must be in the proper position before the Group IV (G4) Fax machine can operate as a DCE unit.

The MLX telephone in the configuration shares the same MLX port as the 7500B

Data Module and can be used to send and receive voice calls. It is not essential to the fax operation. However, if an MLX set is not connected to the

7500B Data Module, a 100-ohm terminating resistor adapter must be installed close to the 7500B Data Module on the line to the carrier.

Fax Configuration 2: RS-232 to V.35 Interface Conversion

This configuration is used when users need to operate other adjunct equipment from the 7500B Data Module. Other equipment may include modems, automatic calling equipment (RS-366 interface), or DTE with V.35 interface.

Figure 5-3 shows the connections required for a Group IV (G4) fax V.35

interface connecting to a data module RS-232D interface.



Figure 5-3. RS-232 to V.35 Interface Conversion

A V.35-to-RS-232 converter must be used between the fax machine and the

7500B Data Module. This configuration requires the use of the optional

Multipurpose Enhancement Board in port 2 of the 7500B Data Module to set up an RS-366 dialing interface. The configuration allows the Group IV (G4) Fax machine to be connected to port 1 for imagery transmission. The fax machine is connected by a V.35 cable to the converter’s V.35 port, and then from the converter’s RS-232 port to the 7500B Data Module’s RS-232 port. The dialing is accomplished from the RS-366 interface on the fax machine to the RS-366 interface in port 2 of the 75008 Data Module.

The Multipurpose Enhancement Board provides an RS-366 auto dial interface on port 2. It converts the RS-232 interace on port 1 on the main circuiit board from asynchronous to synchronous mode. The V.35 adapter cable must be ordered separately from the board in order to operate at data rates of 56 and 64

Kbps. Without the cable, data rates are limited to 1200, 2400, 4800, 9600, and

19,200 bps.



The 7500B Data Module is connected by a D8W cord to the MLX port on the

008 MLX or 408 GS/LS-MLX module (Release 2.0 only).

The PRI interface is established from the 100 DS1 Module to the CSU and then to the ESS switch and the PRI interface.

For the purposes of the compatibility tests, a Shore Microsystems Model SM-

100 RS-232/V.35, converter was used to test this configuration. The converter is customer-supplied equipment that can be purchased from a data equipment vendor.

Fax Configuration 3: Direct V.35 Interface

If the Group IV (G4) Fax machine is equipped with a CCITT-V.35 interface, use this configuration method. If the fax machine is equipped with a V.35 interface, an optional connection board must be used with the 7500B Data Module. This configuration is required if the customer does not wish to purchase a converter or if the dialing is to be done on the front face of the 7500B Data Module. This connection requires dialing from the 7500B Data Module, that is, the 7500B

Data Module must be located in a work area where it cannot be accidentally reconfigured. Accidental reconfiguration could cause the fax transmissions to malfunction, Also, dialing for fax transmissions from the 7500B Data Module is subject to a high error rate due to misdials.

Figure 5-4 shows the connection between a Group IV (G4) fax machine and a

7500B Data module using an EIA-V.35 interface.



Figure 5-4. Direct V.35 Interface

This connection requires an optional High Speed Enhancement Board to be included in the 7500B Data Module.

The High Speed Synchronous Enhancement Board provides a V.35 interface at synchronous data rates of 48, 56, or 64 Kbps on port 2. The connection is via an V.35 external adapter cable that converts the 25-pin male connector on Port

2 to the industry-standard 34-pin V.35 interface. The cable is packaged with the board.

The board comes with an adhesive V.35 label that must be affixed to the back panel of the 7500B Data Module so that port 2 cannot be mistaken for a second

EIA-232D interface.

When the High Speed board is used, no connection is allowed to port 1.

In this configuration, no dialing connection is made between the Group IV (G4)

Fax machine and the 7500B Data Module, and all dialing must be made from the front panel of the data modem.

A D8W cord also connects from the Line port on the 7500B Data Module to the

008 MLX or GS/LS-MLX module port on the communications system. The PRI interface is made from the 100 DS1 Module.



Video Conferencing Application

This section shows an example of how to connect a 7500B Data Module with a

multipurpose enhancement board for synchronous data communication, which allows data transmission at the speed of 56 or 64 kbps.

High-Speed Synchronous Enhancement Board

For instructions on using a high-speed synchronous enhancement board or any information on setting up the 7500B Data Module, not included in this section, see the documentation packaged with the 7500B Data Module,

This section is intended only as a guideline for connecting video conferencing equipment to the system. For any additional information, see the documentation packaged with the video codec.

Hardware Requirements

008 MLX or 408 GS/LS-MLX module in the control unit

ESF T1 CSU

NOTE:

If any other type of CSU is used, your customer support organization cannot support installation and maintenance.

Two Shore Microsystems SM-100 EIA-232/V.35 converters (or equivalent)

Two 7500B Data Modules

Two 7500B Data Module feature package 2 upgrades (user’s manuals included)

Two multipurpose enhancements boards. By installing a multipurpose enhancement board in each 7500B Data Module, you can provide synchronous communication and RS-366 ACU interface.

Two WP901 10-L7 power supplies (one per standalone 7500B Data

Module)

Two 440A4 terminating resisting adapters

Z77A multiple mounting (mounting for multiple 7500B Data Modules)

Cables:

— Two male/male EIA-232-D cables, 8 feet (24 meters), to connect the PORT 1 jacks on the data modules to the EIA-232/V.35

converters

— Two male/male V.35 DB-37 cables, 8 feet (24 meters), to connect the V.35 communication ports on the video codec to the

EIA-232/V.35 converters

— Two male/male RS-366 DB-25 cables, 8 feet (24 meters), to connect the RS-366 dialing port of the video codec to

PORT

2 on the 7500B Data Module



Video Conferencing Connections

Figure 5-5 shows an example of video conferencing connections.

Figure 5-5. Video Conferencing Connections

NOTE:

If you are connecting MLX telephones, omit both 440A4 terminating resistors, and note that the maximum cord length from the data module to the telephone is 80 feet (24 meters).


Product Ordering Information

Ordering Codes

Component PEC

Control Unit

MERLIN LEGEND Control Unit 6140-CU2

Basic carrier and housing

Power Supply module

Processor

Feature Module

MERLIN LEGEND Control Unit w/408 ATL


Power supply module

Processor

Feature module

408 GS/LS

MERLIN LEGEND Control Unit w/408 MLX


Power supply module

Processor

Feature module

408 GS/LS-MLX

MERLIN LEGEND Control Unit w/12 x 24 MLX


Power supply module

Processor

Feature module

408 GS/LS-MLX (qty.3)

6140-CA2

6140-CD2

6140-242

Expansion Unit

Expansion carrier and housing

Power supply module

Expansion Unit Cover

61490

N/A

Comcode App. Code

106388614 403E

105743801 391A 1

106215155 517A27

106874738 517G25

106388614

105743801

106215155

106874738

106064678

403E

391A1

517A27

517G25

517A26

106388614

105743801

106215155

106874738

106698590

403E

391A1

517A27

517G25

517/429

106388614

105743801

106215155

106874738

106698590

403E

391A1

517A27

517G25

517A29

106388630 403F

105743801 391A1

106388259 17A

Product Ordering Information A-1

Product Ordering Information

Ordering Codes (continued)

Component PEC Comcode App. Code

R1/R1.1 Upgrade to R2 6141-102A 106874738 517G25

Upgrade from Merlin II 6140-UD2A

(Basic)

Cover

Processor

008 MLX

Feature module

MLX-20L telephone (choose one):

Black

White

Expansion cover

(zero, one, or two)

Upgrade from Merlin II 6140-U2LA

(w/MLX module)

Cover

Processor

Feature module

Expansion cover

(zero, one, or two)

106388234

106215155

105628010

106064660

106743420

106743438

106388259

106388234

106215155

106874738

106388259

16A

517A27

517A21

517A25

7713D01 B-003

7713001 B-264

17A

16A

517A27

517G25

17A

Trunk and Station Modules

008 MLX

008 OPT

012 (T/R)

100D(DS1)

400EM (tie trunk)

400 GS/LS/TTR

408 GS/LS

408 GS/LS-MLX

800 DID

800 GS/LS

61486

61489

61487

61491

61492

61483

61481

61493

61488

61484

105628010

106387525

106553779

105461560

105311401

105628044

106064678

106698590

105628077

105627996

Vintage Trunk and Station Modules

400 (with TTRs)

408 LS

008 (ATL)

800 LS

61379

61482

61485

61384

105408892 517B12

105512495 517C1

105351092 517B3

105351100 517B4

517A21

517A28

517E13

517B15

517A14

517B18

517A26

517A29

517B20

517A19

A-2 Product Ordering Information



Component PEC Comcode

Telephones

MLX Telephones

MLX-10

English (black) 3156-02B 106743024

English (white) 3156-02W 106743032

French (black) 3156-F2I 106633886

French (white) 3156-F2I 106633894

Spanish (black) 3156-S2I 106613508

Spanish (white) 3156-S2I 106613516

MLX-10D

English (black) 3156-03B 106743040

English (white) 3156-03W 106743057

French (black) 3156-F3I 106633928

French (white) 3156-F3I 106633936



MLX-20L

English (black) 3156-05B 106743420

English (white) 3156-05W 106743438

French (black) 3156-F5I 106634421

French (white) 3 156-F5I 106634439



MLX-28D

English (black) 3156-04B 106743503

English (white) 3156-04W 106743511

French (black) 3156-F4I 106634470

French (white) 3156-F4I 106634488



Analog Multiline Telephones (black)

MLC-5

BIS-10

BIS-22

BIS-22D

3168-MLC 105515332

3165-10B 105161061

3166-22B 105188809

3166-DSB 105630420

BIS-34

BIS-34D

3167-34D 105167027

3167-DSB 105630529

MERLIN PFC (ATL) 3169-PF2 106681562

PFC paper 31690 106673361

App. Code

7712 D01B-003

7712D01B-264

7712D01A(29)-003

7712D01A(29)-264

7712D01A(22)-003

7712D01A(22)-264

7712D02B-003

7712~2B-264

7712~2A(29)-003

7712D02A(29)-264

7712D02A(22)-003

7712D02A(22)-264

7713D01B-003

7713D01B-264

7713D01A(29)-003

7713D01A(29)-264

7713D01A(22)-003

7713D01A(22)-264

7713D02B-003

7713D02B-264

7713D02A(29)-003

7713D02A(29)-264

7713D02A(22]-003

7713D02A(22)-264

7312HO1A-003

7313HO1A-003

7314HO1A-003

7315HO1B-003

7317HO1A-003

7317HO1B-003





Telephones (continued)

Vintage Analog Multiline Telephones (black)

5-Button

10-Button

10-Button HFAI

34-Button

34-Button Deluxe

34-Button BIS

34-Button BIS/DIS

3160-111 105217426 Z7302H01D0-003

3161-172 105217509 Z7303H01D-003

3161-161 105371942 Z7309H01C-003

3162-412 103842050 Z7305H01B-003

3162-417 105217715 Z7305H02D-003

3162-BIS 103981965 Z7305H03D-003

3162-DIS 103981981 Z7305H04C-003

Single-Line Telephones

8110 Analog Voice

Black

White

8102 Analog Voice

Black

White

7102

Black

Misty cream

2500 YMGK

(message waiting, recall, touch-tone, desk)

Black

Misty cream

2500 MMGK

(recall, touch-tone, desk)

Black

Misty cream

2500 MMGJ

(touch-tone, desk)

Black

Misty cream

2554 MMGJ

(touch-tone, wall)

Black

Misty cream

500 MM

(rotary, desk)

Black

Ivory

Beige

3193-001

3192-001

3185-MWR

3178-NHL

3101-ETR

3101-EBD

3101-EBW

3100-ORD

106272321

106272339

106272305

106272313

105335285

105330419

105480578

105480560

105414130

105414122

105414155

105414148

105480081

105480032

103870234

103870226

103870267

8110A01A-003 811

8110A01A-264 811

8102A01A-03 810

8102A01A-264 810

7102A01A-003

7102A01A-215

2500YMGK-003

2500YMGK-215

2500MMGK-003

2500MMGK-215

2500MMGJ-003

2500MMGJ-215

2554MMGJ-003

2554MMGJ-215

500 MM-03

500 MM-50

500 MM-60




Component PEC

Telephones (continued)

Single-Line Telephones (continued)

554 BMPA

(rotary, wall)

Black

Ivory

Beige

Cordless Telephones

Model 5320

3 100-ORW

31CJ3-CLS

Consoles

MERLIN II

System Display Console

DSS

English (black)

English (white)

Spanish (black)

Spanish (white)

61392

3156-DCB

3156-DCW

3156-SDI

3156-SDI

Comcode

103823498

103823506

103823555

105543516

105229744

105685481

105685499

106613672

106613680

App. Code

554BMPA-3

554BMPA-50

554BMPA-60

CS6300U11A-229

7318H01A-003

604A1-003

604A1-264

604A1(22)-003

604A1(22)-264

Applications

SPM Version 2.0 — DOS

SPM Version 2.0 — UNIX System

Call Accounting System (CAS)

CAS Plus V3 Bundle w/80-col.

Parallel Printer

CAS Plus V3 Bundle w/132-col.

Parallel Printer

CAS Plus V3 Software

Rate Table*

CAS Plus upgrade

UNIX CAS (LEGEND)

CAS/H

CAS/H LEGEND 100S

CAS/H LEGEND 200S

CAS/H Rate Table*

61495

61496

1201-NP1

1201-WP1

1201-DR1

12010

12009

1201-U12

1201-H10

1201-H20

12050

106906092

N/A

406362244

406158444

406158394

406478537

405799255

405788289

405788420

* Consult AT&T or an authorized dealer for other area-specific information.





Applications (continued)

Call Accounting Terminal (CAT)

CAT BASIC/B

CAT Terminal

Printer

CAT Basic Rate Table*

(Update Chip)

CAT/B 150S

CAT/H 150S

CAT/B Rate Table* (update)

CAT/H Rate Table* (update)

Call Management System (CMS)

5¼* floppy disk

3½* floppy disk

Board

CONVERSANT INTRO Application

CONVERSANT INTRO Application

Casual User

Conversant Intro Application

Casual Dev. & Data

Inn Manager w/Base Software

Site Specific Software

20 station

40 station

60 station

80 station

100 station

355AF Adapter, Receptacle

355A Adapter, Plug

14’ Cord

3600-010

36014

3600-023

3600-024

36023

36024

1207-100

8301-100

4201-100

4201-101

4201-102

7051-INN

70501

70502

70503

70504

70505

2709-A25

2750-A24

2725-07N

406669762

406637306

406669739

406478800

406478818

406478792

406478784

106496540

106496532

106198815

406670984

406670992

406671008

406671016

406671024

406671032

105012645 355AF

105012637 355A

103687802 D8W-87 14FT

* Consult AT&T or an authorized dealer for other area-specific information.




Component

Applications (continued)

MERLIN LEGEND Integrated Solution Ill

Controllers

100 MB MC-II + Processor

4 x 100 MB MC-II + Processor

4MB Memory Upgrade

COLOR MONITOR

KEYBOARD

9 to 25 PIN ADAPTER

CARTRlDGE TAPES (qty.2)


4 x 200 MB MC-II + Processor

4MB Memory Upgrade

COLOR MONITOR

KEYBOARD

9 to 25 PIN ADAPTER

CARTRIDGE TAPES (qty.2)

200 MB MC-III + Processor

8 x 200 MB MC-III + Processor

COLOR MONITOR

KEYBOARD

9 to 25 PIN ADAPTER

2 x 250 MB CART, TAPES


8 x 500 MB MC-III + Processor

COLOR MONITOR

KEYBOARD

9 to 25 PIN ADAPTER

2 x 250 MB CART. TAPES

SOFTWARE APPLICATlONS

MERLIN LEGEND IS-III

LEGEND IS-III PLATFORM SOFTWARE

UNIX SPM 2.0

LEGEND IS-III System Manager’s

Guide

LEGEND IS-III I+M Guide

LEGEND IS-III Tape

PEC

4200-503

4200-503

4200-912

4200-936


406506329

106219553

406504571

406504563

406139394

106220666

406506337

106219533

406504571

406504563

406139394

106220866

406700930

406504571

406504563

406708503

406760009

406700914

406504571

406504563

406708503

406760009

106894942




Component

Applications (continued)

MERLIN LEGEND R2 AVP. 2.1.1

IVPSS 2.0

AVP2.1.1

AVP/FA MLR2 Switch Integ. SW

MERLIN LEGEND Integ. AVP/FA Admin.

MERLIN LEGEND AVP User’s Guide (50)

MERLIN LEGEND AVP System Manager’s

Guide

MERLIN LEGEND AVP/FA Planning Guide

MERLIN LEGEND FAX Attendant

FAX ATTENDANT 2.1.1 CO-RESIDENT

Base PKG.

AVP/FA MLR2 Switch Integ. SW

FAX ATTENDANT User’s Guide (50)

MERLIN LEGEND AVP/FA Planning Guide

Reference Sheet

OTHER (Voice Boards)

IVP4 Board LEGEND

IVP6 Board LEGEND

IS-III Bundles

100 MB MC-II + 4 x 4 AVP


IVP4 Board

UNIX 3.2.2

CART. TAPE UTILITIES


LEGEND R2 AVP 2.1.1

200 MB MC-II + 4 X 12AVP


IVP4 Board

UNIX 3.2.2



LEGEND R2 AVP 2.1.1

200 MB MC-111 +8 x 12AVP


IVP4 Board (qty.2)

UNIX 3.2.2



LEGEND R2 AVP 2.1.1

PEC Comcode App. Code

106876311

106876220

8306-100 106248651

N/A 106856271

6146-100

4200-503

8306-100 106248651

N/A

N/A

106529548

106632938

N/A

N/A

106894942

106876311

6146-200

4200-503

8306-100 106248651

N/A 106529548

N/A 106632938

N/A

N/A

6146-012

4200-912

106894942

106876311

8306-100 106248651

N/A

N/A

106529548

106632938

N/A

N/A

106894942

106876311

A-8 Product Ordering Information



Component


500 MB MC-III + 8 x 36 AVP

500 MB MC-Ill + Processor

IVP4 Board (qty.2)

UNIX 3.2.2



LEGEND R2 AVP 2.1.1

100 MB MC-II + 4 X 4 AVP, CAS


IVP4 Board

UNIX 3,2.2



LEGEND R2 AVP 2.1.1

CAS—IS-III

200 MB MC-II + 4 X 12 AVP, CAS


IVP4 Board

UNIX 3.2.2

CART, TAPE UTILITIES


LEGEND R2 AVP 2.1.1

CAS —IS-III

200 MB MC-Ill + 8x 12 AVP, CAS


IVP4 Board (qty.2)

UNIX 3.2.2



LEGEND R2 AVP 2.1.1

CAS—IS-III

500 MB MC-III + 8 x 36 AVP, CAS


IVP4 Board (qty.2)

UNIX 3.2.2



LEGEND R2 AVP 2.1.1

CAS—IS-III

PEC Comcode App. Code

N/A

1201-U12

6146-112

4200-912

8306-100

N/A

N/A

N/A

N/A

1201-U12

6146-136

4200-936

8306-100

N/A

N/A

N/A

N/A

1201-U12

6146-036

4200-936

8306-100

N/A

N/A

N/A

N/A

6146-101

4200-503

8306-100

N/A

N/A

N/A

N/A

1201-U12

6146-201

4200-503

8306-100

N/A

N/A

N/A

106248651

106529548

106632938

106894942

106876311

106248651

106529548

106632938

106894942

106876311

406478537

106248651

106529548

106632938

106894942

106876311

406478537

106248651

106529548

106632938

106894942

106876311

406478537

106248651

106529548

106632938

106894942

106876311

406478537




Component PEC


200 MB MC-III + 8 x 12 AVP, 4 x 1000 FAX


IVP4 Board (qty.2)

UNIX 3.2.2



LEGEND R2 AVP 2.1.1

LEGEND FAX ATTENDANT

500 MB MC-III + 8 x 36 AVP, 4 x 3000 FAX


IVP4 Board (qty.2)

UNIX 3.2.2



LEGEND R2 AVP 2.1.1


200 MB MC-III + 8 x 12 AVP, CAS, 4 x 1000 FAX


IVP4 Board (qty.2)

UNIX 3.2.2



LEGEND R2 AVP 2.1.1


CAS—IS-III

500 MB MC-III + 8 x 36 AVP, CAS, 4 x 3000 FAX


IVP4 Board (qty.2)

UNIX 3.2.2



LEGEND R2 AVP 2.1.1


CAS—IS-III

MERLIN Attendant

Hardware

Documentation

6146-212

4200-912

8306-100

N/A

N/A

N/A

N/A

6146-236

4200-936

8306-100

N/A

N/A

N/A

N/A

6146-312

4200-936

8306-100

N/A

N/A

N/A

N/A

1201-U12

6146-336

4200-936

8306-100

N/A

N/A

N/A

N/A

1201-U12

6125-All


106248651

106529548

106632938

106894942

106876311

106876220

106248651

106529548

106632938

106894942

106876311

106876220

106248651

106529548

106632938

106894942

106876311

106876220

406478537

106248651

106529548

106632938

106894942

106876311

106876220

406478537

406406090

106431265




Component


MERLIN MAIL™ Voice Messaging

System for the MERLIN LEGEND

Communications System

Two-port

MERLIN MAIL unit

Remote maintenance device

Four-port

MERLIN MAIL unit

Remote maintenance device

Two-port line card (R2)

(upgrade from two to four)

PEC

6107-005


406824532

406466193

6107-006

406824540

406466193

6107-007 406824524

System Adjuncts and Adapters

Auxiliary Power Unit 9024 61416

Channel service units (CSUs)

ESF T1

Cable (8 mod, wire wrp)

Cable (8 mod, 15-pin sub)

Stand-alone wall mount

Stand-alone wall bracket

551 T1 L1

Power unit

Unit

Cord

Stand-alone wall mount

Stand-alone wall bracket

2152-ESF

21555

2155M

21545

2152-15T

21530

21545

Optional Equipment

Peripheral Interface 62515

Async. Data Unit, Receptacle 2169-004

Async. Data Unit, Plug 2169-001

Aux Power (2 required) 21691

Transformer

Adapter (2486)

Cord

Adapter (4006)

406467142

405616293

406012609

406012591

405!370104

405616277

403768179

403242639

103895660

405970104

405616277

105179303

103964185

103963963

N/A

102600517

102802113

102937620

103848859

9024

305010171-001

513861312-5050 FT

513823015-5050 FT

380-100213-001

380-100542-001

305-10097-001

KS22911LI

DW4A-SE 10FT- IP

380-100213-001

380100542-001

KIT PRTS-D181558

Z3A4

Z3A1

Product Ordering Information A-11




System Adjuncts and Adapters (continued)

Electrostatic discharge (ESD) suppression kits

D-181574

D-181589

D-181590

D-181591

D-181593

EMI filter

In-Range Out-of-Building-343B

(IROB) unit — analog multiline*

IROB unit — MLX*

Fuse block 505A for IROB

(8 fuse blocks per box)

7500B Data Module

Stand-alone power supply

Multiple mounting

7500A upgrade kit

Digital Magic On Hold® player

(replaces analog unit)

Digital Announcement Unit

Modem 2224G

Music Coupler

PagePac® 6

PagePac 6 Pius

PagePac 20 PowerMate™

Trunk Adapter

PagePac6 Portsaver

N/A

N/A

N/A

N/A

N/A

32918

32919

2164-BDM 105657654

21625

21626

21627

3128-020 406659326

3119-001

105179329

105201891

105201909

105201917

105201933

103965206

406721738

106417447

406610337

405509852

105441166

105688501

2224-CEO 105659965

61398

5323-006 405701277

5323-008 405701608

5323-005 403308026

53518

53519

406747774

406143925

405223298

405703026

D181574

D181589

D181590

D181591

D181593

Z200A

343B

505A ASSY OA WD

Z750B-L1

WP90110L7

277A

D 182208

PAKG DMOH1 DIG L

ATTDAU

2224C-L1 D/2

ASSY-K23395 L3

22052-006 PG PC

22052-000 PG PC6

ADP06

22050-900

220520020

* Any multiline off-premises telephone must have an appropriate IROB protector at the control unit location and at the off-premises location.




Component PEC Comcode

System Adjuncts and Adapters (continued)

AT&T Door Phone Speaker

PagePac 20 Talkmate

PagePac 50

PagePac 200

PagePac VS

200 WATT AMPLICENTER

50 WAIT AMPLICENTER

53240

53501

5322-051

5322-201

5322-700

52120

52150

406269860

403307994

403305444

403305469

403307192

403305493

403305501

Ring generator unit 61388 105213201

SMDR Printers

Parallel Printer (80-column)

Parallel Printer (132-column)

CAT Printer (serial)

Uninterruptible Power Supply (UPS)

500 VA (15min)

Reserve (1 hr)

Universal Paging

Access Module (UPAM)

TAM-B

PRS-48

WMT- 1A

4200-570

4200-571

4200-572

2403-050

24035

58500

N/A

N/A

N/A

53505

406637314

406712067

406716464

105610141

105610174

405891698

405899972

405742735

405891680

ZoneMate™ 9

Dialer unit

Control unit

ZoneMate 39

Dialer unit

Control unit

External Alerts

Loud external ringer

E1CM-type

Gray

Ivory

E1CM ringer and parts

290A adapter

Ringer

Mounting plate

Cord

53506

31016A

31019A

61211

404057911

405024134

404057929

405024134

103117016

102872934

102917952

102992252

102872934

102988466

103938494

App. Code

PE53501 AT

5322-051 VC PG

5322-201 VC PG

5322-7003500 VC PG

52120 AT

52150 AT 50W

129B RING GEN

515005C111

0053150

KIT-UPAM

D181900

D181900

D181900

DIAL UNIT-9ZONE

CNTL 22050-020

39 ZONE SELECT

CNTL-22050-020C

RINGER-L1AMP-49

RINGER-E1CM-49

RINGER-E1CM-50

D-181233

290A ADPTR

E1CM-49

1049A

CORD-D4CH-87-25

Product ordering Information A-13



Component PEC Comcode App. Cods

System Adjuncts and Adapters (continued)

Supplemental Alerts

Alert Bell

Network Interface Alert Bell

Alert Horn

Alert Strobe

Alert Chime

5580-002 406293720 TB591-1

61211 102872934 RINGER-E1CM-49

5580-021 406207217 THET4-1

5580-041 403319197 AT-WHL LK

5580-030 405136060 CHBT2-1

Telephone Adjuncts and Adapters

General Purpose Adapter

(GPA) (analog)

Multi-Function Module (digital)

Supplemental Alert Adapter

(SAA)

MLX-10 and MLX-10D protective cover

2301-GPA 103977997 Z1C

3156-MFM 105746474 540A1

2301-SSA 105031199 ADPTR-856A

N/A 406648469 N/A

31757 MLX Telephone Power

48V Power Supply

Modular Power Cord

400B2 Adapter

Analog Multiline Telephone Power

48V Power Suppiy

Modular Power Cord

Z400F Adapter

62510

405331711 KS22911L2

102937620 D6AP-87

104152558 400B2

105105514 D181522

405331711 KS22911 L2

102937620 D6AP-87

103942857 Z400F

Single-line telephones

Program, Pause, and Auto

Dial button conceal kit for

8100 series telephones

4A Speakerphone

Power unit

Block connector

Adapter for single-line telephone

Adapter for multiline telephone

Transmitter (black)

Transmitter (ivory)

Loudspeakers

Black

Ivory

Green

Beige

White

31931

3120-02W

106248370 Kit-D 182363 Analog

102139938

102434925

102813888

102949013

103971891

103971909

103873873

103873881

103873899

103873907

103873964

4A

PWR UNIT-85B1

BLK CON-82B-49

ADPTR-223C

ADPTR-223D IP

TRMR-680AF-03

TRMR-680AF-50

LSPK-108AA-03

LSPK-108AA-50

LSPK-108AA-51

LSPK-108AA-60

LSPK-108AA-58





Telephone Adjuncts and Adapters

Single-line telephones (continued)

(continued)

S201 Speakerphone

Black

Misty cream

CS201 Conference

Speakerphone

Black

Misty cream

S202A Speakerphone

Black

Misty cream

S203A Speakerphone

Black

Misty cream

31 52-007A

3131-004A

3152-008

3131-008

Message Waiting Indicator 31032

103786786 D8W-87 7FT

106192651 MOD-S201AP-003

106192693 MOD-S201AP-215

103786786 D8W-87 7FT

106270325 MOD3-CS201A-003

106270333 MOD-CS201A-215

105721088 TEL-S202A-003

105721096 TEL-S202A-215

105058340 MOD-S203A-003

106508365 MOD-S203A-215

103966396 Z34A

Hands Free Unit (HFU) 3163-HFU 103814356 MOD-S102A

Headsets and Adapters

StarSet® Headset

Mirage® Headset

Supra® Headset

Supra NC® Headset

Headset Adapter

500A Headset Adapter

Modular Amplifier

Plug Prong Amplifier

3122-030

3122-050

3122-040

3122-060

3164-HFA

3152-001

3122-020

3122-010

406445627

406445783

406445791

406741900

105752042

106690043

405331711

102479904

104152558

406445619

406445601

KS23822L3

KS23822L4

KS23822L5

KS23822L12

ADPTR-502C-003

Adapter EL-500A-266

Pwr Sup-KS22911L2

Cord-D4BU-29 Std 7FT

Adaptr-40082

KS23822L2

KS23822L1

MLX Telephones Miscellaneous Add-Ons and Replacement Parts

Handsets and Cords

Handset (black)

Handset (white)

N/A

N/A

Handset, amplified hearing 31052

Black

White

Misty cream

106050065 K2S1-003

106053408 K2S1-264

105581896 K6S2-003

106248248 K6S2-264

105581904 K6S2-215





MLX Telephones Miscellaneous (continued)

Handsets and Cords (continued)

Handset cord, 9’ (2.74 m), black N/A

Handset cord, 9’ (2.74 m), white N/A


Handset cord, 12’ (3.66 m), white N/A


DSS line cord, 2’ (61 cm) N/A

Desk Stands and User Trays

Stand (large, black)

Stand (large, white)

Stand (small, black)

Stand (small, white)

User tray (black)

User tray (white)

N/A

N/A

N/A

N/A

N/A

N/A

846320851 STAND-LARGE BL

846320844 STAND-LARGE WH

846320810 STAND-SMALL BL

846320802 STAND-SMALL WH

846320240 USER TRAY DWR B

846320232 USER TRAY DWR W

Designation (Button Assignment) Cards and Covers

Card*—MLX-1O, MLX-1OD

Card*–MLX-20L

Card*–MLX-28D

Card set†—DSS

Card covers†—DSS (black)

Card covers†—DSS (white)

Card set‡—QCC

Card covers§—

MLX-10, MLX-10D, MLX-20L

Card covers§—MLX-28D

N/A

N/A

N/A

N/A

N/A

N/A

N/A

N/A

N/A

105635429 H4DU-003 9FT

105701809 H4DU-2649’BULK

102401445 H4DU-3 12FT IP

102402609 H4DU-264 12’IP

105523666 H4DU-3 25’

106187545 CORD D8AC-87

846865939

846865947

846865954

106448756 KIT-D182464

106448731 KIT-D 182462 PRT

106448749 KIT-D 182463 PRT

106561673 KIT-D182562 PRT

106448681 KIT-D 182457 PRT

106448699 KIT-D 182456 PRT

Analog Multiline Telephones Miscellaneous Add-Ons and Replacement Parts

Desk Stands and Wall Mounts

Adjustable desk stand,

10-button

Adjustable desk stand,

34-button

32002 103746855 11A

32003 103746863 11C

* 10½ sheets per package.

† Includes both top and bottom cards or covers

‡ 8 cards per kit (four sets)

§ 4 per package





Analog Multiline Telephones Miscellaneous Add-Ons

and Replacement Parts (continued)

Desk Stands and Wall Mounts (continued)

Fixed desk stand,

5- & 10-button

Desk stand/wall mount

14A, BIS-10


14B, BIS-22


14C, BIS-34

Fixed desk stand and wall mount, 5-button

Kit of parts

Wail mount, 10-button

Kit of parts

Wail mount, 34-button

Kit of parts

Faceplates

32004

N/A

N/A

N/A

32000

32001

32006

103746848

103804290

103964458

103979837

103804290

103995882

103747846

103995882

103747853

103995882

BIS-10

BIS-22

BIS-22D

BIS-34 and BIS-34D

N/A

N/A

N/A

N/A

105203186

105336986

105690762

105203194

10A

14A-003

Z14B-003

14C-003

14A

D-181230

201A

D-181230

203A

D-181230

KIT PRTS-D-181582

KIT PRTS-D-181786

KIT PRTS-D-182210

KIT PRTS-D-181583

Button Label Sheets

BIS-10

BIS-22

BIS-22D

BIS-34 and BIS-34D

Display console (FM1 )

(includes one faceplate)

Display console (FM2 & R3)

(includes one faceplate)

N/A

N/A

N/A

N/A

N/A

N/A

105336978

105336960

105690770

105336956

105299754

105486252

KIT PRTS-D-181785

KIT PRTS-D-181784

KIT PRTS-D-182211

KIT PRTS-D-181783

KIT PRTS-D-181727

KIT PRTS-D-182041

Single-Line Telephones Miscellaneous Add-Ons

Ground-Start Button 31021 405792839 Key-KS23566L1




Component

Miscellaneous Parts

Interconnect Wiring Kit

110AB1-100JP12

110A1 trough

D-Rings

D8W cords

Parts list

SYSTIMAX

MERLIN Wiring Kit

110A1 trough (5)

110AB1-100JP12 modular block (2)

110AB1-100FT punch down block (1)

D-Rings (6) patch cords

12 cords, 4-pair, 5’ (1.5 m)

D8W cords

24 cords, 14’ (4.3 m)

Template

Instruction sheet

Parts List

PEC

N/A

N/A

N/A

N/A

N/A

N/A

N/A

N/A

N/A


N/A

N/A

N/A

N/A

N/A

3103-MER

104409396

104407960

842139248

103786802

N/A

106393671

104407960

104409960

103823845

842139248

846619989

103786802

846613933

846613941

846623924


Glossary

7500B Data

Module account code

Accunet address adjunct alternate mark inversion

AMI

A data communications device that allows connection between RS-232 data terminal equipment (DTE) and the communications system control unit via MLX station jacks on the 008 MLX or 408 GS/LS-MLX module. The 7500B

Data Module is used together with a modem in a modern pool to change digital signals to analog signals, and vice versa, which allows transmission between digital and analog data stations.

A code used to associate incoming and outgoing calls with corresponding accounts, employees, projects, and clients.

AT&T’s switched digital service for 56-kbps, 64-kbps restricted, and 64-kbps clear circuit-switched data calls.

A coded representation of the destination or the originating terminal of data, such as the dialed extension number assigned to the data terminal, Multiple terminals on one communications line, for example, must each have a unique address.

Optional equipment used with the communications system such as an alerting device that connects to a multiline telephone or to a telephone jack.

See AMI.

(Alternate mark inversion) A line-coding format in which a binary 1 is represented by a positive or negative pulse and a binary 0 is represented by no line signal, Subsequent binary 1‘s must alternate in polarity or a bipolar violation will occur. AMI is used in the DS1 interface.

Glossary GL-1

Glossary analog transmission

ANI application

ARS

ASCAP

ASN asynchronous data transmission

AT&T Switched

Network

AUDIX Voice

Power

A mode of transmission in which information is represented in continuously variable physical quantities such as

amplitude, frequency, phase, or resistance. See a/so digital transmission.

(Automatic Number Identification) The process of automatically identifying a caller’s billing number and transmitting that number from the caller’s local central office to another point on or off the public network. INFO-2

(Information Forwarding-2) is AT&T’s ANI service.

Software and/or hardware that adds functional capabilities to the communications system. For example, the Call

Management System (CMS) is a DOS–based application that simulates the actions of a system operator by answering calls and distributing them to individual telephones.

(Automatic Route Selection) Routes calls over outside trunks according to the number dialed and the trunks available.

American Society of Composers, Artists, and Producers

(AT&T Switched Network) AT&T telecommunications services provided through a PRI line or trunk: Accunet switched digital service, Megacom WATS, Megacom 800,

Software Defined Network (SDN), MultiQUEST, and Shared

Access for Switch Services (SASS).

A method of transmitting a short bit stream of data, such as printable characters represented by a 7-or 8-bit ASCII code. Each string of data bits is preceded by a start bit and followed by a stop bit, permitting data to be transmitted

at irregular intervals. See also synchronous data transmission.

See ASN.

Automated

Attendant auxiliary power unit

A voice-processing application, part of Integrated Solution

II or Ill (IS II/III), that provides automated attendant, call answering, voice mail, message drop, and information services for use with the communications system.

An IS II, MERLIN MAIL, and MERLIN Attendant application that automatically answers incoming calls with a recorded announcement and directs callers to a department, an extension, or the system operator.

A device that provides additional power to the communications system.

GL-2 Glossary

Glossary

B8ZS

B-channel barrier code basic carrier

(Bipolar 8 zero substitution) A line-coding format that encodes a string of 8 zeros in a unique binary sequence using bipolar violation. See also bipolar signal and bipolar violation.

A 64-kbps channel that carries a variety of digital information streams, such as voice at 64 kbps, data at up to

64 kbps, wideband voice encoded at 64 kbps, and voice at less than 64 kbps, alone or combined with other digital information streams. Also called bearer channel.

A password used to limit access to the Remote Access feature of the communications system.

A piece of hardware that holds and connects the processor module, power supply module, and up to five line/trunk or station modules in the communications system. See also

expansion carrier.

See BRI.

Basic Rate interface baud rate bearer channel

Behind Switch mode binary code

A unit of transmission speed equal to the number of signal events per second. See also bit rate and bits per second,

See B-channel.

A mode of operation in which the communications system control unit is connected to (is “behind”) another communications system.

An electrical representation of quantities or symbols expressed in the base-2 number system.

See B8ZS.

bipolar 8 zero substitution bipolar signal bipolar violation

BIS bit bit rate bits per second

A digital signal in which pulses (1's) alternate between

positive and negative. See also AMI, B8ZS, and bipolar

violation.

A condition that occurs when two positive or two negative

pulses are received in succession. See also AMI, B8ZS,

and bipolar signal.

Built-in speakerphone

(Binary digit) One unit of information in binary notation, having two possible values: zero or one.

The speed at which bits are transmitted, usually expressed in bits per second. Also called data rate. See also baud rate and bits per second.

(bps) The number of binary units of information that are transmitted or received per second. See also baud rate and bit rate.

Glossary GL-3

Glossary blocking

BMI

BRI bus button byte calling group

CAS

CAT

CCITT

CCS central office

Centrex

A condition in which end-to-end connections cannot be made on calls because of a full load cm all possible services and facilities.

Broadcast Music Incorporated

(Basic Rate Interface) A standard ISDN frame format that specifies the protocol used between the communications system and a terminal. BRI runs at 192 kbps and provides two 64-kbps voice or B-channels and one 16-kbps signaling or D-channel per port. The remaining 48 kbps are used for framing and D-channel contention.

A multiconductor electrical path used to transfer information over a common connection from any of several sources to any of several destinations.

A key on the face of a telephone or console that is used to access a line, activate a feature, or enter a code on a communications system.

A sequence of bits (usually eight) processed together.

"Octet" is used instead of “byte” in CCITT (International

Telegraph and Telephone Consultative Committee) documentation.

A team of agents who answer the same types of calls.

(Call Accounting System) A DOS or UNIX-based application that monitors and manages telecommunications costs.

(Call Accounting Terminal) A stand-alone unit with a built-in microprocessor and data buffer that provides simple call accounting at a low cost.

International Telegraph and Telephone Consultative

Committee.

(Common-channel signaling) Signaling in which one channel of a group of channels carries signaling information for each of the remaining channels, permitting each of the remaining channels to be used to nearly full capacity. In the system’s DS1 module, channel 24 can be designated as the signaling channel for channels 1 to 23 by selecting

“common channel” for emulated service when programming the system. CCS must be PRI service.

See CO.

A set of communications system features a user can subscribe to on telephone lines from the local telephone company.

GL-4 Glossary

Glossary channel

CSU

A telecommunications transmission path for voice and/or data.

See CSU.

Channel

Service Unit circuit-switched data call

A data call made via a connection exclusively established and maintained between data stations for the duration of the data call.

The operation of digital facilities from a common clock.

clock synchronization

CMS

CO codec commonchannel signaling communications system

(Call Management System) A DOS-based application that simulates the actions of a system operator by answering and distributing calls. CMS also produces management reports for call analysis.

(Central office) The location of telephone switching equipment that provides local telephone service and access to toll facilities for long distance calling.

(Coder-decoder) A device used to convert analog signals such as speech, music, or television to digital form for transmission over a digital medium and back to the original analog form.

See CCS.

control unit

The software–controlled processor complex that interprets dialing pulses, tones, and/or keyboard characters and makes the proper interconnections both inside and outside the system. The communications system itself consists of a digital computer, software, a storage device, and carriers with special hardware to perform the actual connections. A communications system provides voice and/or data communications services, including access to public and private networks, for telephones and data terminals on a customer’s premises.

The housing, carriers, power supply, processor, and modules of a communications system.

See modem pool.

conversion resource

CRC (Cyclic redundancy check) An error-detection code used on DS1 facilities with the ESF format.

(Channel Service Unit) Equipment used at a customer’s premises to provide DS1 facility terminations and signaling compatibility.

Glossary GL-5

Glossary cyclic redundancy check

D4 framing format

D-channel data channel data communications equipment data hunt group data rate data terminal

See CRC.

A framing format consisting of a sequence of individual frames of 24 eight-bit slots and one signal bit ( 193 bits) in a

12-frame superframe. See also ESF.

The 16-kbps or 64-kbps channel carrying signaling or data on a Basic Rate Interface (BRI) or 64–kbps channel carrying signaling or data on a Primary Rate Interface (PRI).

See D-channel.

See DCE.

See DHG.

See bit rate.

An input/output device that can be connected to the communications system control unit via an interface.

See DTE.

data terminal equipment

DCE

DCP dedicated feature buttons

DFT

DHG

(Date communications equipment) Equipment such as modems or data modules used to establish, maintain, and terminate a connection between the communications system and DTE, such as printers, host computers, or workstations.

An AT&T proprietary protocol to transmit both digitized voice and data over the same communications link. A DCP link is made up of two 64-kbps information (1) channels and one signaling (S) channel similar to the B- and D-channels used in an ISDN.

The imprinted feature buttons on a telephone: Conf or

Conference, Drop, HFAI (Hands Free Answer on

Intercom), Hold, Mute or Microphone, Speaker or

Speakerphone, Transfer, Message, and Recall.

(Direct facility termination) A CO line/trunk that terminates directly on one or more telephones; in Hybrid/PBX mode, a

DFT cannot be part of a trunk pool.

(Data hunt group) A group of analog or digital data stations that share a common access code. Calls are connected in a round–robin fashion to the first available data station in the group.

GL-6 Glossary

Glossary dial access

See feature code.

Dialed Number

See DNIS.

Identification

Service dial-out code A code (usually a 9) dialed by single-line telephone users and multiline telephone users with System Access buttons to get an outside line.

DID

DID trunk digital

(Direct Inward Dialing) A service that transmits the called extension to the communications system from the central office and routes incoming calls directly to the called extension, calling group, or outgoing trunk pool, bypassing the system operator.

An incoming trunk that receives dialed digits from the local exchange, allowing the communications system to connect directly to an extension without assistance from the system operator.

The representation of information in discrete elements such

as off and on or 0 and 1. See also analog transmission.

See DCP.

Digital

Communications

Protocol digital switch element digital transmission

See DSE.

DIP switch

A mode of transmission in which information is first converted to digital form and then transmitted as a serial

stream of pulses. See also analog transmission.

(Dual in-line package switch) A switch on a 400EM module used to select the signaling format for tie-line transmission.

DIP switches are also used on other equipment for setting hardware options.

direct facility termination

Direct Inward

Dialing

Direct Station

Selector display buttons

See DFT.

See DID.

See DSS.

DLC

The buttons on an MLX display telephone used to access the telephone’s display.

(Direct-Line Console) An answering position used by system operators to answer calls, transfer calls, make calls, set up conference calls, and monitor system operations.

Calls can ring on any of the line buttons, and several calls can ring simultaneously (unlike the QCC where calls are

Glossary GL-7

Glossary

DNIS

DOS

DPI

DS0

DS1 sent to a common QCC queue and wait until a QCC is available to receive a call).

(Dialed Number Identification Service) A service provided by ASN that routes incoming 800 or 900 calls according to customer-selected parameters, such as area code, state, or time of call.

Disk operating system

Dots per inch.

(Digital Signal 0) A single 64-kbps voice or data channel.

(Digital Signal 1) A bit-oriented signaling interface that multiplexes 24 64-kbps channels into a single 1.544-Mbps stream.

DSE

DSS

(Digital switch element) A device in each jack on each module in the communications system control unit that interfaces with the TDM bus.

(Direct Station Selector) A 60-button adjunct that enhances the call-handling capabilities of an MLX-20L telephone or

MLX-28D telephone when used as an operator console.

DTE (Date terminal equipment) The equipment that makes up the endpoints in a connection over a data circuit, for example, a data terminal, host computer, or printer.

DTMF signaling (Dual-tone multifrequency signaling) Touch-tone signaling from telephones using the voice transmission path. The code for DTMF signaling provides 12 distinct signals, each composed of two voice–band frequencies.

SEE DTMF signaling Dual-tone multifrequency signaling

E&M signaling Trunk supervisory signaling, used between two communications systems, in which signaling information is transferred through two-state voltage conditions (on the

E&M leads) for analog applications and through two bits for

digital applications. See also tie trunk.

EIA (Electronic Industries Association) A trade association of the electronics industry that establishes electrical and functional standards.

EMI endpoint

ESD

Electromagnetic interference.

The final destination in the path of an electrical or telecommunications signal.

Electrostatic discharge.

GL-8 Glossary

Glossary

ESF expansion carrier

(Extended superframe format) A framing format consisting of individual frames of 24 eight-bit slots and 1 signal bit

(193 bits) in a 24-frame extended superframe. See also D4 framing format.

A carrier added to the control unit when the basic carrier cannot house all the modules needed. An expansion carrier houses a power supply module and up to six additional line/trunk and station modules.

See ESF.

extended superframe format facility factory setting fax feature feature code frame frequency generator

FX gateway general purpose adapter glare

The equipment constituting a telecommunications path between the communications system and the central office.

The default state of a device or feature if the user does not choose an optional setting.

(Facsimile) A processor the result of a process in which graphic material is scanned and the information converted into electrical signal waves to produce an exact likeness.

A function or service provided by a hardware or software product.

A code entered on a dialpad to activate a feature. For example, a user might press the Feature button and dial 33 or might dial

#33.

One of several segments of an analog or digital signal that has a repetitive characteristic. For example, a DS1 frame consists of a framing bit and 24 octets, which equals 193 bits.

A circuit pack added to the power unit module that generates a high-voltage, 20-30 Hz signal to ring a telephone. Also called a ring generator.

(Foreign exchange) A central office (CO) other than the one providing local access to the public network.

A workstation on a local area network (LAN).

See GPA.

GPA

The loud dual-tone multifrequency (DTMF) signal an incoming caller hears when another caller tries to call out on a line/trunk at the same time the call is coming in on that line/trunk.

(General Purpose Adapter) A device that connects an analog multiline telephone to optional equipment, such as an answering machine or a fax machine.

Glossary GL-9

G l o s s a r y ground-start trunk

Group IV (G4)

Fax machine

GS trunk headset

HFAI

HFU

Hybrid/PBX mode

See GS trunk.

A fax unit offering 400x 100 dots per inch (DPI) in fine mode, which can operate at any speed for communication with a Group Ill (G3) Fax machine or another Group IV (G4)

Fax machine,

(Ground-start trunk) A trunk on which the communications system, after verifying that the trunk is idle (no ground on tip), transmits a request for service (puts ground on ring) to a distant central office (CO).

An ultralight earpiece and microphone for hands–free telephone operation.

Hands Free Answer on Intercom

Hands-free Unit

A mode of operation in which the communications system uses trunk pools and Automatic Route Selection (ARS) in addition to personal lines—that is, direct facility terminations on line buttons. The Hybrid/PBX mode also provides a single interface to users for both internal and external calling.

A tie trunk on which no start signal is necessary and dialing can begin immediately after the tie trunk is seized.

See robbed-bit signaling.

immediate-start tie trunk in-band signaiing

In-Range Outof-Building protector inside dial tone

Inspect screen

See lROB protector.

integrated

Solution II/III integrated Voice

Power

Automated

Attendant interface

A tone the user hears when connected to an Intercom line.

A display screen on digital telephones that allows users to preview incoming calls and see a list of the features programmed on line buttons.

See IS II/IS Ill.

An IS II/III application that automatically answers incoming calls with a recorded announcement and directs callers to a department, an extension, or the system operator.

l/O device

Hardware, software, or both that links systems, programs, or devices.

(input/output device) Equipment that can be attached to a computer internally or externally for managing a computer system’s input and output of information.

GL-10 Glossary

Glossary

IROB

IS II/III jack kbps

Key mode

LAN

LED line and trunk assignment line coding line compensation line/trunk and station module local area network local host computer access logical ID

(in-Range Out-of-Building protector) A surge protection device for off-premises telephones at a location within 1000 feet (305 meters) of cable distance from the communications system control unit.

(Integrated Solution II/III) A UNIX-based platform of applications for improving voice and data communications and automating office operations.

A device, accessed by inserting a plug, that is used to terminate the permanent wiring of a circuit.

Kilobits per second

A mode of operation in which the communications system uses direct facility terminations on line buttons with a separate path for internal calling.

(Local area network) A networking arrangement designed for a limited geographical area.

(Light-emitting diode) A semiconductor device that produces light when voltage is applied. LEDs show the operational status of hardware components, the results of maintenance tests, the alarm status of circuit packs, and the activation of telephone features.

The assignment of lines and trunks connected to the communications system control unit to specific buttons on each telephone.

The pattern data assumes as it is transmitted over a communications channel,

An adjustment for the amount of cable loss in decibels

(dBs), based on the length of cable between a DS1 module and a Channel Service Unit (CSU) or other far-end connection point.

A module on which the jacks for connecting central office

(CO) lines/trunks and/or the jacks for connecting the stations are located.

See LAN.

A method for connecting a station jack to an on-site computer for data-only calls through a modem or data module.

A numbering sequence used to identify station and line/trunk locations on the communications system control unit.

Glossary GL-11

Glossary loop-start line/trunk

LS line/trunk

Magic On Hold

Mbps

Megacom

Megacom 800

MERLIN

Attendant

MERLIN MAIL

Voice

Messaging

System

MFM

MLX-lO/MLX-

10D telephone

MLX-20L telephone

MLX-28D telephone mode codes modem

See LS line/trunk.

A line/trunk on which a closure between the tip and ring leads is used to originate or answer a call. High-voltage 20-

Hz AC ringing from the central office (CO) signals an incoming call.

A customized Music-on-Hold system enhancement that promotes the customer’s products and services.

Megabits per second

AT&T’s tariffed digital WATS offering for outward calling.

AT&T’s tariffed digital 800 service for inward calling.

An application with equipment that connects to one or more tip/ring station ports and automatically answers incoming calls with a recorded announcement. In response to touchtone digits dialed by the caller, MERLIN Attendant directs the caller to a department, an extension, or the system operator.

An application that provides automated attendant, call answering, and voice-mail services on the communications system.

(Multi-Function Module) An adapter that provides a tip/ring interface for the connection of optional equipment such as answering machines, external alerts, and fax machines to an MLX (digital) telephone. The optional equipment and the

MLX telephone operate simultaneously and independently.

The MFM is installed inside the MLX telephone.

A 10-button telephone offered with or without a 2-line by

24-character menu-driven display.

A telephone with 20 programmable line or feature buttons and a 7-line by 24-character menu-driven display.

A telephone with 28 programmable line or feature buttons and a 2-line by 24-character menu-driven display.

Streams of touch-tone codes used by voice messaging applications to communicate with the communications system’s control unit.

A device that converts digital data signals to analog signals for transmission over telephone lines. The analog signals are converted back to the original digital signals by another modem at the other end of the line.

GL-12 Glossary

Glossary modem pool module

Multi-Function

Module multiplexing

Music-on-Hold network network interface

Off–premises telephone ones density

OPT outcalling out-of-band signaling parity

A pair, or group of pairs, of modems and data modules with interconnected RS-232 interfaces that converts digital signals to analog, or analog signals to digital, thereby allowing users with digital data stations to communicate with users who have analog stations.

A module in the control unit provides the capability to connect central office trunks and/or telephones to the system.

See MFM.

A process in which a transmission channel is divided into two or more channels, either by splitting the frequency band into a number of narrower bands or by dividing the channel into successive time slots.

Magic On Hold or a customer-provided music source connected to the communications system via a loop-start jack, Most Music-on-Hold equipment is designed for loopstart operation.

A configuration of communications devices and software connected for information interchange.

Hardware, software, or both that links two systems in an interconnected group of systems, for example, between the local telephone company and a PBX.

See OPT.

The requirement for channelized DS1 service to the public network that eight consecutive zeros cannot be in a digital data stream.

A telephone located in a building other than where the control unit is located.

A feature of the MERLIN MAIL Voice Messaging System application. When outcalling is activated, the user is automatically called by the system at a programmed number when a new message is received in the user’s mailbox.

Signaling that uses the same path as voice-frequency transmission and in which the signaling is outside the band used for voice frequencies.

The addition of a bit to a bit string so that the total number of 1s is odd or even. Parity can be used to detect and correct transmission errors.

Glossary GL-13

Glossary

PBX

PC

PCM personal line pool port power supply module

PRI

Private branch exchange

Personal computer

Pulse code modulation.

A central office line that rings only at the user’s telephone.

On a Hybrid/PBX system, a grouping of outside trunks that users can choose with multiple pool buttons or by dialing access codes on a System Access button on the telephone.

Pools are also used by the Automatic Route Selection (ARS) feature to choose the least expensive method to route a call.

A point of access into a communications system, computer, network, or other electronic device.

A device that directs electricity to modules and telephones on the communications system. One power supply module is needed for each carrier, and an auxiliary power unit is added if the module exceeds capacity.

(Primary Rate Interface) A standard interface that specifies the protocol used between two or more communications systems. PRI runs at 1.544 Mbps and, as used in North

America, provides twenty–three 64-kbps B-channels (voice or data) and one 64–kbps D-channel (signaling). The Dchannel is the 24th channel of the interface and contains multiplexed signaling information for the other 23 channels.

See PRI Primary Rate

Interface prime line processor module protocol public network

QCC

An individual extension number assigned to a telephone in a Behind Switch System. Each telephone user has a prime line and is automatically connected to that line upon lifting the handset.

The module in the second slot of the basic carrier that contains the software that runs the communications system.

A set of conventions governing the format and timing of message exchanges between devices, such as an analog multiline telephone and the communications system control unit.

A network that is commonly accessible for local or longdistance calling. Also called public switched telephone network (PSTN).

(Queued Call Console) An answering position available to

MLX-20L telephone users only in the Hybrid/PBX mode.

The QCC is used by system operators to answer and direct

(transfer) calls, serve as a message center, and monitor system operation. Calls are sent to a common QCC queue

GL-14 Glossary

Glossary

Queued Call

Console

RAM robbed-bit signaling

ROM

RS–232

SAA

SDN

SID signaling simplex signaling single-line telephone

SMDR

SMDR printer

Software

Defined

Network where they wait until a QCC is available to receive a call

(unlike the DLC where calls can ring on any of the line buttons, and several calls can ring simultaneously).

See QCC.

(Random access memory) Computer memory in which an individual byte or range of bytes can be addressed and read or changed without affecting other parts of the memory.

Signaling in which the least significant bit of every sixth frame per channel is used for signaling in that channel.

(read-only memory) Computer memory that can be read but cannot be changed.

A physical interface, specified by the EIA (Electronics

Industries Association), that transmits and receives asynchronous data at speeds of up to 19.2 kbps over cable distances of 50 feet.

(Supplemental alert adapter) A device that permits 48 VDC alerting equipment to be connected to an analog multiline telephone jack so that people working in noisy or remote areas of a building can be alerted to incoming calls.

(Software Defined Network) An AT&T private networking service created by specialized software within the public network.

Station ID.

The sending of control and status information between devices to set up, maintain, or take down a connection.

The transmission of signals in one direction only across a telecommunications channel.

An industry-standard touch-tone or rotary telephone that only handles one trunk and is connected to the communications system via a jack on a basic telephone module.

(Station Message Detail Recording) Captures detailed usage information on incoming and outgoing voice and data calls.

A printer used for Station Message Detail Recording

(SMDR) that is connected to the communications system via an RS-232 jack on the processor.

See SDN.

Glossary GL-15

Glossary

SPM

Square Key

SSN station station jack

Station

Message Detail

Recording

Supplemental

Alert Adapter switched service network switchhook flash synchronous data transmission system date

(System Programming and Maintenance) A DOS-or

UNIX-based application for programming and maintaining the communications system.

A way of configuring the communications system in Key mode so that all lines appear on all telephones.

(Switched service network) A network consisting of terminals, transmission lines, and at least one exchange on which a user can communicate with any other user at any time.

The endpoint on the internal side of the communications system, A station can be a telephone with or without an adjunct or can be a data terminal with a modem (analog) or a 7500B Data Module (digital) attached.

An analog, digital, or tip/ring interface on the control unit module for connecting telephones and other equipment.

See SMDR.

See SAA.

See SSN.

Operation of the telephone switchhook in which the on–hook period is in the range of 250–500 ms.

A method of transmitting a continuous digital data stream in which the transmission of each binary bit is synchronized with a master clock.

The date that appears on MLX display telephones and

Station Message Detail Recording (SMDR) reports.

See SPM.

System

Programming &

Maintenance system renumbering system time

T1

A process used to change the extension numbers assigned to telephones, adjuncts, calling groups, paging groups, Call

Park Zones, Remote Access, and lines/trunks.

The time that appears on MLX display telephones and is printed on Station Message Detail Recording (SMDR) reports.

A digital transmission carrier path that in North America transmits at the DS1 rate of 1.544 Mbps.

GL-16 Glossary

Glossary

TDM (Time-division multiplexing) A process by which the transmission channel is divided.

Equipment that provides power to an individual telephone.

telephone power supply unit tie trunk A private line directly connecting two communications systems.

See TDM.

Time-division multiplexing timer tip/ring

A built-in timing device in a display telephone.

The contacts and associated conductors of a single-fine telephone plug or jack.

See TTR.

touch-tone receiver

T/R trunk jack

TTR

See tip/ring.

A jack that connects an outside trunk to the communications system control unit.

(Touch-tone receiver) A device used to decode touch–tones dialed from single–line telephones or Remote

Access telephones.

See UPS.

Uninterruptible power supply unit load

UPS

VAC

VDC

VMI voice-band channel voice-only voice signal pair

WATS

A measure of the power load drain of a module, telephone, or adjunct,

(Uninterruptible power supply) A device that connects to the communications system to provide 117 VAC to the equipment when the commercial power source fails.

Volts AC.

Volts DC.

Voice messaging interface.

A transmission channel, generally the 300-3400-Hz frequency band.

A telephone that is set up for making and receiving voice calls but not data calls.

A pair of leads on an analog multiline telephone used for the Voice Announce to Busy feature.

(Wide Area Telecommunications Service) A service that allows calls to certain areas for a flat–rate charge based on expected usage.

Glossary GL-17

Glossary wink-start tie trunk

ZCS

A tie trunk on which the originating end transmits an offhook signal and waits for the remote end to send back a signal (a wink) that it is ready for transmission.

(zero code suppression) A binary coding scheme that ensures a data stream contains at least a minimum number of information bits (1s) for receiver synchronization.

GL-18 Glossary

Index

008 module, on QCC 2-26, 5-13

008 MLX module 5-8, 5-12 - 5-13,

5-20 - 5-21

008 OPT module 5-4, 5-12

012 Modules 4-6, 4-18, 4-39, 5-4,

5-12 - 5-13

100D module 3-11, 3-16, 5-15

Services supported 5-15

10-button analog multiline

telephone 1-3

10-button HFAI analog multiline

telephone 1-3

129B Frequency Generator 2-4

146A surge protector 1-41


2500MMGB single-line telephone

1-3

2500MMGJ single-line telephone 1-3

2500MMGK single-line telephone

1-3

2500SM single-line telephone 1-3

2500YMGK single-line telephone 1-3

2514BMW single-line telephone 1-3

2526BMG single-line telephone 1-3

2554MMGJ single-line telephone 1-3

25-pin connector 1-30

329A wall telephone power unit with

MFM 2-35


telephone 1-3

391A1 power supply 1-41, 2-10

400 modules 4-7

400EM module 3-4 - 3-5

408 GS/LS-MLX module 2-9, 2-26,

5-8, 5-14, 5-20 - 5-21

408 MLX module 5-8, 5-1 2 - 5-14

440AY terminator resisting

5-1 2 - 5-1 3

486SX Microprocessor 4-61

500MM single-line telephone 1-4

500SM single-line telephone 1-4

551 T1 L1 CSU 2-31

554BMPA single-line telephone 1-4


telephone 1-2

6386/SX WGS 4-59

705 Multi-tasking terminal 4-59

7101A single-line telephone 1-3

7102A single-line telephone 1-3

7500B Data Module 1-4, 2-37 - 2-41,

5-2, 5-4, 5-6 - 5-9, 5-1 2 - 5-1 3,

5-19, 5-25

High-Speed Synchronous Interface

Enhancement Board 2-40

Multipurpose Enhancement

Board 2-40

800 DID trunks module (see also

Direct Inward Dialing trunks)

3-11 - 3-13

800 GS/LS 5-14

8102 single-line telephone 1-3

8110 single-line telephone 1-4

9-pin connector 1-30

9-pin to 25-pin adapter 1-30

A

AC power outlet 1-9

Accessories, power 2-51

Account Code Entry 4-21 - 4-22,

4-24, 4-36, 4-45, 5-16

Accunet switched digital service

3-17, 5-15

Adapter, 9-pin to 25-pin 1-30

Adapters 1-4, 2-28, 2-31, 2-43

7500B Data Module 1-4,

2-37 - 2-41 , 5-4 - 5-9

Channel Service Unit (CSU) 2-31


(GPA) 1-4, 2-35, 5-4

Headset 2-46 - 2-49

Loop-Start Trunk Adapter 2-33

Multi-Function Module (MFM) 1-4,

2-34, 5-4 - 5-5

Modem 1-4

Not supported 2-50


(SAA) 1-5, 2-42

System 1-4, 2-31 - 2-33

Telephone 1-4, 2-33, 2-43

Universal Paging Access Module

(UPAM) 2-33

Index IN-1

Index

Adapters and adjuncts 2-28 - 2-30

Add digits for Direct Inward Dialing

(DID) trunks 3-13

Address, I/O 1-31

ADDS (see Automated Document

Delivery System)

Adjuncts 1-4, 2-28 - 2-30, 2-43

Delay announcement 2-46

Dial dictation 2-45

Fax machine 2-46

Headsets 2-46 - 2-49

Loudspeaker Paging system 2-44

Music-on-Hold 2-45

Not supported 2-23, 2-50

SMDR printer 2-44

Speakerphones 2-49

Specialty handsets 2-50

SPM PC 2-44

System 2-44 - 2-45

Unit loads 1-42

used with MFM 2-34 - 2-35

Agent Splits 4-27

Alarm

PFT 2-58

Power failure 2-58

System 2-57

Trouble 2-58

Alarm button, on QCC 2-25

Allowed Lists, System

capacities 1-34

AMI (see alternate mark inversion)

Alternate mark inversion 3-19

Analog data stations 1-7, 5-4 - 5-6

Configurations 5-5

Tip/ring interface 5-4

Analog multiline telephones 1-2 -

1-3, 1-7, 2-21

Analog tie-trunk connection 3-3

Announcement Service with MERLIN

MAIL Voice Messaging

System 4-11

ANI (see Automatic Number

Identification)

Answer Delay 4-26

Answer supervision time for tie

trunks 3-4

Applications 1-5, 1-8, 4-1

Automated Document Delivery

System (ADDS) 4-58 - 4-59

Call Accounting System

(CAS) 4-20 - 4-22

Call Accounting Terminal

(CAT) 4-23 - 4-25

Call Management System

(CMS) 4-26 - 4-29

Capacities 4-3

Centrex 4-52 - 4-55

CONVERSANT Intro 4-60 - 4-61

Group IV (G4) Fax 4-50, 5-18 -

5-24

InnManager Guest Management

System 4-30

Integrated Solution II (IS II) 4-33 -

4-39

Integrated Solution III

(Is Ill) 4-40 - 4-49

MERLIN Attendant 4-16 - 4-19

MERLIN MAIL Voice Messaging

System 4-9 - 4-15

MERLIN PFC Telephone 4-56 -

4-57

Primary Rate Interface

(PRI) 4-50 - 4-51

Printers 4-62

System Programming and

Maintenance (SPM) 4-31 - 4-32

Video Conferencing 4-50 - 4-51,

5-25 - 5-26

Voice messaging systems 4-4 -

4-8

ARS (see Automatic Route

Selection)

ASN services 3-16

Asynchronous transmission 5-19

AT&T Switched Network (ASN)

services 3-16 - 3-17

Audio Subsystem 4-51

AUDIX Voice Power-lS II 4-33 - 4-34

AUDIX Voice Power-lS Ill 4-40 -

4-41

Auto Answer All 5-16

Auto Dial buttons 5-16

Auto Logout 4-18

Autobaud 2-42

Automated Attendant,

MERLIN MAIL Voice

Messaging 4-9 - 4-10

IN-2 Index

Index

Automated Document Delivery

System (ADDS) 4-58 - 4-59

Automatic Callback 5-17

Automatic Number Identification (ANI) with Call Accounting Terminal

4-23, 5-15

Automatic Route Selection (ARS)

1-14, 4-53 5-16

Automatic signaling type for tie

trunks 3-5

Auxiliary power units 2-52 - 5-53

Built-in modem 1-11, 1-30

Bus

Input/output in control unit 1-11,

2-2

Time-division multiplexing in control

unit 1-11

Busy-hour calls 4-15, 4-19, 4-35

Busy-out, Power failure for DID

trunks 2-58

Buttons, Fixed feature on QCC

2-25 - 2-26

B

B8ZS line coding 3-19

Backboard for control unit

Dimensions 1-37

Mounting hardware 1-38

Backplane assembly in control

unit 1-9, 2-2

Backup 1-10

Battery power 2-3, 2-51

RAM 2-3

Barge-in 1-31

Basic carrier 1-2, 1-7

Dimensions 1-37

in control unit 2-1 - 2-2

Basic Rate Interface 5-20

Basic Telephone and Modem

Interface 4-58

Battery Backup 2-3, 2-51

Battery processor 2-3

B-channels 3-1 - 3-14, 3-18, 5-16

Behind Switch mode 1-22

FCC registration 1 -27 - 1 -29

Features not available 1-25

Host system features 1-27

Trunk access 1-26

Types of trunks 1-24 - 1 -25

Bipolar mark inversion 3-19

BIS telephones 1-3, 2-21

BRI (see Basic Rate Interface)

Broadcast Lists with MERLIN MAIL

Voice Messaging System 4-11

BTMI (see Basic Telephone and

Modem Interface)

C

Call Accounting and Rating System

(CARS) 4-30

Call Accounting System (CAS)

4-20 - 4-22

Call Accounting Terminal (CAT)

4-23 - 4-25

Call Answer Service

AUDIX Voice Power 4-33, 4-40


System 4-10

Call button, on QCC 2-25

Call Management System (CMS)

4-26 - 4-29

Call Restrictions 5-16

Call Waiting 5-17

Cancel button, on QCC 2-25

Capacities

Applications 4-3

Hardware 1-33 - 1-36

Software 1-33 - 1-36

System 1-33 - 1-36

Time-slot 1-33

Carriers 1-7, 2-1

Basic 1-2, 1-7

Dimensions 1-37

Expansion 1-2, 2-1

CAS (see Call Accounting System)

CAS Plus (see Call Accounting

System)

CAT (see Call Accounting Terminal)

CAT Basic 4-23

CAT Plus/Business 4-23

Index IN-3

Index

CAT Plus/Hospitality 4-23

Centralized Telephone

Programming 1-31

Centrex 1-22, 1-27, 4-52 - 4-55

Channel Service Unit (CSU)

2-31 - 2-33, 3-1 9 - 3-20

Class of service 4-34

CMS (see Call Management

System)

Classmark 5-6

Clock synchronization 3-16, 3-20 -

3-21

Common-channel signaling mode for

DS1 facility 3-17, 3-21

Components 1-2

AC power outlet 1-9

Adapters 1-4, 2-28 - 2-43

Adjuncts 1-4, 2-43 - 2-50

Analog data stations 1-7

Analog multiline telephone 1-7

Applications 1-8

Carriers 1-7

Consoles 2-13, 2-24 - 2-27

Control unit 1-2, 1-7, 2-1

Digital data station 1-8

Direct Station Selector (DSS) 1-7

External alert 1-8

Fax machine 1-8


(GPA) 1-7

Line/trunk and station modules 1-7


Magic On Hold 1-7

MLC-5 Cordless analog multiline

telephone 1-8


5-4

Off-premises telephone 1-8

Power supply module 1-7

Processor module 1-7

Single-line telephone 1-8

SMDR printer 1-8

Telephones 1-2 - 1-4, 2-13 - 2-27

Connectors 1-30

Consoles 2-13, 2-24 - 2-27

DLC 2-26

QCC 2-24 - 2-25

System operator 2-24

Contamination, Airborne requirements

for control unit 1-39

Control unit 1-2, 1-7, 2-1

Backboard dimensions 1-37

Backboard-mounting hardware

1-38

Backplane assembly 1-9, 2-2

Basic carrier 1-2, 2-1 - 2-2

Carriers 1-2, 2-1 - 2-2

Cover 2-9 - 2-10

Digital switching 1-11

Dimensions of carrier 1-37

Environmental requirements

1 -37 - 1 -39

Expansion carrier 1-2

for data communications 5-2

Functional units 1-9, 1-11

Input/output bus 1-11, 2-2

Line/trunk and station modules

2-4, 2-6 - 2-9

Location 1-37 - 1-38

Mean time between failures 1-37

Mean time between outages 1-37

MERLIN II Release 3 reusable

modules 2-10 - 2-12

Power requirements 1-39 - 1-43

Power supply module 2-3 - 2-4

Processor module 2-2 - 2-3

RS-232 jacks 2-3

SMDR jack 2-3

SPM jack 2-3

Coverage 3-15, 4-6, 4-12, 4-17,

4-36, 4-45

CONVERSANT Intro 4-60

Conversion resources 5-2

Cover 2-9 - 2-10

CS6402U01A single-line

telephone 1-3

CSU (see Channel Service Unit)

IN-4 Index

Index

D

D4 Framing Format 3-18 - 3-19

Data communications 5-1

Configuration 5-2 - 5-13

Connectivity to LAN 5-13

Connectivity to local host

computer 5-13

Control unit connectivity 5-2

Data Hunt Group 5-2, 5-9 - 5-10

Data stations 5-3 - 5-9

Endpoint features supported 5-18

Equipment 5-4

Features to be disabled 5-17

Metering 5-18

Modem pool 5-10

Outside trunks 5-13 - 5-16

Primary Rate Interface (PRI)

5-15 - 5-16, 5-18 - 5-26

Resource pools 5-12

System features used 5-16 - 5-17

Video conferencing application

5-25 - 5-26

Data hunt group

(DHG) 3-15, 5-2,

5-9 - 5-10

Modem pool 5-10

configurations 5-12

data call direction 5-11 dialing 5-11

Shared resource assignments 5-10

Data stations 5-4 - 5-9

7500B Data Module 5-2 - 5-3, 5-4,

5-6 - 5-9

Analog 5-4 - 5-6

Analog, off-premises data-only 5-5

Analog, tip/ring interface 5-4

Configurations 5-3, 5-9

Digital 1-8, 5-6 - 5-8

Off-premises analog data-only 5-5

Synchronous high-speed digital

2-41, 5-24 - 5-25

Data Status button 5-16

Data Terminal Equipment 5-4, 5-20

DB-9 connector 1-30

DB-25 connector 1-30

DCE (see Data Communication

Equipment)

Delay announcement 2-46

Delay signaling type for tie trunks

3-5

Delete digits for Direct Inward Dialing

(DID) trunks 3-13

Destination button, on QCC 2-25

DFT (see Direct Facility Termination)

DHG (see Data hunt group)

Dial access to DFT 5-16

Dial by name 4-34

Dial dictation 2-45 - 2-46

Dial mode for tie trunks 3-4

Dial tone 1-15

Dial tone for tie trunks 3-4

Dialed Number Identification Service

(DNIS) 3-17

Dial-repeating tie trunks (see wink,

immediate, or delay signaling type

for tie trunks)

DID trunk (see Direct Inward Dialing trunks)

Digital data stations 1-8, 5-6 - 5-8

Digital emulated tie-trunk

connection 3-3

Digital (MLX) telephones 2-14 - 2-18

Digital switch element (DSE), on

modules 1-9

Digital switching in control unit

1-11 - 1-12

Digits

Add for Direct Inward Dialing (DID)

trunks 3-13

Delete for Direct Inward Dialing

(DID) trunks 3-13

Expected for Direct Inward Dialing

(DID) trunks 3-13

DIP switches and tie-trunk

signaling 3-3 - 3-10

Direct Facility Termination 5-16

Direct Inward Dialing (DID) 1-14,

2-5, 3-11 - 3-15, 4-56, 5-14 - 5-15

Power failure busy-out 2-58

Release 2.0 functionality 3-11


Buttons, programmable

2-19 - 2-20

Description 2-19 - 2-20

Message Status button 2-20

Index IN-5

Index

on QCC 2-26

Page button 2-20

with digital DLC 2-27

with MLX-20L telephone with MLX-28D telephone

Direction for tie trunks 3-4

Direct-Line Console (DLC)

2-27

2-27

2-24,

2-26 - 2-27

Directory Look up 4-21

Disconnect time for tie trunks 3-4

Dissipation, Heat requirements for

control unit 1-38

DNIS (see Dialed Number

Identification Service)

DOS version 1-30

DS0 channels 3-16, 3-19, 5-14

DS1

ones-density requirement 3-19

In-band signaling 3-21

facilities 1-14, 1-18, 1-24,

3-16 - 3-17, 5-14 - 5-16, 5-24

framing formats and signaling

modes 3-18 - 3-19

modules used 5-15, 5-21

options and factory settings 3-16,

3-18 - 3-22


T1 2-5, 3-18, 5-15

DS1 trunks 2-5

DSE (see Digital switch element)

DSS (see Direct Station Selector)

DTE (see Data Terminal Equipment)

E

EIA Interface 5-19, 5-23

EIA RS-232 5-4, 5-13

Electrical fields requirements for

control unit 1-30

Electromagnetic interference (EMI)

filter 2-57

Electrostatic discharge (ESD)

suppression kit 2-57

E&M signaling 3-3, 3-5 - 3-9

Environmental requirements, Table of

specifications 1-37 - 1-38

Environmental requirements for

control unit 1 -37 - 1-39

ESF, T1 CSU 2-31, 3-19, 5-25

Expansion carrier 1-2, 2-1 - 2-2

Dimensions 1-37

Expected digits for Direct Inward

Dialing (DID) trunks 3-13

Extension Directory, IS Ill 4-41 - 4-43

Extension Programming 1-31

Extension Status 4-28

External alert 1-8, 4-27

F

Far-end disconnect, VMI ports 4-8

FAX Attendant System 4-41

Fax board ports 4-47

Fax Call Answer, FAX Attendant

System 4-41

Fax machine 1-8, 2-46, 5-10

Fax Mail, FAX Attendant System

4-41

Fax MERLIN PFC 4-56 - 4-57

Fax on demand 4-58

Fax Response, FAX Attendant

System 4-41, 4-58

Fax Transfer, Integrated

Solution II 4-34

Fax Transfer with MERLIN MAIL Voice

Messaging System 4-11

FCC registration 1-27 - 1-29

Features, System capacities 1-33 -

1-36

Federal Communications Commission

(see FCC)

Forced Release button, on QCC

2-26

Foreign exchange trunk (see FX trunk)

Framing formats and signaling modes

for DS1 facilities 3-21 - 3-22

Frequency generator 2-4

Functional description of system

1-9 - 1-12

Functional units 1-9 - 1-11

FX trunk 1-14, 1-18, 1-24

IN-6 Index

Index

G

General Purpose Adapter (GPA) 1-4,

1-7, 2-35 - 2-37, 4-58, 5-4

Generic VMI port 4-6

Glare 3-1 - 3-2, 4-54, 5-14

GPA (see General Purpose

Adapter)

Grounding and power

requirements 1-39 - 1-42

Grounds, Approved 1-40 - 1-41

Ground-start button for PFT 2-60

Ground-start lines/trunks 3-2

Ground-start pool 1-20

Ground-start trunk 1-14, 1-18, 2-5,

5-13 - 5-14

Group Calling 4-12, 4-17, 4-28, 4-36,

4-45

Group Ill Fax machine 4-59, 5-6

Group IV (G4) Fax 4-50, 5-18 - 5-24

H

Handsets, Specialty 2-50

Hands-free Unit (see

Speakerphone)

Hardware capacities 1 -33 - 1 -36

Hardware components

Adapters 2-31 - 2-43

Adjuncts 2-43 - 2-50

Table of 2-28 - 2-30

Telephones and consoles 2-13 -

2-27

Headset Auto Ans button 2-25

Headset Hang Up 1-31

Headset Mute button 2-25

Headset Status button 2-25

Headsets

Amplifiers 2-47

Hardware description 2-46 - 2-49

Heat dissipation requirements for

control unit 1-38

HFU (see Speakerphone)

High-Speed Synchronous Interface

Enhancement Board for 7500B

Data Module 2-41, 5-24 - 5-25

I

Humidity requirements for control

unit 1-38

Hybrid Key system 1-17

Hybrid/PBX mode 1-18 - 1-22


QCC 1-21

Trunk access 1-20

Trunk pools 1-18, 1-20

Types of trunks 1-18

ICOM buttons 1-13, 1-15 - 1-16

Idle line preference 5-17

lmmediate-start signaling type for tie

trunks 3-5

Immediate-start type for Direct Inward


In-band signaling for DS1 facility

3-21

Incoming only direction for tie

trunks 3-4

Information Service, AUDIX Voice

Power-lS III 4-40

InnManager Guest Management

System 4-30

Call Accounting and Rating System

(CARS) 4-30

In/out of service, VMI ports 4-8

Input/output bus, in control unit 2-2

Input/output bus in control unit 1-11,

2-2

Inside dial tone 4-14, 4-47

Integrated Administration 4-42 - 4-43

Integrated Solution II (IS II)

4-33 - 4-39

Integrated Solution Ill (IS Ill) 4-40 -

4-49

Intercom button (see ICOM button)

Interrupt 1-31

I/O address 1-31

l/O bus, in control unit 2-2

IROB protection units 2-13, 2-53 -

2-55

IS II (see Integrated Solution II)

IS Ill (see Integrated Solution Ill)

Index IN-7

Index

IVP 4 boards 4-39, 4-61

IVP 4/6 boards 4-47

J

Join button, on QCC 2-25

Location of control unit 1-39

Loop-start lines/trunks 1-14, 1-18,

1-24, 2-5, 3-1 - 3-2, 5-13 - 5-14

Modules used 5-14

Loop-start pool 1-20

Loop-Start Trunk Adapter 2-33

Loudspeaker Paging system 1-8,

2-44 - 2-45

K

Key mode 1-13 - 1-18

FCC registration 1-28

Features not available 1-14

Trunk access 1-15

Types of buttons 1-13

Types of trunks 1-14

Key system 1-16 - 1-17

KF classification 1-28

KS22911,L2 telephone power unit,

with MFM 2-35

L

Labeling 3-15, 4-45

LAN (see Local Area Network)

Last number dial 5-17

Direct Inward Dialing (DID) 3-11 -

3-15

Ground-start 3-2

Loop-start 3-1 - 3-2

Leave word calling 4-12, 4-36, 4-45

Line button 1-13

Line coding 3-19 - 3-20

Line compensation 3-20

Lines/trunks 1-12

Line/trunk and station modules 1-7,

2-6 - 2-9

Hardware description 2-4

Lists, Allowed, system capacities

1-34

Loads, Unit 1-42

Local area network 5-2, 5-13

Local setting for tie-trunk dial

tone 3-4

M

Magic On Hold 1-7

Mean time between failures for control unit

1-39

between outages for control

unit 1-39

Megacom 800 AT&T Switched

Network (ASN) service 3-17

Megacom WATS AT&T Switched

Network (ASN) service 3-17



modules 2-10 - 2-12

MERLIN II System Display

Console 2-27


System 4-9 - 4-15

MERLIN PFC Telephone 1-3, 4-56 -

4-57

Message Drop, AUDIX Voice

Power-lS III 4-40

Message Status button, on DSS 2-20

Message-waiting indicator, on

single-line telephone 2-50

MF classification 1-29

MFM (see Multi-Function Module)


telephone 1-3, 1-8

MLX telephones 1-2, 2-1 4 - 2-18

MLX-10 telephone 1-2, 1-8, 2-18

MLX-10D telephone 1-2, 1-8, 2-17

MLX-20L telephone 1-2, 1-8, 1-30,

2-16, 2-24

Jack connection on DLC 2-27

on DLC for system

programming 2-27

IN-8 Index

Index

MLX-28D telephone 1-2, 1-8, 2-15,

2-27

Mode, Dial for tie trunks 3-4

Mode codes, Voice messaging

systems 4-6

Modem, Built-in 1-11

Modem pool 5-10

Configurations 5-12

Data call direction 5-11

Dialing 5-11

Modems 1-4, 2-41 - 2-42

Built-in 1-11, 1-30

Features 2-42

Modes of Operation 1-12 - 1-29

Behind Switch 1-22 - 1-27


Hybrid/PBX 1-18 - 1-22

Key 1-13 - 1-18

Summary table 1-12 - 1-13

Modified Key system 1-17

Modular adapter

355A 1-30

355AF 1-30

Modular cord, 4-pair (D8W) 1-31

Modules

408 GS/LS-MLX 2-9, 5-13

Line/trunk and station

1-7, 2-4, 2-6 - 2-9


2-10 - 2-13

Power supply 2-2 - 2-4

Processor 2-2 - 2-3

Monitor 1-31

Mu-Law 255 1-11


1-8, 2-34 - 2-35, 5-4

Adjuncts used with 2-34

Multiline telephones, Jack connection

on DLC 2-27

Multipurpose Enhancement Board for

7500B Data Module 2-40, 5-22,

5-25

MultiQuest AT&T Switched Network

(ASN) service 3-17

Music coupler 2-45

Music-on-Hold 2-45, 4-28

O

N

Networking, Tie trunk 3-10

Night Service

button on QCC 2-25

feature 4-13, 4-14,

4-17, 4-36, 4-46

No Transfer with Automated

Attendant 4-10

Off-Premises Range Extender

(OPRE) 2-13, 2-55 - 2-56

Off-Premises Analog Data-Only 5-5

Off-Premises Telephone (OPT) 1-8

Ones-density requirement for DS1

facility 3-19

OPRE (see Off-Premises Range

Extender)

OPT (see Off-Premises Telephone)

Optical scanner 5-6

Outcalling with MERLIN MAIL Voice


Outgoing only direction for tie

trunks 3-4

Outside dial tone 1-15

P

Page button, on DSS 2-20

PagePac paging system 2-33, 2-45

Pager Notification with MERLIN MAIL


Paging, Loudspeaker, hardware

description 2-44

Parallel port 1-31

PC, Requirements for system

programming 1-30 - 1-30

Personal line button 1-21

Personal speed dial 5-17

PF classification 1-29

PFT alarm 2-58

Index IN-9

Index

PFT jack 2-60

PFT telephone 2-22, 2-60

Pool access 5-17

Pool button 1-21

Pool Status button, on QCC 2-25

Pools

Ground-start 1-20

Hybrid/PBX mode 1-18, 1-20

Loop-start 1-20

Tie 1-20

Port signature 5-6

Position Busy button, on QCC 2-25

Positions

System operator 2-24, 2-27

System operator, types of

telephones used 2-24, 2-27

Power

129B frequency generator 2-57


147A surge protector 1-41 - 1-42,

2-56

329A telephone power unit 2-35,

2-52

391A1 power supply 1-41 - 1-42

Accessories 2-51

Auxiliary power units 2-51 - 2-53

Battery backup 2-51

Control unit requirements 1-39

EMI filter 2-57

ESD suppression kit 2-57

lROB protection 2-53 - 2-55

KS22911,L2 telephone power

unit 2-35, 2-52

Local auxiliary power

requirements 2-52

Off-Premises Range Extender

(OPRE) 2-55 - 2-56

Power-related hardware

2-51 - 2-60

Protection accessories 2-53 - 2-57

Ring generator 2-57

Surge protectors 1-41, 2-56 - 2-57

Telephone power units 2-51 - 2-52

WP90110,L1 power unit for 7500B

Data Module 2-39

Z77A data mounting for 7500B Data

Module 2-39

Power and grounding

requirements 1-39 - 1-43

Power failure alarm 2-58

Power failure DID busy-out 2-58

Power failure transfer jack 2-60

Power failure transfer (PFT)

telephone 2-60

Power outlet, AC 1-9

Power requirements for control

unit 1-39

Power supply, Uninterruptible 2-51

Power supply module 1-7, 2-3 - 2-4

PRI (see Primary Rate Interface)


3-16 - 3-18, 4-50, 5-1 5 - 5-16,

5-18

Group IV (G4) Fax 5-18 - 5-24

on DS1 trunk 2-5

Prime Line button 1-26

Printers

Applications 4-62

SMDR 1-8

Privacy 4-13, 4-17, 5-17

Processor module 1-7, 2-2 - 2-3

Programming 1-30 - 1-32

Centralized Telephone 1-31

Direct Inward Dialing (DID)

trunks 3-13 - 3-15

DS1 options and factory

settings 3-16, 3-18 - 3-22

PC requirements 1-30 - 1-31

System 1-30

Telephone 1-31

with built-in modem 1-30 with MLX-20L telephone 1-30 with SPM software on PC 1-30

Q

QCC (see Queued Call Console)

Queued Call Console (QCC) 1-14,

1-21, 2-24 - 2-26

R

RAM 1-30

IN-10 Index

Index

RAM backup 2-3

Recall button 4-53

Release 1.0 operation 1-14, 1-20,

3-18, 4-52

Release 1.1 enhancements 1-43 -

1-44, 4-52

Release 2.0 enhancements 1-44 -

1-46, 3-18, 4-12, 4-53

Release button, on QCC 2-25

Release differences 1-14, 1-20,

1-43 - 46

Remote access 3-15

Remote setting for tie-trunk dial

tone 3-4

Requirements

Environmental for control unit

1-37 - 1-39

Grounding and power 1-39 - 1-43

System 1-33 - 1-43

Resource pools 5-12

Ring generator, Single-line

telephones 2-4

Ringing options 4-13, 4-57

Robbed-bit signaling mode for DS1

facility 3-21

Rotary dial mode for tie trunks 3-4

Rotary signaling for Direct Inward


Rotary trunks 4-53

Routing by dial plan 3-17

RS-232 jacks in control unit 2-3

S

SA button (see System Access button)

SAA (see Supplemental Alert

Adapter)

SASS (see Shared Access for

Switched Services)

Send/Remove Message button, on

QCC 2-25

Serial port 1-30

Shared Access for Switched Services

(SASS) 3-17

Shared System Access button 1-21

Signaling

DID trunks 3-13

DS1 3-21

Tie trunks 3-7 - 3-9

Type 1 Compatible 3-8

Type 1 Standard 3-8

Type 5 3-8

Types, E&M 3-8

Types, for tie trunks 3-5, 3-8

Signaling modes for DS1 facilities

3-21

Signaling modes for tie trunks

E&M 3-8

Simplex 3-7

Simplex signaling mode for tie

trunks 3-7

Simultaneous voices data 5-2

Single-line telephones 1-3 - 1-4, 1-8,

2-22

as PFT telephone 2-60

Message-waiting indicator 2-50

PFT 2-22

Ring generator 2-4

SMDR printer 1-8, 2-44

Jack in control unit 2-3

Software capacities 1-3 - 1-36

Software Defined Network (SDN)

3-17

Source button, on QCC 2-25

Speakerphones 2-49

Specialty handsets 2-50

SPM (see System Programming and

Maintenance)

Square Key system 1-16

Start button, on QCC 2-25

Station Message Detail Recording

(SMDR) 1-8, 4-22, 4-24, 4-32,

4-37, 4-46

Fields 2-44

Jack in control unit 2-3

Supervised transfer with Automated

Attendant 4-10

Supervised transfer with MERLIN

Attendant 4-16

Supervision, Answer time for tie

trunks 3-4


(SAA) 1-5, 2-42 - 2-43

Index IN-11

Index

Surge protectors 1-41 - 1-42,

2-56 - 2-57

146A 2-56

147A 2-56

Switched Network, AT&T (ASN)

services 3-16

Switches and tie-trunk signaling 3-7

Switching, Digital, in control unit

1-11 - 1-12

System, Functional description

1-9 - 1-12

System Access button 1-14, 1-18,

1-20 - 1-21

System adapters 1-4 - 1-5, 2-31 -

2-33

System adjuncts 2-44 - 2-46

SMDR printer 2-44

System alarms 2-57 - 2-60

System capacities 1-33

System components 1-6 - 1-9

AC power outlet 1-9

Adapters 1-4 - 1-5

Analog data station 1-7

Analog multiline telephone 1-7

Applications 1-8

Carriers 1-7

Control unit 1-2, 1-7

Digital data station 1-8


External alert 1-8

Fax machine 1-8


(GPA) 1-7

Line/trunk and station modules 1-7


Magic On Hold 1-7


telephone 1-8

Multi-Function Module (MFM) 1-8

Off-premises telephone (OPT) 1-8

Power supply module 1-7

Processor module 1-7

Single-line telephone 1-8

SMDR printer 1-8

Telephones 1 -2 - 1 -4

System dial tone 1-15

System Menu Hierarchy 1-32

System operator positions 2-27

Types of telephones used 2-24

System numbering 4-28

System programming 1-30

DS1 facilities 3-16

PC requirements 1-30 - 1-31 with built-in modem 1-30 with MLX-20L telephone 1-30 with SPM software on PC 1-30

System Programming and

Maintenance (SPM) 4-31 - 4-32

IS II 4-34

IS Ill 4-41

Jack for PC in control unit 2-3

System renumbering 4-46

System requirements 1-30 -1-43

System speed dial 5-17

T

T1, on DS1 trunk 2-5, 5-15

T1 facility 1-14, 1-18, 1-25, 5-15

T1 service 3-17

Telephone adapters 1-4

7500B Data Module 1-4, 5-4,

5-6 - 5-9

GPA 1-4

MFM 1-4

Modem 1-4

SAA 1-5

Telephone programming 1-31

Centralized 1-31

Extension 1-31

Telephones 2-13 - 2-27

Adapters 2-33-2-43

Adjuncts 2-46 - 2-50

Adjuncts, DSS 2-19 - 2-20

Analog DLC 2-27

Analog multiline 1-2 - 1-3, 1-7,

2-21

BIS 1-3, 2-21, 2-27

Digital DLC 2-27

Digital (MLX) 2-14, 2-18

Digital (MLX), Features 2-14

lROB protector 2-13

MERLIN II System Display

Console 2-27

IN-12 Index

Index

MERLIN PFC Telephone 1-3, 2-21

MLC-5 Cordless analog

multiline 1-3, 1-8, 2-21

MLX 1-2

MLX-10 1-2, 1-8, 2-18

MLX-10D 1-2, 1-8, 2-17

MLX-20, Jack connection on

DLC 2-27

MLX-20, on DLC for system

programming 2-27

MLX-20L 1-2, 1-8, 1-30, 2-16, 2-24,

2-27

MLX-28D 1-2, 1-8, 2-15, 2-27

Multiline, jack connection on

DLC 2-27

Not supported 2-23

Off-premises Telephone (OPT) 1-8

OPRE 2-13

PFT 2-60

Power Units 2-51 - 2-52

QCC 2-24 - 2-26

Single-line 1-3, 1-8, 2-22

Single-line with message-waiting

indicator 2-50

Unit loads 1-42 - 1-43, 2-45

Telephone adapters 2-33 - 2-43

Tie pool 1-20

Tie trunks 1-14, 1-18, 1-25, 2-5,

3-2 - 3-10, 5-14

Time

Answer supervision for tie

trunks 3-4

Mean time between failures for

control unit 1-37

Mean time between outages for

control unit 1-37

Timed flash 4-52 - 4-53

Time-division multiplex bus in control

unit 1-11

Time-slot capacity 1-33

Tip/ring ports 4-6

Touch-tone dial mode for tie

trunks 3-4

Touch-tone receivers 4-6

Touch-tone signaling for Direct

Inward Dialing (DID) trunks 3-13

Transfer, feature 4-13, 4-18, 4-37,

4-46

Transfer redirect, VMI ports 4-7

Trouble alarm 2-58

Trunk access

Behind Switch mode 1-26

Hybrid/PBX mode 1-20

Key mode 1-15

Trunk pools 1-14 - 1-20

Trunks

Definition 1-12

DID 1-14, 2-5, 3-11 - 3-15, 5-14

DID, power failure busy-out 2-58

DS1 2-5, 5-14 - 5-16

FX 1-14, 1-18, 1-24

Ground-start 1-14, 1-18, 2-5, 3-2,

5-13 - 5-14

Loop-start 1-14, 1-18, 1-24, 2-5,

3-1 - 3-2, 5-1 3 - 5-1 4

Tie 1-14, 1-18, 1-25, 2-5,

3-2 - 3-10, 5-14

Tie, digital emulated 3-3, 3-17

Tie, networking 3-10

Types by mode 1-12 - 1-13

TTRs, Required by voice messaging

systems 4-6 - 4-7

Two-way direction for tie trunks 3-4

Type 1 Compatible signaling type for

tie trunks 3-3, 3-8

Type 1 Standard signaling type for tie

trunks 3-3, 3-8

Type 5 signaling type for tie

trunks 3-3, 3-8

U

Unit loads 1-42 - 1-43

Universal Paging Access Module

(UPAM) 2-33

Unsupervised transfer with MERLIN

MAIL 4-9

Unsupervised transfer with MERLIN

Attendant 4-16

UPAM (see Universal Paging Access

Module)

Index IN-13

Index

V

Ventilation requirements for control

unit 1-37 - 1-39

Video codec 4-51

Video Conferencing 4-50, 5-25 -

5-26

VMI (see Voice Messaging

Interface)

Voice announce 5-17

Voice announcement 4-57

Voice Mail, AUDIX Voice

Power-lS Ill 4-40

Voice Mail with MERLIN MAIL Voice

Messaging System 4-10 - 4-11

Voice Messaging Interface ports

far-end disconnect 4-8

Generic 4-6

in/out of service 4-8

Integrated 4-6

transfer redirect 4-7

Voice messaging systems

Announcement Service with


System 4-11

AUDIX Voice Power 4-40 - 4-41,

4-33 - 4-34

Automated Attendant with MERLIN


System 4-9

Broadcast Lists with MERLIN MAIL


Call Answer with MERLIN MAIL


Fax Transfer with MERLIN MAIL


Integrated Voice Power Automated

Attendant 4-33



System 4-9 - 4-15

Mode codes 4-6

Outcalling with MERLIN MAIL Voice


Overview 4-1 - 4-8

Pager Notification with MERLIN


System 4-11

TTRs required 4-6 - 4-7

Voice messaging interface

4-6 - 4-8

W

WATS 1-14, 1-18, 1-24

Wink signaling type for tie trunks 3-5

Wink-start type for Direct Inward


Z

Zonemate equipment 2-33, 2-45

IN-14 Index