AT&T StarLAN 10 Instruction manual

555-520-200
Issue 1, June 1987
AT&T System 25
Reference Manual
© 1987 AT&T
All Rights Reserved
Printed in USA
TO ORDER COPIES OF THIS DOCUMENT REFER TO DOCUMENT NUMBER 555-520-200.
Contact: Your AT&T sales representative or
Call: 800-432-6600, Monday to Friday between 7:30 am
and 6:00 EST, or
Write: AT&T Customer Information Center
2855 North Franklin Road
P.O. Box 19901
Indianapolis, Indiana 46219
Every effort was made to ensure that the information in this document was complete and accurate at
the time of printing. However, information is subject to change. This document will be reissued
periodically to incorporate changes.
Reference Manual
Prepared by System 25
Document Development Group and the
AT&T Documentation
Management Organization
FCC NOTIFICATION AND REPAIR INFORMATION
AT&T SYSTEM 25
This telephone equipment is registered with the Federal Communications Commission (FCC)
in accordance with Part 68 of its Rules. In compliance with the Rules, be advised of the
following:
MEANS OF CONNECTION
Connection of this telephone equipment to the nationwide telecommunications network shall
be through a standard network interface USOC RJ21X jack. Connection to private line
network channels requires USOC RJ2GX jack for tie lines or USOC RJ21X jack for offpremises station lines. These can be ordered from your telephone company.
NOTIFICATION TO THE TELEPHONE COMPANY
If the system is to be connected to off-premises stations (OPSs), you must notify the
telephone company of the OPS class of service, OL13C, and the service order code, 9.OF.
Upon the request of the telephone company, inform them of the following:
— The Public Switched Network “lines” and the Private “lines” to which you will
connect the telephone equipment.
— The telephone equipment’s “registration number” and “ringer equivalence number”
(REN) from the label on the equipment.
— For private line connections, provide the facility interface code, TL31M for tie lines.
You must also specify the service order code, 9. OF.
— The quantities and USOC numbers of the jacks required.
— For each jack, provide the sequence in which lines are to be connected; the type lines
and the facility interface code and the ringer equivalence number by position, when
applicable.
This telephone equipment should not be used on coin telephone lines. Connection to party
line service is subject to state tariffs.
REPAIR INSTRUCTIONS
If you experience trouble with this telephone equipment, contact the AT&T Business
Customer Service Center on 1-800-242-2121. The telephone company may ask that you
disconnect this equipment from the network until the problem has been corrected or until
you are sure that this equipment is not malfunctioning.
RIGHTS OF THE TELEPHONE COMPANY
If your telephone equipment causes harm to the telephone network, the telephone company
may discontinue your service temporarily. If possible, they will notify you in advance. But
if advance notice isn’t practical, you will be notified as soon as possible. You will be
informed of your right to file a complaint with the FCC.
Your telephone company may make changes in its facilities, equipment, operations, or
procedures that could affect the proper functioning of your equipment. If they do, you will
be notified in advance to give you an opportunity to maintain uninterrupted telephone
service.
HEARING AID COMPATIBILITY
The voice terminals described in this manual are compatible with inductively coupled
hearing aids as prescribed by the FCC.
FCC REGISTRATION INFORMATION
Registration Number
AS593M-71565-MF-E
Ringer Equivalence
0.5A
Network Interface
RJ21X or RJ2GX
PRIVATE LINE SERVICE
Service Order Code
9.OF
Facility Interface Code
●
●
Tie Lines
Off-Premises
TL31M
Stations
OL13C
FCC WARNING STATEMENT
Federal Communications Commission (FCC) Rules require that you be notified of the
following:
●
●
●
This equipment generates, uses, and can radiate radio frequency energy and, if
not installed and used in accordance with the instruction manual, may cause
interference to radio communications.
It has been tested and found to comply with the limits for a Class A computing
device pursuant to Subpart J of Part 15 of FCC Rules, which are designed to
provide reasonable protection against such interference when operated in a
commercial environment.
Operation of this equipment in a residential area is likely to cause interference in
which case the user at his or her own expense will be required to take whatever
measures may be required to correct the interference.
DANGER
The AT&T System 25 cabinets are not user serviceable.
Some voltages inside the cabinets are hazardous. This
equipment is to be serviced only by qualified technicians.
CONTENTS
SECTION l—OVERVIEW
SECTION
2—FEATURES AND SERVICES
SECTION 3—FUNCTIONAL DESCRIPTION
SECTION 4—HARDWARE DESCRIPTION
SECTION 5—TECHNICAL SPECIFICATIONS
SECTION 6—ENVIRONMENTAL REQUIREMENTS
SECTION 7—PARTS INFORMATION
SECTION 8—REFERENCE DOCUMENTATION
SECTION 9—GLOSSARY
SECTION 10—INDEX
SECTION 1—OVERVIEW
This reference manual provides general technical information on AT&T System 25 (System
25). It includes a description of the system, its hardware and software, features and
services, environmental requirements, and technical specifications. This manual is intended
to serve as an overall technical reference for System 25.
This manual replaces AT&T System 25 Reference Manual (555-50-200, Issue 1), which
covered Release 1 Version 1 (RIV1 ) of System 25. This new issue contains the original
coverage plus complete information on R1V2, a more powerful and versatile configuration of
the system. R1V2 provides new features and services that enhance system operation,
particularly in the area of networking. Here are some of the improvements:
●
●
●
●
●
The AT&T STARLAN NETWORK (STARLAN NETWORK) Access feature provides
connectivity between System 25 and an associated STARLAN NETWORK.
Switched Loop Attendant Console operation makes the handling of incoming calls
from the network more efficient than in RIV1.
With the Tandem Trunking feature, tie trunks can be used to call through System 2 5
to reach another switching system (CO or PBX).
●
●
The Virtual Facilities addition enhances the customer’s outgoing network
capabilities.
New voice terminals are provided for meeting specific system user needs.
A new voice feature, Last Number Dialed, is added, and existing voice features are
improved.
Data services are enhanced with new options for placing and controlling calls.
Both RIV1 and R1V2 of System 25 are described in this manual. Unless specifically marked
as “VI” or “V2”, all of the information pertains to both versions. VI information applies
only to RIV1 systems; V2 information applies only to R1V2 systems.
Organization
The manual is divided into 10 Sections. The remaining Sections are as follows:
●
SECTION
2–FEATURES
AND
SERVICES
●
SECTION 3–FUNCTIONAL DESCRIPTION
●
SECTION 4–HARDWARE DESCRIPTION
●
SECTION 5–TECHNICAL SPECIFICATIONS
●
SECTION 6–ENVIRONMENTAL REQUIREMENTS
●
SECTION 7–PARTS INFORMATION
●
SECTION 8–REFERENCE DOCUMENTATION
●
SECTION 9–GLOSSARY
●
SECTION 10–INDEX
1-1
System 25 Description
System 25 (Figure l-l) is an advanced digital switching system that integrates voice and
data communications. It not only provides the features of a state-of-the-art PBX, but goes a
step further by allowing data to be switched point-to-point without first being converted to
analog format. This capability can be used to set up connections between data terminals,
word processors, personal computers, and host computers.
System 25 uses intelligent port circuits equipped with distributed network processor
elements to provide (essentially) nonblocking voice and data switching.
Voice communications features combine traditional telephone features, such as Call Transfer
and Hold, with advanced features, such as Individual and Group Call Coverage, Hands-FreeAnswer On Intercom, and Speed Dialing (see Section 2, “Features and Services”).
Data communications features provide switched data connections supporting transmission of
voice and data over Premises Distribution System wiring. Data connections can be made
between two digital data modules (asynchronous data units), two analog modems, or between
an analog modem and a digital data module. Release 1 Version 2 (R1V2) also provides access
to STARLAN NETWORKs.
The system has data modules that provide an RS-232 interface for full duplex, asynchronous,
transmission of data up to 19,200 bps, and an integrated 212A-compatible modem pool
resource.
System 25 supports the following:
●
Trunk and Network Facilities–Dual Tone Multifrequency (DTMF) and Dial Pulse
Signaling on incoming and outgoing trunks (dial pulse only on DID trunks).
— Loop Start (LS)
— Ground Start (GS) (Strongly Preferred over Loop Start in most installations)
— Tie Trunks (Type I and Type I Compatible E&M, Type V Simplex)
— Direct Inward Dialing (DID)
●
●
Voice Terminals
MERLIN®
Data
–
Single-Line
Touch-Tone,
Single-Line
Rotary
(V2),
MET,
and
Facilities
— Digital Data End Points – RS-232 Interfaces via Asynchronous Data Units
— Analog Data End Points — Tip/Ring-Type Modem Interfaces
— STARLAN NETWORK Access (V2 only).
●
Networking Capability
— Tie Trunks
— Tandem Trunking
— Endpoint in Electronic Tandem Network (Tributary only, not Satellite)
— Endpoint of Enhanced Private Switched Communications Services (EPSCS)
— Endpoint of Tandem Tie Trunk Network (TTTN)
— Endpoint of Common Control Switching Arrangement (CCSA).
1-2
DIGITAL SWITCH
RS-232C
SYSTEM ADMINISTRATION
TERMINAL
OR
ADVANCED ADMINISTRATION
PC TERMINAL
RS-232C
EMERGENCY
-48V DC
TRANSFER
UNIT
TAPE
BACKUP
UNIT
COMMON
CONTROL
COMPLEX
SMDR
OUTPUT
DEVICE
RS-232C
SINGLE-LINE
VOICE
TERMINALS
ANALOG
MULTILINE
VOICE
TERMINALS
HYBRID
DIRECT TRUNK
ATTENDANT
CONSOLE OR
SWITCHED LOOP
ATTENDANT
CONSOLE
ATTENDANT
DIRECT
EXTENSION
SELECTOR
CONSOLE
STARLAN
NETWORK
WORKSTATIONS
STARLAN
NETWORK
PRINT & FILE
SERVICES
STARLAN
NETWORK
HOSTS
TRUNK FACILITIES
DID
l FX, WATS (LOOP/GROUND START)
l T I E
l CO (LOOP/GROUND START)
l PAGING SYSTEMS
ANALOG FACILITIES
SWITCHING
NETWORK
(PORTS)
HYBRID
DATA
ANALOG
l
ASYNCHRONOUS
D A T A
UNIT (ADU)
MODEM
RS-232C
HOST COMPUTERS
TERMINALS
RS-232C
HYBRID
STARLAN
NETWORK
GATEWAY
ANALOG
.
MUSIC SOURCE
. EXTERNAL ALERTS
DELAY ANNOUNCEMENT
DICTATION EQUIPMENT
AUXILIARY TRUNKS
Figure 1-1. System 25 Block Diagram
1-3
.
.
. PAGING SYSTEMS
. DICTATION EQUIPMENT
Call Handling Capabilities
System 25 can be arranged as a stand-alone system or can be part of a private network. The
system provides 256 ports to support the following:
●
115 simultaneous two-party conversations
●
Traffic Handling Capacity of 4140 CCS/Hour (Trunking Limited)
●
Busy Hour Call Capacity of 2500 calls (DTMF Register Limited)
●
Up to 104 trunk ports including Central Office (CO), DID, Tie, Foreign Exchange
(FX), Wide Area Telecommunications Service (WATS), and 800 Service
An Auxiliary Trunk interface for paging (Vl and V2) and dictation systems (V2).
●
●
Up to 240 ports that support a combination of the following:
— Up to 200 ports for voice terminals and auxiliary feature port equipment.
— Up to 104 data ports providing RS-232C connections to data terminals,
personal or multiport computers.
Refer to Hardware and Software Parameters as provided in “Technical Specifications”
(Section 5) for detailed specifications.
Safety
System 25 meets all requirements found in Underwriters Laboratories Standard for
Telephone Equipment (1459).
Business Communications Needs
The remainder of this Section describes how System 25’s R1V2 features may be used to
satisfy a customer’s communications needs. This material may be thought of as the reverse
of the “Features and Services” section which follows.
The business communications capabilities of the majority of small businesses with more
than thirty phones are provided by a Private Branch Exchange (PBX). System 25 is a PBX
designed to meet the business communications needs of customers in the 30 to 150 station
range.
The communications needs of most business customers may be broken down into five basic
categories. Customer experience has shown that a PBX needs to provide–
●
●
●
Prompt handling of incoming calls to maximize revenue opportunities and client
satisfaction,
Ease of access to and cost control of outgoing calls over public network and private
facilities,
Easy movement of calls between on-premises phones and between on-premises and
off-premises phones,
●
Sharing of data between PCs and/or host computers and data terminals, and
●
Growth
and
rearrangement
of
facilities.
The following pages outline System 25’s outstanding ability to provide these services.
1-4
Incoming Business Communications
Successful call termination is the key to capturing all incoming communications associated
with revenue issues, client inquiries, decision data, etc. Call termination involves identifying
the called party and routing the call to a primary or secondary answering position. System
25 provides powerful tools for both call screening and call termination.
●
●
●
●
Attendant Consoles allow one or two attendants to answer, screen, and steer
incoming calls using either Direct Trunk or Switched Loop operation. With
attendant operation, incoming calls can be screened and forwarded to the appropriate
party for resolution, messages taken for absent clients, or forwarded to alternate
locations. Calls may arrive over any of the network facilities described in later
sections of these notes.
Direct Inward Dialing allows incoming callers to reach specific individuals or
facilities without attendant assistance. This allows specific numbers to be advertised
for direct customer access to brokers, emergency services, etc. over a shared pool of
DID trunks.
Direct Group Calling (DGC) allows incoming calls to be directed to a specific group
of stations. Calls to a DGC group hunt for an idle station in a circular manner,
starting at the station following the last one called. If all group members are busy,
calls are queued and can be sent to a delay announcement. A DGC group can
terminate calls to sales, services, computer, announcement, etc. over either ordinary
CO trunks or DID trunks.
Personal Lines provide dedicated outside lines for multiline voice terminal users and
are accessed via a dedicated button for both incoming and outgoing service. Up to
sixteen terminals may share a Personal Line with up to four parties simultaneously
off-hook. A personal line provides direct access to brokers, emergency service, etc.
over a dedicated loop start or ground start trunk.
Frequently, the called party is not available to handle an incoming call. System 25 provides a
number of methods for redirecting incoming calls to alternate resources.
●
●
●
l
Call Following allows users who are away from their phone to receive calls at
another phone. Users may login their Personal Dial Code (PDC) at any other System
25 voice terminal and receive their calls at that terminal. This feature supports
roving personnel and shared office space for company staff.
Call Coverage allows calls that are not answered within a specified number of rings
to be redirected to an individual covering station and/or a group of covering stations.
This is especially useful for Boss-Secretary arrangements, staff backup, and message
service. This feature is versatile enough to permit suitable alternate answering
arrangements for virtually every level of employee. Special features, such as the
Send All Calls feature, which routes a user’s calls directly to covering station(s),
accommodate the day-to-day variations that occur in an employee’s work schedule.
Station Hunting provides automatic redirection of incoming calls to an idle member
of a hunt group when the called party is busy.
Call Pickup allows a user to answer a call ringing at another terminal. Directed Call
Pickup allows a user to answer a call ringing at any terminal by dialing the call
pickup code and the PDC of the ringing station. Group Call Pickup permits calls to
any other terminal in the call pickup group to be answered by dialing the group call
pickup code. With Call Pickup, users do not have to leave their phone to answer
other’s calls. This feature is especially useful for local coverage in group offices not
1-5
supported by secretarial service and equipped with economical single-line phones.
When alternate resources are not available to handle an incoming call, System 25 provides
for attendant handling of the call utilizing camp-on, redirection and/or message service.
●
●
●
Camp-On allows the attendant to extend an outside call to a busy station. A burst of
tone is heard at the called station to notify the user of the camped-on call. The
caller is placed on hold and hears music-on-hold, if available. When the user hangs
up, the camped-on call begins ringing immediately. Only one call may be camped on
at a time. The Return Coverage on Busy feature returns unanswered camped-on calls
to the attendant for service after a specified interval.
Return Coverage on Don’t Answer returns unanswered attendant-extended calls for
additional service (redirection/messaging).
Messaging Service supports activation of an LED at the called station to indicate
that the attendant, message desk, or another station has a message for the user.
Special arrangements are needed to handle incoming calls during periods when the normal
staff is not available, for example at night and on weekends. System 25’s Night Service
feature allows on-duty personnel to answer incoming attendant-seeking calls when the
attendant is not on duty. Directed Night Service redirects incoming attendant-seeking calls
to designated voice terminals, such as a guard desk or coverage position. Trunk Answer
From Any Station allows users to answer incoming calls from any station by dialing the
Night Service access code. Night personnel can be alerted by a Night Bell.
Outgoing Business Communications
One of the key functions of a customer premises communications system is to provide easy
access to the most cost effective network facilities for outgoing calls. The system needs to be
capable of steering calls based on cost, and must also keep records of incoming and outgoing
calls and associated costs. Building on its ground start trunk capability, System 25 features
control costs and record usage as follows.
●
●
●
●
●
Call Restrictions allow the manager to restrict users from making certain types of
calls. Restriction is administered through outward restriction, toll call restriction,
and facility access restriction.
Automatic Route Selection provides manager defined routing of calls over the
telecommunications network based on preferred routes (normally the least expensive
route available at the time the call is placed) with capacity for multiple common
carriers and routing through tandem switch points. The user dials a standard DDD
number and the system selects the call route.
Station Message Detail Recording (SMDR) generates detailed call information on all
incoming and outgoing calls and sends this information to an output device (PC or
printer).
Call Accounting Systems provide multiple types of customer reports on
communication costs and usage.
Account Code Entry allows a user to associate calls with an account code for chargeback purposes.
1-6
Ease of access to multiple types of network facilities (provided for minimum cost) is
managed by:
●
●
●
●
●
●
●
●
Automatic Route Selection (ARS) allows the customer to dial a standard DDD
number. ARS selects the preferred route and does any number conversions required
for the facilities selected.
System 25’s Virtual Facility feature provides convenient and inexpensive access to
OCCs. This feature provides access to OCC facilities over a user specified physical
facility; dedicated OCC trunks are not needed. Local OCC access numbers and
account codes are automatically added by System 25. System 25’s Virtual Facility
feature is fully integrated with its ARS, Toll Restriction, and SMDR/CAS features.
Last Number Dialed automatically saves the last number dialed and allows the user
to retry the number without redialing. (Multiline voice terminals only)
Repertory Dialing allows multiline voice terminal users to store a telephone number
or account and associate that number with a button on their voice terminal.
Pressing a Repertory Dial button is equivalent to dialing the stored number (onetouch dialing).
System Speed Dialing allows all users to dial 90 selected numbers using three-digit
codes. Users can also program up to seven Personal Speed Dial Numbers which are
accessible only from their terminals. System Speed Dialing can be used by the system
administrator to hide business account codes from users.
Pooled Facility-Dial Access allows both
to access a common pool of trunks for
or, on multiline voice terminals, by
resource pooling which results in better
multiline and single-line voice terminal users
outgoing calls by dialing a facility access code,
pressing a button . This grouping provides
service with a given number of trunks.
Personal Lines provide dedicated outside lines for multiline voice terminal users.
Personal lines are accessed via a dedicated feature button. Up to sixteen terminals
may share a personal line.
Third-Party Call Setup allows PCs to set up calls between a System 25 voice/data
terminal and any other facility. A PC application program could use this capability
to retrieve information from a database.
Last Number Dialed, Repertory Dialing and Speed Dialing are also applicable to dialing and
managing internal calls. Personal lines provide both incoming and outgoing service.
Internal Call Movement
Typically, about 40 percent of PBX calls are internal calls, call transfers to another location,
conference of multiple locations, temporarily suspended calls, page to locate the called party,
etc. Rapid placement of internal calls and easy call movement from the answering station to
a new station are supported in System 25 with numerous features.
To provide easy internal call setup, System 25 provides the following features.
●
Direct Station Selection (DSS) allows one-button access from a multiline voice
terminal to another voice terminal, a pooled facility, paging zone or DGC group. The
DSS status LED reflects the idle/busy status of the associated termination point.
This feature is used to track and contact frequently called associates.
1-7
●
●
Automatic Intercom allows multiline voice terminal users to call each other by use of
a dedicated line appearance. A private dedicated path ensures that a path is always
available. This feature is frequently used in Boss/Secretary arrangements.
The Dial Plan for System 25 is based on the concept that, whenever possible, calls
should be placed to individuals rather than to pieces of equipment. To implement this
concept, individuals are assigned Personal Dial Codes (PDCs) and are allowed to
login those PDCs at other terminals. The system automatically routes the call to the
home terminal or logged-into terminal. This significantly increases the probability
of reaching the called party. In addition, the dial plan is built on a flexible
numbering scheme which allows the number of dialed digits to match assigned PDCs
(2/3/4 digit dial plans) and to be administered to match telephone company assigned
Direct Inward Dialing numbers.
Efficient internal call termination is supported with the following features.
●
●
Distinctive Ringing provides two types of ringing to allow users to distinguish
between outside calls and inside calls,
Hands-Free Answer on Intercom (HFAI) allows Speakerphone and
to auto-answer inside or attendant extended calls. With HFAI
generates a tone burst over its speaker to alert the calling and called
completion. Both parties may then converse; no action by the
required.
HFAI terminals
active, the set
party of the call
called party is
Frequently, the first termination point for a call is not its final destination. To support
internal call movement, System 25 provides the following features.
●
●
●
●
Transfer allows a user to transfer any call to another voice terminal. This feature
supports transfer of calls from the answering position to another location for
completion of a transaction. Examples are secretary to boss, office to lab, broker to
specialist, etc.
Conference allows up to five parties (maximum two outside), including the originator,
to participate in a call. This feature supports add-on of additional parties to a call
for joint consultation, crisis management, schedule coordination, etc.
Hold allows a user to suspend a call. The Hold feature allows users to temporarily
disconnect from one conversation and either place or answer another call. Music or
information bulletins may be provided to the held party. Called parties frequently
use the hold period to access computer data bases, search categories and/or consult
with others via a second phone call.
Call Park allows a user to place a call or conference on hold and then pick up the call
from any voice terminal. The user can page another party to pick up the parked call
or may move to another location and then re-access the call.
Data Communications
Small Business customers have started to integrate PCs into their day-to-day business
operations. Businesses have found a need to access the data bases (sales, inventory,
personnel) in these PCs from more than one location (both on- and off-premises). System 2 5
data features are specially engineered to enhance a user’s ability to access data from
multiple locations. System 25 has been designed to help these businesses use their personal
computers, data terminals and host computers more effectively by providing the following
features.
1-8
●
●
●
Circuit switched data communications up to 19,200 bps (RS232 interface). This
provides circuit switched connections from asynchronous data terminals, PCs, or host
computers to host computers or network facilities. Users can be located and/or
moved to any on-premises office equipped with the standard AT&T four-pair wiring
plan. Thus an asynchronous terminal or PC can have access to multiple host
computers, remote data bases via a modem pool, and a local area network
(STARLAN) via System 25’s STARLAN NETWORK gateway.
Packet switched data connections at 1 million bps over AT&T’s STARLAN
NETWORK local area network. This provides data transfer between client PCs and
servers (PCs/host computers/printers, etc.) on the local area network (LAN). LAN
users can be located and/or moved to any on-premises office equipped with standard
AT&T four-pair wiring. The LAN allows PCs to share facilities (printers, disk
systems, modem pools, etc.)
System 25’s STARLAN NETWORK ACCESS software and STARLAN NETWORK
gateway provide access to the STARLAN NETWORK for off-premises and occasional
on-premises users. These users do not need to install a Network Access Unit (NAU)
in their PCs to use the STARLAN NETWORK ACCESS software. The data transfer
rate is limited to 9600 bps or, for off-premises users, by the modem.
LAN users can access hosts connected to System 25, or, if their System 25 is equipped
with a modem pool, remote hosts. Finally, terminals and PCs connected to System 25
data ports can access host computers on the LAN.
Frequently a user needs to access a LAN data base at or from a remote location (home,
motel, client office, branch, etc.). To support out-of-building access to computer data over
network facilities or OPS lines, System 25 provides the following features.
●
●
●
Modem pooling allows data terminals to communicate over analog facilities utilizing
the standard dialing plan and provides full access to all network facilities, cost
incoming
management
mechanisms,
ARS,
and
call
control
tools
(DID/attendant/DGC, etc.).
Transfer to data allows a data call to be set up on a voice terminal and then be
transferred to a data terminal or computer. This feature can also be used to enter an
account code for the data call.
The System 25 STARLAN NETWORK gateway allows the LAN environment to be
extended to occasional users or remote locations. Off-premises users can access the
LAN utilizing all the network features, cost control mechanisms, and incoming call
management facilities of System 25. The data transfer rate is governed by the
modem.
Setting up data communications with PCs, host computers, and/or remote access can be a
source of confusion for occasional users. Special data features are provided by System 25 to
assist the user in utilizing its rich set of data communications capabilities.
●
●
The integrated voice-data dialing plan recognizes the different types of data
endpoints (digital/ analog and remote/local) in a connection and automatically
inserts the required data communication equipment. In addition, autobauding
supports the alignment of equipment with the capacity to transmit at different data
rates.
Station Hunting supports the use of a single dial code to access a group of host
computer ports.
1-9
●
●
●
●
●
Terminal Dialing provides the user with fast access to data communications via
keyboard dialing at a terminal or PC.
Command Mode provides a menu of data services supporting terminal dialing, and
display and control of user data port options. A user friendly Change Options menu
is provided for user administration of data options.
Expert Mode is an enhancement that provides an alternative method of accessing
Command Mode functions. Expert Mode eliminates the display of menus and allows
multiple commands to be entered on a single line. Expert mode lends itself well to
computer-driven scripts for call setup.
Communication Access Manager (CAM) is an MS-DOS* software application that
provides a phone manager for placing voice and data calls for the user and VT100†
terminal emulation. CAM may be used on either STARLAN NETWORK client
workstations or on PCs connected to System 25. CAM has a 200-entry directory with
one-touch dialing for both voice and data calls and auto-login capability for data calls
to host computers. CAM’s Remote Access feature provides password protected
unattended access to PC files and electronic mail. File transfer is supported with the
popular XMODEM protocol.
STARLAN NETWORK ACCESS is an MS-DOS application that allows PCs not
connected to the STARLAN NETWORK to call through the System 25 STARLAN
NETWORK Interface and run STARLAN NETWORK client software to access file
and printer servers on the STARLAN NETWORK. ACCESS uses a PC’s serial
communications port to communicate with the STARLAN NETWORK Interface.
ACCESS is compatible with NETBIOS, permitting execution of most applications
written for the IBM‡ PC Network and IBM Token Ring Network.
Growth & Rearrangement
Historical data indicates that clients in the System 25 station range have a need for
communications systems capable of significant growth and rearrangement. Clients need
flexibility over the life of the system to easily add capacity, move stations, modify cost
control options, etc. The architecture of System 25 was implemented with the objective of
meeting this need.
●
●
●
Advanced Administration (optional) is an easy-to-use, menu driven personal computer
software package for configuring the rich set of system options.
Uniform Wiring Plan (four-pair) allows a building to be prewired for the rich set of
AT&T Small Business PBX service offerings. This modular wiring plan supports client
reconfiguration of an office with variations in station type (Analog, MET, MERLIN,
futures) and data configurations (LAN, asynchronous, synchronous). It supports
simultaneous voice and data from standard four-pair modular jacks.
System 25/75/85V2 Standard Architecture supports efficient growth with modular
cabinets, universal carrier slots, non-blocking network and uniform wiring plan. (See
* Registered trademark of Microsoft Corp.
† Trademark of Digital Equipment Corp.
‡ Trademark of International Business Machines Corp.
1-10
Figures 1 and 2.) Every circuit slot in the system can be used for trunk cards or
voice/data station cards. All these attributes allow the client to add future capability
without breakage and re-engineering of existing equipment. Thus, the client is able to
minimize initial investment while not restricting future growth.
Over time, the type of tools and facilities that a business utilizes changes. It is important
that a PBX provide support for the full set of Telco network options over its installed life,
even when only a subset is initially used. Trunks link two switching systems, such as System
25 and the local Central Office or System 25 and another PBX. System 25 supports five
different telephone company trunk interfaces to provide desired connectivity at minimum
expense. Thus the opportunity exists to select the best trunk types, depending on tariffs and
customer needs.
●
●
●
●
●
Loop Start (LS) trunks for public network access at minimum tariff. These trunks
handle outgoing and incoming attendant calls, incoming DGC calls, outgoing pooled
facility calls, and personal line calls.
Ground Start (GS) trunks for public network access. These trunks handle the same
type of calls as LS trunks. They provide protection against call reorigination without
toll restriction, more reliable automatic route selection, virtual facilities, SMDR and
CAS. Simultaneous incoming and outgoing call seizure of the same trunk under
heavy traffic conditions is essentially eliminated with ground start trunks. GS
trunks should usually be selected in preference to LS trunks unless tariff
considerations are overriding. Note, however, that Centrex Service requires LS
trunks.
Direct Incoming Dial (DID) trunks for dialing a station directly from outside
(attendant assistance not required). Outside dial access to stations, trunks (optional),
and answering groups (Direct Group calling) is provided.
Tie Trunks for linking PBXs with dedicated private circuits for high volume calling.
Dial access to stations, other trunks, answering groups (Direct Group Calling) and an
Electronic Tandem Network endpoint capability are provided.
Off-Premises Stations (OPS) allow a single-line voice terminal to be located remotely
and connected to System 25 via arrangements with the local Telco. This service is
used to provide users at secondary sites (or their residences) many of the same
features as an on-premises single-line station.
To enhance the usage and control of the above set of network facilities, System 25 provides
the rich set of access features outlined in the Outgoing Business Communications Section. In
addition, System 25 can support networking between systems by:
. Serving as an endpoint on an electronic tandem network (ETN) using its tie trunks
and flexible dialing plan.
. Serving as an off-network or on-network access point with its dial access/transfer
between tie-trunks and Telco trunks (LS/GS/DID). This allows usage of tie trunks
to reach a distant System 25 and then connect through that System 25 to local Telco
facilities to complete the call.
To support efficient utilization of trunks, they can be grouped together (up to 16 groups) if
all trunks in the group perform the same function. This resource pooling provides better
service with a given number of trunks, and simplifies administration and calling.
1-11
Types of trunks which can be assigned in System 25 are:
●
●
●
●
●
Central Office, which provide a link with the local telco for incoming and outgoing
calls (LS/GS)
Foreign Exchange (FX), which connect to a CO other than the local CO for high
volume calling from a distant location
Wide-Area Telecommunications Service (WATS), which connect to an Outward
WATS office or a dial “800” Service Office
Direct Inward Dial (DID), which provide incoming service from a CO to directly
access a station or facility (STARLAN NETWORK interface, trunk group)
Tie trunks, which provide a link with another private switching system.
To support efficient utilization of this rich set of network options, System 25 provides the
functions outlined in the Outgoing and Incoming Business Communications sections.
Conclusions
System 25 has been targeted at providing excellent small business communications capability
at the right price. The thousands of systems in service in the first year of production have
confirmed that these capabilities are an excellent match with small business customers’
communications needs.
1-12
USER
CHANGEABLE
VIRTUAL
OPTIONS
FACILITIES
(V2)
(V2)
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2-253
2-258
LIST OF FIGURES
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2-13
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2-16
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2-17
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2-20
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2-24
Figure 2-6.
Model 23A1 Attendant Direct Extension Selector Console . . . . .
2-26
Figure 2-7.
Attendant Direct Extension Selector Console Connections . . . . .
2-28
Figure 2-8.
Automatic
Route
Selection
Flow
Figure 2-9.
Automatic
Route
Selection
Routing
Figure 2-1.
Direct
Trunk
Attendant
Console
.
Figure 2-2.
Direct
Trunk
Attendant
Console
Connections
Figure 2-3.
Switched
Loop
Figure
2-4.
Console
Buttons
Figure
2-5.
Switched
Loop
Attendant
and
Console
Display
Attendant
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Console
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Connections
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2-54
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2-59
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2-69
Figure 2-11. SMDR Call Record Format . . . . . . . . . . . . . . . .
2-70
Figure 2-12. SMDR Call Record Header Format . . . . . . . . . . . . .
2-71
Figure 2-13. SMDR Output Equipment or Call Accounting System—On-Premises
Direct Connections (Sharing Same AC Outlet) . . . . . . . . .
2-72
Figure 2-14. SMDR Output Equipment or Call Accounting System—On-Premises
Direct Connections (Greater Than 50 Feet From System
Cabinet) . . . . . . . . . . . . . . . . . . . . .
.
2-73
Figure 2-15. SMDR Output Equipment–On-Premises Switched
Connections . . . . . . . . . . . . . .
.
2-74
Figure 2-16. SMDR Output Equipment—Off-Premises Direct Connections . . . .
2-75
Figure 2-17. SMDR Output Equipment–Off-Premises Switched
Connections . . . . . . . . . . . . . .
.
2-76
Figure 2-18. Communications Access Manager Architecture . . . . . . . . .
2-99
Figure
2-10.
Typical
SMDR
Call
Detail
Chart
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Pattern
Report
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2-107
Figure 2-20. Dictation System Connections (FCC Registered) . . . . . . . .
2-119
Figure
.
2-120
Figure 2-22. Digital Tape Unit–On-Premises Direct Connections (Sharing Same AC
Outlet) . . . . . . . . . . . . . . . . . . . . . .
2-121
Figure 2-23. Delay Announcement Equipment Connections (FCC
Registered) . . . . . . . . . . . . . . .
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2-126
Figure
2-24.
External
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2-140
Figure
2-25.
Supplemental
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2-141
Figure
2-19.
2-21.
Asynchronous
Digital
Data
Tape
Unit
Alert
Unit
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Interface
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Connections
Alert
Adapter
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Signals
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Connections
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2-146
Figure 2-27. Typical Headset Adapter Connections For 7300H Series Multilane Voice
Terminals (Except 34-Button Deluxe, BIS, or BIS with
Display) . . . . . . . . . . . . . . . . . . . . . .
2-147
Figure 2-28. Typical Headset Adapter Connections For 34-Button Deluxe, BIS, or
BIS with Display Voice Terminals . . . . . . . . . . . .
.
2-148
Figure 2-29. Typical Headset Adapter Connections For 12-Button MET
Sets . . . . . . . . . . . . . . . . . . .
.
2-149
Figure 2-30. Music-On-Hold Equipment Connections (FCC Registered) . . . . .
2-168
Figure 2-31. Music-On-Hold Equipment Connections (Non-Registered) . . . . .
2-169
Figure 2-32. Delay Announcement Equipment Connections (FCC
Registered) . . . . . . . . . . . . . . .
Figure 2-26. 500A/502A
Headset
Adapter
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2-174
Figure 2-33. Paging Equipment Connections Using CO Trunk Ports (FCC
Registered) . . . . . . . . . . . . . . . . .
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2-179
Figure 2-34. Paging Equipment Connection to TN763 Causing 278A
Adapter . . . . . . . . . . . . . . . . .
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2-180
Figure 2-35. 10B
Emergency
Figure 2-36. Emergency
Transfer
Transfer
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(ETU)
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2-191
Connections
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2-192
Unit
Unit
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2-193
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2-208
Figure 2-39. Speakerphone Connections For 7300H Series Multilane Voice Terminals
(Except 34-Button Deluxe) . . . . . . . . . . . . . . . .
2-209
Figure 2-40. Speakerphone Connections For 34-Button Deluxe Multiline Voice
Terminals . . . . . . . . . . . . . . . . . . .
.
2-210
Figure 2-41. Speakerphone Connections For 12-Button MET Sets . . . . . . .
2-211
Figure 2-42. STARLAN NETWORK and System 25 Configuration . . . . . . .
2-216
Figure 2-43. STARLAN NETWORK Connection to System 25 (With 2500 SingleLine Telephone) . . . . . . . . . . . . . . . . . .
.
2-219
Figure 2-44. STARLAN NETWORK Connection to System 25 (With ATL-Type
Telephone) . . . . . . . . . . . . . . . . . . .
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2-220
Figure 2-45. Model
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2-228
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2-229
Figure 2-47. SAT On-Premises Direct Connections (Greater Than 50 Feet From
System Cabinet) . . . . . . . . . . . . . . . . . .
.
2-230
Figure 2-48. SAT
On-Premises
Switched
Figure 2-49. SAT
Off-Premises
Direct
Figure 2-50. SAT
Off-Premises
Switched
Figure 2-37. Multiple
ETU
Figure 2-38. Speakerphone
703
Arrangements
Adjuncts
System
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Administration
Terminal
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Figure 2-46. SAT On-Premises Direct Connections (Sharing Same AC
Outlet) . . . . . . . . . . . . . . . . .
Figure 2-51. Command
Mode
Menu
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2-231
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2-232
Connections
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2-233
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2-254
Connections
Connections
Tree
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v
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TABLE
2-A.
System
Features
TABLE
2-B.
Network
TABLE
2-C.
Data
TABLE
2-D.
Station
TABLE
2-E.
Attendant
.
Features
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2-2
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2-3
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2-3
Features
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2-4
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2-5
TABLE 2-F. Partial List of Permissible Data Port (TN726) Options . . . . . .
2-97
TABLE 2-G. Typical Option Profiles for Different Types of Data Port
Endpoints . . . . . . . . . . . . . . . . .
.
2-98
TABLE 2-H. Call Progress Messages for Data Terminal Dialing . . . . . . .
2-114
TABLE
2-I.
LED
Indications
.
TABLE
2-J.
User
Changeable
Options
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Features..
Features
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2-157
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2-253
SECTION 2—FEATURES AND SERVICES
This section describes the System Features, Network Features, Data Features, Station
Features, and Attendant Features of AT&T System 25. It also covers certain services that
support and implement the features; included in this category are the digital tape unit, the
dial plan, system administration, and system maintenance. A general discussion of data
topics is also provided.
The feature descriptions are arranged in alphabetical order, regardless of the feature group
to which they belong. Information for each feature is presented under five headings:
Description, Considerations, Interactions, Administration, and Hardware Requirements.
●
Description
Defines the feature, describes what it does for the user, and how it is used.
●
Considerations
Discusses the applications a n d b e n e f i t s o f t h e f e a t u r e , f o l l o w e d b y f e a t u r e
parameters and factors to be considered when the feature is used.
●
Interactions
Lists and briefly describes other features that can affect the feature being described.
Interacting features are those that:
— Depend on each other—One of the features must be provided if the other one
is.
— Cannot coexist–One of the features cannot be provided if the other one is.
— Affect each other–The operation of one feature modifies, or is modified by,
the operation of the other.
— Enhance each other—The features, in combination, provide improved service
to the user.
●
Administration
Requirements
States whether or not administration is required and lists items requiring
administration.
●
Hardware Requirements
List any additional hardware needed to use the feature.
Tabular listings of features by group (System, Network, Data, Station, or Attendant)
immediately follow this introduction. Each type, standard, or optional of the feature, is also
noted on these lists:
●
●
Standard features—Built into each system (always provided but can require
administration to make them operational)
Optional features–Such
additional hardware.
as Music-On-Hold, require both administration and
Features restricted to single-line or multiline voice terminals are noted where applicable.
The MET sets operate the same way as 5-button 7300H series voice terminals, unless
otherwise noted.
2-1
System Features
System features (Table 2-A) are those that affect the entire operation of the system. All
system features are available with both Vl and V2.
TABLE 2-A. System Features
FEATURE NAME
FEATURE
TYPE *
O
Call Accounting
Dial Plan
S
Dictation System Access
O
Digital Tape Unit
O
Direct Group Calling
Direct Group Calling Delay Announcement
O
End-to-End Signaling
Extended Stations
O
S
S
External Alerts
O
Intercept Treatment With Reorder Tone
S
Interdigit Timeouts
S
O
Music-On-Hold
Night Service (Directed and TAAS )
S/O†
Night Service Delay Announcements
Out-Of-Building
Stations
O
O
Paging System Access
O
Personal Dial Codes
Pooled Facility-Dial Access
S
S
Power Failure Transfer
O
O
Remote Administration Interface
Station Message Detail Recording
O
O
System Administration
System Maintenance
S
S
Touch-Tone and Dial Pulse Service
* Feature types: S= Standard; O= Optional (requires additional equipment).
† S/O - Standard for Directed, Optional for TAAS Night Service.
2-2
Network Features
This group of features (Table 2-B) supports communications with the public network and
with other locations in the private network of which System 25 can be a part.
TABLE 2-B. Network Features
FEATURE
TYPE *
FEATURE NAME
S
O
O
O
O
S
S
Automatic Route Selection
Direct Inward Dialing
Off-Premises Stations
Tandem Trunking (V2)
Tie Trunks
Trunk Groups
Virtual Facilities (V2)
Data Features
Data Features (Table 2-C) support the switched data services of the system. Data services
provide switched connections between analog and digital data endpoints.
TABLE 2-C. Data Features
MULTILINE TERMINAL
BUTTON LABEL †
FEATURE NAME
FEATURE
TYPE *
Command Mode
S
Communications Access Manager (V2)
Data Call Setup
O
S
Data Services Overview
Data Terminal Dialing
S
Expert Mode (V2)
Modem Pooling
S
O
AT&T STARLAN NETWORK Access (V2)
O
S
Third-Party Call Setup (V2)
[DATA]
Transfer to Data
User Changeable Options (V2)
S
S
* Feature types: S= Standard; O= Optional (requires additional equipment).
† Bracketed items are associated voice terminal feature button labels;
these labels are also used in feature descriptions where applicable.
2-3
Station Features
The many Station Features (Table 2-D) available allow individual user needs to be met. As
these needs change, assigned features can also be changed. Station Features provide many
important services that help save time and make calling more convenient.
TABLE 2-D. Station Features
FEATURE NAME
Account Code Entry
Automatic Intercom
Busy-to-Idle Reminder
Call Accountability
Call Coverage-Group
Call Coverage-Individual
Call Following (Forwarding)
Call Park
Call Pickup
(’all Progress Tones
Calling Restrictions
Conference
Conference Drop
Direct Station Selection (DSS)
Distinctive Ringing
Exclusion
Hands-Free-Answer On Intercom
Headset Adapter Adjunct
Hold
Last Number Dialed (V2)
Line Selection
Line Status And I-IJse Indications
Manual Signaling
Messaging Services
Personal Lines
Pooled Facility-Button Access
Program
Recall
Repertory Dialing
Send All Calls
Speaker (Spokesman Service)
Speakerphone Adjunct
Speed Dialing
Station Hunting
Station-To-Station Message Waiting
Test
Transfer
Trunk-To-Trunk Transfer
SINGLE-LINE MULTILINE TERMINAL FEATURE
BUTTON LABEL †
TYPE*
TERMINAL
x
x
x
x
x
x
x
x
x
x
x
x
x
x
[ACCT ENTRY]
[AUTO ICOM]
x
x
[COVER-GRP]
[COVER-IND]
x
x
x
x
x
x
[DSS or FLEX DSS]
x
[EXCLUSION]
[AIJTO ANS]
x
[LAST # DIALED]
x
[SIGNAL]
x
[PERS LINE]
[FACILITY]
x
x
[REP DIAL]
[SEND ALL CALLS]
x
x
x
x
x
x
x
x
x
[MSG WAIT]
x
x
x
x
x
s
s
s
s
s
s
s
s
s
s
s
s
s
s
s
s
o
o
s
s
s
s
s
s
s
s
s
s
s
s
s
o
s
s
s
s
s
s
* Feature types: S= Standard; O= Optional (requires additional equipment).
† Bracketed items are associated voice terminal feature button labels;
these labels are also used in feature descriptions where applicable.
2-4
Attendant Features
Attendant Features (Table 2-E) are available to the attendant using the Direct Trunk
Attendant Console (DTAC) or the Switched Loop Attendant Console (SLAC) (V2 only) and
(optionally) a Direct Extension Selector Console. In addition, most multiline voice terminal
station features are available to the attendant.
TABLE 2-E. Attendant Features
FEATURE NAME
CONSOLE BUTTON
LABEL †
FEATURE
TYPE *
Attendant Call Extending
[START]
Attendant Camp-On
Attendant Cancel
S
S
[CANCEL]
S
O
Attendant Console, Direct Trunk
Attendant Console, Switched Loop (V2)
O
Attendant Display (V2; SLAC only)
S
Attendant Direct Extension Selection
O
Attendant Forced Release V2; SLAC only)
Attendant Join (V2; SLAC only)
[FORCED RELEASE]
[JOIN]
S
S
Attendant Message Waiting (DTAC)
[ATT MSG]
S
Attendant Message Waiting (SLAC)
[ATTENDANT
MESSAGE WAITING]
Attendant Position Busy
[POS BIJSY]
S
S
Attendant Release
[RELEASE]
S
Attendant
[RTN-BUSY]‡
S
[RTN-DA]‡
[SOURCE]
S
S
[DEST]
S
Attendant Splitting One-Way Automatic
Attendant System Alarm Indication
[ALARM]
S
S
Message Center-Like Operation (V2; SLAC only)
Night Service
[NIGHT]
S
Return-Coverage-on-Busy
Attendant Return-Coverage-on-Don’t-Answer
Attendant Source/Destination (V2; SLAC only)
S
*
Feature
†
Bracketed words are the labels for button-activated features;
these labels are also used in feature descriptions where applicable.
‡
This button is assigned on the DTAC only.
types:
S=Standard;
O=Optional.
2-5
ACCOUNT CODE ENTRY
Description
Allows voice terminal users to associate an account code with incoming and outgoing calls.
This is accomplished by entering the account code at the voice terminal before hanging up.
The account code is appended to the SMDR call record and can be used later for accounting
or billing purposes.
To association account code with a call, the user, after completing a call but before hanging
up, must:
Single-Line Voice Terminal User:
●
— Flash the switchhook and dial *O; then dial the account code directly or dial a
System or Personal Speed Dial Number that contains the account code. If the
number is entered incorrectly, redial *0 and the correct number before
hanging up.
●
Multilane Voice Terminal User:
— Press Account Code Entry (ACCT ENTRY) button and then dial the account
code directly or dial a System or Personal Speed Dial Number that contains
the account code. A Repertory Dial (REP DIAL) button can also be used to
enter an account code. If the number is dialed incorrectly, press ACCT
ENTRY again (before hanging up) and dial the correct number.
●
When the correct number of account code digits have been entered (o r# is entered to
signal end-of-dialing), Confirmation Tone followed by Dial Tone is returned to the
user and the account code is appended to the SMDR call record.
Account Code Entry is optional.
Considerations
Account Code Entry provides an easy method of allocating the costs of specific calls (and
associated staff time) to the correct project, department or user. The account code is
appended to the SMDR call record and sent to the SMDR output channel.
Account Codes can include up to 15 digits.
The validity of the entered account code is not checked by the system.
If the user is active on a call, invoking the feature will drop the call.
Incorrectly dialed codes (prior to last digit entry) can be corrected by dialing *0 or pressing
ACCT ENTRY and reentering the code. Partial account codes entered by going on-hook
before completing entry are recorded and cannot be corrected.
If, before all digits have been entered, (1) the user goes on-hook, (2) a button other than
ACCT ENTRY is pressed, or (3) 30 seconds have elapsed since the feature was invoked, the
SMDR call record will show the digits dialed up to that point.
If a call is on hold, this feature cannot reinvoked.
2-6
Interactions
C o n f e r e n c e : If more than one user attempts to enter an account code on a
Conference Call, the first to enter a code will prevail.
●
Repertory Dialing: An Account Code can be stored on a REP DIAL button. Press
REP DIAL after ACCT ENTRY has been pressed.
●
Speed Dialing: An Account code can be stored in System or Personal Speed Dial
Number.
●
Transfer: A user can transfer a call to another user, then, before hanging up, enter
an account code. Subsequent account code entries for the same call will be ignored,
even though confirmation tone has been returned.
●
Administration Requirements
System:
●
Maximum number of Account Code digits (0-15)--Default = 15.
Voice Terminal: (Station Port)
●
●
Multiline terminals–Account Code Entry Button is required.
Single-line terminals–none.
Hardware Requirements
Requires an RS-232 compatible 80-column ASCII (serial) printer or other device to output
Station Message Detail Recording (SMDR)/Account Code entries.
2-7
ATTENDANT CALL EXTENDING
Description
Allows the attendant to put a call in a special hold condition, call another station, then
connect the two calls together. The attendant can withdraw from the connection and
separate the call from the console or remain connected to the other parties. Attendant Call
Extending is a feature used at either a Direct Trunk Attendant Console (DTAC) or a
Switched Loop Attendant Console (SLAC).
Note: In general, the attendant should not use the TRANSFER button, which invokes
the standard multiline voice terminal Transfer feature, to extend calls. If Transfer is
used, busy or unanswered calls cannot return to the attendant console for further
handling.
The attendant, after placing or answering a call, can use Step 1 or 2 to extend this call to an
inside extension or Step 1 to extend it to an outside number:
1. Press START to place the incoming call on hold via the Attendant Splitting OneWay Automatic feature. After receiving Dial Tone, the attendant then dials the
requested inside or outside number.
or
2. Press the Selector Console Group Select and Direct Extension Selection (DXS)
buttons associated with the requested inside station. This operation is equivalent to
pressing START and dialing the extension.
If ringing tone is heard, the attendant presses RELEASE (Manual Release) to connect the
caller to the ringing line and separate the call from the console. As an alternative, the
attendant can press any facility button such as System Access, Automatic Intercom, or an
outside line (Attendant Automatic Release) to complete the call extending procedure.
The attendant has the option of staying connected to the ringing line to announce the call
before connecting the two parties. The attendant can then release or (SLAC only) join the
other parties in a 3-way connection by using the Attendant Join feature.
If busy tone is heard and Attendant Camp-On (see associated feature description) is not
desired, the attendant presses CANCEL and is reconnected to the calling party.
If busy tone is heard on a call to an inside station and Attendant Camp-On is desired, the
attendant presses RELEASE or any facility button. The called party hears a tone burst, and
the call waits at the called voice terminal. When a busy single-line station goes on-hook, or
a busy multiline station System Access button becomes idle, the call automatically begins
ringing at the station. Only one Camped-On call is permitted per voice terminal.
Calls extended to an idle voice terminal that are not answered within a specified time return
to the Attendant Console on an idle LOOP button (SLAC only) or on the Return-On-Don’tAnswer (RTN-DA) button (DTAC only). Calls camped-on at a busy voice terminal that are
not answered within a specified time return to the Attendant Console on an idle LOOP
button (SLAC only) or on the Return-On-Busy (RTN-BUSY) button (DTAC only). If a SLAC
is not available to incoming calls (busy on another call, in Position Busy mode, etc.), a
returning call remains in the console queue until the console can handle it. If the Return
buttons on a DTAC are busy, an extended call remains at the called terminal until that
button becomes idle.
2-8
Considerations
Attendant Call Extending allows the attendant to utilize the additional attendant related
features such as Attendant Splitting One-Way (automatically places incoming canon hold),
Release, Cancel, Return-On-Don’t-Answer, Return-On-Busy, Forced Release (SLAC), Join
(SLAC), and Source/Destination (SLAC).
Interactions
Refer to the other Attendant Feature descriptions for information on related features (Table
2-E).
Administration Requirements
System:
●
●
Number of seconds before a Camped-On call returns to the Attendant Console (1-120
seconds), or No Attendant Camp-On (0) - Default = 30 seconds
Number of rings before unanswered call returns to the Attendant Console (1-31) Default = 5.
Hardware Requirements
Selector Console (optional)
2-9
ATTENDANT CAMP-ON
Description
Allows the attendant to extend a trunk call to a busy voice terminal and leave it waiting or
“camped on” there. After hearing busy tone, the attendant presses RELEASE to camp-on
this call at the busy terminal. When this is done, a burst of tone is heard in the handset of
the called terminal and the caller is placed on hold (hearing music-on-hold if available).
When a System Access button at a multiline set becomes idle or a single-line terminal hangs
up, the camped-on call is connected automatically and ringing begins. Only one call can be
camped-on to a voice terminal. This feature is referred to as a “Waiting Call” in the U s e r
Guides for the System 25 voice terminals.
Considerations
A camped-on call can be answered by a busy single-line user without losing
by momentarily pressing the switchhook (which places the current call on
dialing *9. Multiline terminal users cannot do this. However, if they have a
Originate Only button, they can place both calls on hold, go off-hook on that
*9 to pick up the camped-on call.
the current call
hold) and then
System Accessbutton and dial
If the camped-on call is not answered within a specified time, the call will be returned to the
Attendant Console in one of the following ways:
●
●
Switched Loop Attendant Console: The call returns to the common queue, where
it remains until the console can receive it at a LOOP button.
Direct Trunk Attendant Console: The call returns to the Return-On-Busy (RTNBUSY) button. If that button is busy, the call remains camped-on at the called
terminal until the RTN-BUSY button of the console becomes idle.
Interactions
●
●
●
Call Coverage/Direct Group Calling (DGC): If the called party is a member of
a hunt or Call Coverage group (or, for V1 systems only, a DGC group) and all
members of the group, or all receivers of the Coverage group are busy, the call will
not hunt or receive coverage. Once camped-on, calls will no longer hunt or receive
coverage even if the hunted-to station or group member becomes idle.
Direct Group Calling: For V2, the attendant can camp-on more than one call per
DGC group. For VI, the attendant can camp-on only one call per DGC group; if the
attendant attempts to camp-on a second call, it is immediately returned on the RTNBUSY button on the DTAC. Voice terminals in the group do not receive a burst of
tone when a call is camped on.
Direct Inward Dialing (DID): DID calls can be covered by the attendant and
then given Camp-On treatment. They do not automatically receive Call Waiting.
Refer to the Attendant Feature descriptions for information on other related features (Table
2-E).
2-10
Administration Requirements
System:
●
Number of seconds before a camped-on call returns to the Attendant Console (1-120
seconds) or No Attendant Camp-On allowed (0) - Default = 30 seconds.
Hardware Requirements
None
2-11
ATTENDANT CANCEL
Description
Allows the attendant to terminate an attempt to extend any incoming call if the called
station does not answer, or if the station answers but declines to accept the call. Before
pressing RELEASE, the attendant presses CANCEL and is automatically reconnected to the
calling party.
Pressing CANCEL when the Start facility is not active will be ignored.
Considerations
Attendant Cancel allows the attendant to terminate a call transfer attempt and return to
the incoming held party via a one-button operation. This enhances the attendant’s ability to
handle calls quickly and efficiently.
Interactions
None
Administration Requirements
None Required
Hardware Requirements
None
2-12
ATTENDANT CONSOLE, DIRECT TRUNK
Description
In System 25, the Attendant Console is used to answer incoming trunk calls that are not
directed to specific user stations, to answer calls from inside users, to extend calls to inside
stations and outside numbers, and to assist system users in placing outgoing calls and
setting up conferences. The attendant can also manage and monitor some areas of system
operation. System 25 R1Vl supports only the Direct Trunk Attendant Console (DTAC). The
R1V2 supports either the DTAC or the Switched Loop Attendant Console (SLAC), that is
described in the next subsection of this manual.
The DTAC (Figure 2-1) is a 34-Button Deluxe Voice Terminal administered with special
features, buttons, and capabilities to serve as an attendant position. In addition to the
attendant features, all standard multiline terminal features are also available. (Refer to the
Hardware Description section of this manual for a complete identification of the external
controls, indicators, and components of the basic voice terminal. )
OPTIONAL SELECTOR
CONSOLE
DIRECT TRUNK ATTENDANT
CONSOLE
Figure 2-1. Direct Trunk Attendant Console
The DTAC is always equipped with the following feature buttons that provide unique
attendant console functions. Each button has a green status LED that indicates when the
feature is activated.
●
●
●
●
Start [START] Initiates the call extending process by placing a caller on hold and
providing internal dial tone to the attendant
Cancel [CANCEL]: Terminates the “Start” operation and reconnects the attendant
to the calling party.
Release [RELEASE]: Releases the attendant from an active call and completes the
call extending process.
Return-On-Busy [RTN-BUSY]: Camped-on calls are returned to the console on
this button if not answered within a specified interval.
2-13
●
●
●
Return-On-Don’t-Answer [RTN-DA]: Extended calls not answered are returned
to the console on this button if not answered within a specified interval.
Attendant Message Waiting [ATT MSG]: Used by the attendant to remotely
control Message LEDs on voice terminals.
Alarm [ALARM]: The associated status LED flashes when a system trouble has
been detected; the LED can be changed from flashing to steadily lit by pressing the
button.
Two other attendant-only features are assigned to console feature buttons if required,
Position Busy [POS BUSY] and Night Service [NIGHT]. In a dual attendant console system,
Position Busy removes an Attendant Console from service. Only one of two consoles can be
in the “Position Busy” mode at a time. When Night Service is activated, attendant-seeking
calls can ring a night bell, can be directed to assigned voice terminals, or can be sent to a
night service announcement.
Considerations
Direct trunk operation means that trunks are terminated on individual buttons, called
Personal Line buttons, where outside calls are answered and originated. The console can
have several incoming calls ringing simultaneously.
Each console can also have an optional Attendant Direct Extension Selector Console to
enhance internal calling. The Selector Console is described in the “Attendant Direct
Extension Selection” subsection.
Dual Console Operation:
A System 25 can be equipped with up to two DTACs that operate simultaneously when both
are in service. If the system has two attendant consoles, one is called the first attendant
console (primary attendant console in VI); the other is called the second attendant console
(secondary attendant console in Vi). The calls in the following list will be routed to the first
attendant console.
●
●
●
Dial “0” calls
DID calls to unassigned numbers (when administered to route to the attendant)
Calls to Floating Personal Data Codes (FPDCs) not logged in (when administered to
route to the attendant)
If the first attendant has activated the Position Busy feature or is busy on both System
Access buttons, these calls will be routed to the second console. If that console is also busy
on both System Access buttons, busy tone is provided to the calling party.
For V2 only: See the “Call Coverage—Individual” feature description for information about
simulating additional System Access buttons for handling more incoming calls.
System users and DID callers can reach a particular attendant by dialing that personal Data
Codes (PDC).
Position Busy:
A POS BUSY button can be assigned
modes of operation: (1) simultaneous
(Note that only one console is allowed
BUSY status LED is lighted when the
to each console; this permits selection of one of two
operation or (2) only one Attendant Console active.
to be inactive at any given time.) An associated POS
console is inactive. Ringing is disabled on all trunk
2-14
terminations on the rightmost two columns of buttons of the inactive console. Ringers
disabled on an inactive console will be enabled on the active console for those trunks with
dual appearances (appearances on both consoles). All other features on all buttons,
including those on the associated Attendant Direct Extension Selector Console will continue
to function normally even though the console is inactive.
Interactions
A DTAC cannot operate in the same system with a SLAC.
Refer to the Attendant Feature descriptions for information on other related features as
listed in Table 2-E.
Administration Requirements
System:
●
Assign Primary and Secondary Attendant Positions (Vi)
●
Display attendant position number (first or second) (V2)
●
●
●
●
Assign number of rings before unanswered calls return to the Attendant Positional31) -Default = 5 rings
Send DID calls to unassigned numbers to the Attendant Position (Yes, No) -Default
= Yes
Send calls to Floating Personal Dial Codes that are not logged-in to the Attendant
Position (Yes, No) -Default = Yes
Assign number of seconds before an unanswered Camped-On Call returns to the
Attendant Console (1-120 seconds), or No Attendant Camp-On (0) - Default = 30
seconds.
Attendant Console: (Station Port)
●
Voice terminal type = 309 (V2)
●
Special Programmable Buttons:
— Night Service
— Position Busy
— Attendant Message Waiting (assigned by default).
Note: The following buttons are predefined on the Attendant Console and are
not administrable:
– Alarm
— Return-On-Don’t-Answer
— Return-On-Busy
— Start
– Cancel
2-15
— Release
●
Trunk terminations–The following is required for each trunk terminated on the
console (administered as Personal Line appearances; DID trunks cannot be
terminated on a DTAC):
Trunk Number
— Make this the Principal Station (owner) of the trunk (Yes, No)
— Enable Ring (Yes, No).
Hardware Requirements
Each console requires a port on a ZTN79 ATL Line circuit pack.
Figure 2-2 provides a connection diagram for the DTAC.
SYSTEM 25
CABINET
ZTN79
HYBRID
LINE CP
PART OF
OCTOPUS CABLE
— — — —
— — — —
Ž
PART OF SIP
SIP
ADAPT
W1
B1
C1
DIRECT TRUNK
ATTENDANT
CONSOLE T1
— — — —
— — — —
— — — —
LEGEND :
B1 —
C1 –
C2 –
T1 –
W1 –
*–
TYPICAL-103A CONNECTING BLOCK*
MODULAR CORD (D8W-87) - FURNISHED WITH SET
OCTOPUS CABLE (WP90780) - PEC 2720-05P
7305H02B DELUXE 34-BUTTON VOICE TERMINAL - PEC 3162-417
4 PAIR INSIDE WIRING CABLE*
FURNISHED BY INSTALLER
RANGE: WITHIN 2000 FEET OF SYSTEM CABINET (LOCAL POWER REQUIRED >1000 FEET)
Figure 2-2. Direct Trunk Attendant Console Connections
2-16
ATTENDANT CONSOLE, SWITCHED LOOP (V2)
Description
In System 25, the Attendant Console is used to answer incoming trunk calls that are
specified to ring at an attendant position, to answer calls from inside users, to extend calls
to inside stations and outside numbers, to assist system users in placing outgoing calls, and
to set up conferences. The attendant can also manage and monitor some areas of system
operation. System 25 R1V1 supports only the Direct Trunk Attendant Console (DTAC),
which is described in the preceding subsection of this manual. The R1V2 supports either the
DTAC or the Switched Loop Attendant Console (SLAC), but not both in the same system.
The SLAC (Figure 2-3) is a 34-Button Built-In Speakerphone (BIS) Voice Terminal with a
16-character display module. It is administered with special buttons, features, and
capabilities to serve as an attendant console. In addition to the attendant features, most
standard multiline terminal features are also available. (Refer to the “Hardware
Description” section of this manual for a complete identification of all the external controls,
indicators, and components of the basic voice terminal. )
OPTIONAL SELECTOR
CONSOLE
SWITCHED LOOP
ATTENDANT CONSOLE
Figure 2-3. Switched Loop Attendant Console
Associated with the SLAC are message center-like capability and display support. The
message center feature provides for efficient handling of calls that should be sent to message
takers. These calls are directed to a message center console position through administration
of call type translations. Display service allows identifiers (names) to be assigned to
extension numbers and trunks. The system then displays the appropriate information to the
attendant when calls are processed at the console.
Fixed Buttons (Figure 2-4)
The SLAC has five fixed line appearance, or “LOOP,” buttons where all incoming calls are
answered. Each button has a red I-use LED and a green status LED. These buttons
represent voice links (loops) between the console and the switch. The loops also provide the
paths for outgoing calls.
2-17
In addition to the LOOP buttons and standard multiline terminal buttons (HOLD,
TRANSFER, etc.), the console is equipped with the following feature buttons that provide
unique attendant functions. Unless noted, the buttons have green status LEDs.
●
●
●
●
●
●
●
●
●
●
●
●
●
●
Start [START]: Initiates the call extending process by placing a caller on hold (on
the Source button) and providing internal dial tone to the attendant. No LED.
Cancel [CANCEL]: Terminates the “Start” operation and reconnects the attendant
to the calling party (on the Source button). No LED.
Release [RELEASE]: Releases the attendant from an active call and completes the
call extending process. No LED.
Source [SOURCE]: Reconnects the attendant to the calling party after a call has
been initiated to the called party but before the two parties have been connected
together.
D e s t i n a t i o n [ D E S T ] : Connects the attendant to the called party again after the
attendant has operated the Source button to speak to the calling party.
Join [JOIN]: Joins together (in a 3-way connection) the attendant and the other
parties in an extended call. No LED.
Forced Release [FORCED RELEASE]: Drops all active parties from a call. No
LED.
Last Number Dialed [LAST # DIALED]: Redials the last number dialed. No
LED.
Position Busy [POS BUSY]: Temporarily removes the attendant position from
service.
Attendant Message Waiting [ATTENDANT MESSAGE WAITING]: Used by
the attendant to remotely control Message LEDs on voice terminals.
Alarm [ALARM]: The associated status LED flashes when a system trouble has
been detected; the LED can be changed from flashing to steadily lit by pressing the
button.
I n s p e c t [ I N S P E C T ] : Puts the display into a mode for inspecting the status or
stored information of certain buttons.
Scroll [SCROLL]: Causes display to present additional call information, when
available. No LED.
Local [LOCAL]: Allows display to be used for clock and calendar functions. No
LED.
The buttons not assigned to normal voice terminal functions or to attendant functions are
defaulted to the Flex DSS feature. One of these programmable buttons can be assigned to
Night Service, if the feature is required, and any of the others to multiline voice terminal
features.
Programmable Feature Buttons (Figure 2-4)
The features in the following list can be assigned to the programmable feature buttons. On
the SLAC these buttons are not equipped with lamps for indicating feature status conditions.
●
Exclusion
2-18
●
Manual
Signaling
●
Transfer
to
●
Account Code Entry
●
Auto Intercom
●
Auto A n s w e r
●
Direct
Facility
Access
●
Direct
Station
Selection
●
Flex
●
Repertory
Data
(DSS)
DSS
Dialing.
The last two features can be programmed with dialable numbers by the attendant. When a
call is placed using a Flex DSS button or a Repertory Dial button, one of the five switched
loops is automatically selected for routing the call to the switch.
Display (Figure 2-4)
The SLAC contains an alphanumeric call information display. This module is built into the
top of the console. It contains a 16-character 5x7 dot matrix liquid crystal display, timer
controls, and a thumbwheel Contrast adjustment. Timer functions are available only when
the attendant presses the Local button. The Time/Timer Exit button allows the user to select
ordinary clock/calendar display or a timer. In the Time mode, Set, Fwd, and Rev are used to
set the clock. In Timer mode, Start and Stop are used to time events.
The primary purpose of the console display is to provide the attendant with descriptive
information about incoming and outgoing calls. This information includes extension numbers
and associated names, trunk identifiers, reasons for call return and redirection, and number
of calls waiting in the queue for service. Refer to the “Attendant Display” feature
description, for a detailed discussion of call information displays.
Considerations
An R1V2 system configuration can support either one or two SLACs or one or two DTACs,
but not a combination of a SLAC and a DTAC.
The Switched Loop Console derives its name from the ability of the system to hold incoming
attendant-bound calls in a queue and switch them on voice loops to an available console.
Calls are directed to a console in a preadministered, prioritized sequence. The SLAC differs
from the DTAC in the following basic respects:
●
It receives calls one at a time, regardless of the number of incoming calls to the
system (at the DTAC, many incoming calls can be ringing simultaneously).
●
It displays pertinent information about incoming and outgoing calls.
●
It can serve as an attendant console, a message center, or a combination of both.
●
It has speakerphone and Hands-Free Answer on Intercom (HFAI) capabilities.
The optional Direct Extension Selector Console can be connected to a SLAC to provide
busy/idle status of inside stations and quick calling of their extension numbers. In a system
with two consoles, either or both can have a Selector Console as an adjunct. The Selector
Console is described in the “Attendant Direct Extension Selector Console” subsection.
2-19
Switched Loop Operation
All calls that are intended for an attendant position are first routed by the system to a
common queue where they wait to be sent to a console. In a configuration having two
consoles, the same queue serves both consoles. When an attendant console becomes available
to receive a call, the system removes a call from the queue and directs it to an idle loop on
the console. Calls are selected from the queue on the basis of “first in/first out” and in
accordance with administered priorities. An available attendant console is one that is not
active on a call, has no calls ringing, has at least one LOOP button idle, is not in Position
Busy or Inspect mode, and is not in a split condition.
In a two-console arrangement, each console can be administered to receive all types of calls
or to receive only specific types. A call that can be received by either position goes to the
first available attendant; when both are available, the call goes to the attendant who has
been idle the longest time. If one of the two consoles is in “Position Busy” mode, all calls
(except Attendant PDC, Attendant PDC via DID, and DGC calls) direct to the other one.
Timer
Start
Set
RED
I-USEGREEN
STATUS
Loop
stop
Time/
Timer
Rev
Exit
●
●
-
-
-
-
-
-
-
-
-
-
-
-
Contrast
Alarm
Local
Scroll
● Loop
●
● Pos
● Busy
Flex DSS
Forced
Release
● Loop
●
● Flex
DSS
●
Flex DSS
Last #
Dialed
● Loop
●
Flex DSS
Flex DSS
● Loop
●
●
Source
●
●
Des t
●
Flex DSS
Flex DSS
Conference
Transfer
Flex DSS
Flex DSS
Flex DSS
Flex DSS
Flex DSS
Flex DSS
Flex DSS
Flex DSS
Drop
Hold
.
Cancel
Start
Attendant
Message
Waiting
.
Figure 2-4. Console Buttons and Display
2-20
Join
Release
Inspect
-
An incoming call from the queue to a console appears on one of the five LOOP buttons; the
attendant is alerted to the call by audible ringing, a steadily lighted red lamp, and a flashing
green lamp. While the call is ringing and while the attendant is handling the call, the
system will direct no more calls to the console. After the attendant ends or releases the call
or puts it on hold, another call can come in on an idle button.
It should be emphasized that even when all LOOP buttons on a console are idle, only one call
can be directed from the queue to the console. If the attendant puts a call on hold, that
LOOP button is no longer available, but a new call can come in on another button that is
idle.
Answering a Call on a LOOP Button
At an available SLAC, an incoming call will ring at an idle LOOP button automatically
selected by the system. The attendant has only to lift the handset to answer the call;
pressing the button is not necessary.
Placing a Call on a LOOP Button
In general, originating a call at an idle SLAC involves going off-hook and then dialing the
desired number. However, if Prime Line Preference has been changed to a non-LOOP button,
the attendant will have to press an idle LOOP button before dialing.
If the console is not idle, the attendant can generally use one of these procedures:
●
●
●
Split the active call (that is, put it on temporary hold by pressing the Start button)
and place another call on the same button; this is the normal call-extending
procedure.
If the Automatic Hold feature is enabled, press another LOOP button to place a new
call; the first call goes on hold.
If the console does not have Automatic Hold (that is, it has the default Automatic
Release), use the Hold button to put the active call on hold; then select a new loop to
place a new call.
Dual Console Operation
A System 25 can be equipped with
both are in service. Both consoles
administered to receive only certain
Busy below), most calls are directed
message center.
up to two SLACs, which operate simultaneously when
can receive the same types of calls, or each can be
types. When one console is out of service (see Position
to the other. Either or both consoles can function as a
Inside users can reach either attendant by dialing O, or a particular one by dialing the
attendant’s PDC. The DID callers use the Attendant DID number.
Position Busy
Operation of the Position Busy button by the attendant makes the console unavailable to
most incoming calls from the common queue and directs the calls to another answering
station. The only types of calls that are not diverted by Position Busy are Attendant DID,
DGC, and PDC calls. The placing of outgoing calls is not affected. When the Position Busy
condition is active, the green status lamp of the button lights steadily. Position Busy is
similar to the Send All Calls feature, that is not administered on the SLAC.
The Position Busy feature is automatically assigned to a button position (see Figure 2-4)
when the console is administered. In a one-console configuration, however, the feature is
enabled only if a multiline voice terminal in the system is administered as a receiver of calls
2-21
from the common queue while the console is unavailable. If this is not done, the button
should be reassigned to another of the permissible features.
In a two-console configuration, an attendant in Position Busy mode will be covered by the
other attendant. Only one console can be in Position Busy mode at a time.
Call Types
The following types of attendant-seeking calls are sent to the common queue and then
directed to an idle LOOP button at a console:
●
●
●
●
Dial Operator calls (placed from inside stations by dialing O)
Call Following calls logged in at the console
Calls to Floating Personal Dial Codes (FPDCs) that are not logged in at a specific
station
Direct Inward Dialing (DID) calls to numbers that are not assigned to specific
stations
●
Attendant DID calls, including Direct Group calls
●
Calls to the attendant’s Personal Dial Code (PDC), including DGC calls
●
Coverage calls for which the common queue is a covering receiver
●
Returning calls.
The order in which calls are serviced is established by system administration. Each type of
call is assigned a priority that determines its position in the common queue with respect to
other types. System administration also establishes where the calls go. Obviously, in a oneattendant system, all calls automatically go to that attendant. If a system has two
attendants, however, administration can direct calls of each type (with the exceptions noted
below) to either position or to both positions.
Call Following and Attendant PDC calls can be assigned priorities but cannot be directed to
a specific attendant in a two-position system. Any trunk except types 901-902 and 1003-1008
(DID and Dial-in Tie Trunks) can be assigned a priority and be directed to a specific
attendant position or to both.
Calls accessed by dialing a code [call pickup at other extensions, Trunk-Answer-from-AnyStation (TAAS) Night Service calls, and calls parked by other stations] are originated at a
LOOP button and brought to the console on that same button. These calls do not enter the
common queue.
Ringing
The SLAC receives normal ringing on incoming calls. Abbreviated alerting (one short burst
of ringing), accompanied by a change in the LOOP button wink rate, indicates to the
attendant that a held call has exceeded the preset hold time interval. Abbreviated alerting
can also be administered as a reminder for new calls entering the queue.
Interactions
A SLAC cannot operate in the same system with a DTAC.
Connection of a headset adapter to the SLAC allows the optional use of a headset instead of
the handset in handling calls.
2-22
Call Originations
Placing a new call from an active console causes interactions with the currently active call.
At a console that has the default Automatic Release feature, the active call is dropped when
the attendant presses anew LOOP button. The new loop becomes the active one; dial tone is
provided, and the attendant can dial a number. If the optional Automatic Hold feature is
administered, the interrupted call is put on hold instead of being lost.
If anew call is originated with the START button or at the Selector Console, the active call
is split. The current loop becomes the active loop for the new call. The display shows the
split call information. This is the normal operating procedure for extending calls.
When a Rep Dial button is pressed while the console is active on a call, the active party is
not dropped or split, and the display does not change. If the active call is with an inside
station, pressing Rep Dial has no effect. But if the active call is on an outside trunk, then
pressing Rep Dial will cause the digits stored on the button to be sent out over the trunk
(“thru-dialing” or “end-to-end signaling”).
When an active call is put on hold manually by operation of the HOLD button, the system
does not automatically select a new loop for placing a call. In this case, the attendant can
select a new loop by pressing an idle LOOP button, then dial a number. Pressing a DSS, Flex
DSS, Auto Intercom, Last Number Dialed, Rep Dial, or Selector Console button will select a
new loop and dial a number in a single operation.
Attendant Features
Refer to the Attendant Feature descriptions for information on other related features (Table
2-E).
Administration Requirements
Station Port:
●
Assign voice terminal type.
Assign Prime Line Preference (default: top LOOP button).
●
●
Assign flexible buttons.
Enable ring reminder when calls enter queue.
●
Trunk Port:
●
Assign priorities to calls directed to the console queue.
●
Assign the attendant(s) to handle calls from this trunk.
●
Assign unique trunk identifiers.
System:
●
●
Assign DID number for attendant “0” treatment.
Assign Call Coverage Group number for which the console queue is to serve as a
receiver.
●
Assign Automatic Hold or Automatic Release.
●
Assign Hold timer interval.
●
Assign destination of held calls that time out.
2-23
●
Assign call type priorities and attendant specification.
●
Assign call types and
●
Assign Position Busy “backup’’ station, inapplicable.
attendant
two-console configuration.
specification
for
Message
Center-Like
operation
in
Hardware Requirements
Each console requires a port on a ZTN79 ATL Station circuit pack.
Figure 2-5 provides a connection diagram for the SLAC.
SYSTEM 25
CABINET
ZTN79
HYBRID
LINE CP
PART OF
OCTOPUS CABLE
—
—
——
.
—
—
——
C2
PART OF SIP
SIP
ADAPT
—
—
——
—
—
——
—
—
——
W1
C1
SWITCHED LOOP
ATTENDANT
CONSOLE T1
LEGEND :
TYPICAL-103A CONNECTING BLOCK*
MODULAR CORD (D8W-87) - FURNISHED WITH SET
OCTOPUS CABLE (WP90780) - PEC 2720-05P
7305H04C BUILT-IN SPEAKERPHONE VOICE
TERMINAL - PEC 3162-DIS
W1 - 4 PAIR INSIDE WIRING CABLE*
* - FURNISHED BY INSTALLER
B1 C1 C2 T1 -
RANGE : WITHIN 2000 FEET OF SYSTEM CABINET (LOCAL POWER REQUIRED >1000 FEET)
Figure 2-5. Switched Loop Attendant Console Connections
2-24
a
ATTENDANT DIRECT EXTENSION SELECTION
Description
Permits the attendant to extend calls to stations by pressing one or two buttons instead of
pressing START and dialing the PDC or DDC. Each attendant console in a system can have
an associated Direct Extension Selector Console. A Selector Console can be used with either
a Direct Trunk Attendant Console (DTAC) in an R1V1 or R1V2 system or with a Switched
Loop Attendant Console (SLAC) in an R1V2 system.
The Selector Console is also used by the attendant for simply calling inside stations, in
addition to “extending” calls.
The Selector Console (Figure 2-6) has an array of 100 Direct Extension Selection (DXS)
buttons plus seven Group Select buttons and a Test button. The DXS buttons are labeled 00
through 99. Pressing a Group Select button causes the DXS buttons to be associated with
PDCs from an associated hundreds group. Default assignments for the Group Select buttons
are 200-299, 300-399, etc., up to 800-899. Group Select buttons can be assigned any hundreds
group in the dialing plan.
Pressing a DXS button when off-hook on an incoming call is equivalent to pressing START
and dialing a station. Such action will busy out the Start facility until the call is released.
The Selector Console can be used to monitor the on-hook/off-hook status of stations in the
system. If the attendant, while on-hook, presses a Group Select button, the Group Select
LED and the LEDs of any busy stations in that group will light steadily.
The DXS button LED (and, in R1V2 systems, the Group Select button LED) associated with
a particular station will flash when one of the following events occurs:
●
●
●
The station calls the attendant
A call extended by the attendant to the station returns on a RTN-BSY or RTN-DA
button (DTAC only) or on a LOOP button (SLAC only)
The station is covered and a call to it is redirected to a COVER button (DTAC only)
or to a LOOP button (SLAC only).
The LEDs stop flashing when the call is answered. When the attendant answers a returning
call, the LEDs will return to the state that reflects the current on-hook/off-hook status of
the station. In all of the above cases, the Group Select lamp associated with the current
“hundreds page” remains lighted steadily.
An outside call can be parked via the Selector Console by pressing one of the eight DXS
buttons that can be programmed with Call Park extension numbers. On the DTAC, the
facility status LED of the parked call winks (to indicate that the call is on hold) and the
status LED on the Selector Console lights steadily. On the SLAC, the call is removed from
the attendant console, with the Selector Console LED lit steadily.
A call parked via the Selector Console and not picked up within 2 minutes will return to the
console. The status LED of the parked-on button will flash while the call is ringing the
attendant.
A call parked via the Selector Console can be picked up at any voice terminal by dialing the
Call Park retrieval code (*8) and the number on which the call is parked.
2-25
❑ ❑ ❑ ❑ ❑
81
84
82
83
80
❑
❑
❑
❑
❑
72
74
71
70
73
❑
❑
❑
❑
❑
62
63
60
64
61
❑
❑
❑
❑
❑
53
52
54
51
50
❑
❑
❑
❑
❑
41
42
43
44
40
❑
❑
❑
❑
❑
31
30
32
33
34
❑
❑
❑
❑
❑
23
24
22
20
21
❑
❑
❑
❑
❑
11
13
14
12
10
❑
❑
❑
❑
❑
02
03
01
04
00
❑
❑
❑
❑
❑
91
90
❑
93
92
❑
❑
94
❑
❑ ❑ ❑ ❑
88
86
87
85
❑
❑
❑
❑
78
75
76
77
❑
❑
❑
❑
68
66
65
67
❑
❑
❑
❑
56
57
55
58
❑
❑
❑
❑
48
45
46
47
❑
❑
❑
❑
38
36
35
37
❑
❑
❑
❑
28
26
27
25
❑
❑
❑
❑
18
15
16
17
❑
❑
❑
❑
05
06
08
07
❑
❑
❑
❑
96
95
97
❑
❑
❑
\
99
98
89
❑
79
❑
69
❑
59
❑
❑
39
❑
29
❑
19
❑
09
❑
/
49
100
DXS
BUTTONS
WITH LEDS
❑
❑
/
\
GROUP SELECT BUTTONS
AND ASSOCIATED LEDS
TEST
BUTTON
NOTE:
STATUS LEDs are located
to the left of each DXS
button (00-99) under
transparent front cover.
Figure 2-6. Model 23A1 Attendant Direct Extension Selector Console
2-26
The rightmost button on the bottom of the console is a Test button. When it is pressed, all
DXS LEDs will light sequentially; a second press allows individual LEDs to be tested and a
third press ends the test.
Considerations
When there are two Attendant Selector Consoles in the system the Group Select button
assignments are identical. Whenever an administrative change is made to one console, the
other console is automatically changed.
Buttons on the Selector Console point to either station PDCs, FPDCs (FPDCs), Call Park
codes, DGC access codes, or (V2) pooled facilities. Calls extended by the Selector console are
directed as described in the “Personal Dial Codes’’ feature description.
When a station calls the attendant, the associated LED on the Selector Console will flash
while the call is ringing and will light steadily when the attendant answers the call. The
LED will light steadily whenever the terminal is off-hook. Station busy indication is not
provided for buttons pointing to FPDCs.
If a call to a PDC is directed to a COVER or LOOP button on the Attendant Console, the
covered status LED of the voice terminal on the Selector Console will flash and then go dark
when the call is answered by the attendant. If the covered call was intended for a FPDC
that was logged in at a terminal with attendant coverage, the Selector Console status LED
associated with the FPDC (if assigned) will flash. In this case, the Cover button status LED
will also flash (DTAC only).
A call can arrive at an Attendant Console SYSTEM ACCESS or LOOP button because the
PDC or FPDC is logged in at the Console or because the FPDC is not logged in. For these
calls, the status LED on the Selector Console will not light.
If the attendant extends a call to a station or DGC group and that call returns to the
attendant, the status LED of the called station or group on the Selector Console will flash
and then go dark when the call is answered by the attendant. This is true regardless of the
login status of the PDC.
Interactions
●
.
Attendant Position Busy: The Selector Console functions normally when the
associated Attendant Console is in the inactive mode.
Attendant Return-Coverage-On-Busy/Or-Don’t-Answer: If a call to a FPDC
is returned to the attendant on a RTN-BUSY or RTN-DA button or on a LOOP
button, the status LED of the FPDC on the Selector Console will flash during ringing
and go dark when answered.
. Call Coverage: If the attendant receives a coverage call for a FPDC, the associated
status LED on the Selector Console will flash and then go dark when the call is
answered by the attendant.
●
Direct Group Calling: When all stations in a DGC group are busy, the DXS status
LED on the Selector Console lights.
2-27
Administration Requirements
Special Feature Ports:
●
●
Requires a port assignment on a ZTN79 ATL Station Circuit Pack (CP) for each
Selector Console.
Assign Group Select button hundreds groups.
System:
●
Assign Attendant Call Park codes.
Hardware
Requirements
Requires an Attendant Selector Console, and a port interface on a ZTN79 ATL Station CP.
The Selector Console requires a KS-22911, List 1 Power Supply, associated 115V ac power
outlet, and a 400 B-type Adapter. The 400B2 Adapter provides power to the console at the
wall jack. The Console connects to a port on the ZTN79 ATL Station CP.
Detailed connection information is provided in Figure 2-7. Descriptions of the Station
Interconnect Panel(SIP), Trunk Access Equipment (TAE), and associated cables and
adapters, as shown on the figures, are provided under the heading “Connectivity” in Section
4.
SYSTEM 25
CABINET
PART OF
OCTOPUS CABLE
PART OF SIP
—
—
——
ZTN79
HYBRID
LINE CP
.
SIP
ADAPT
C2
—
—
——
—
—
——
—
—
——
W1
B1
400B2
ADAPT
C1
B2
ATTENDANT
SELECTOR
CONSOLE T1
C7
-48VDC
P1
LEGEND :
B1
B2
C1
C2
C7
P1
T1
W1
*
-
TYPICAL-103A CONNECTING BLOCK*
40062 ADAPTER - FURNISHED WITH CONSOLE
MODULAR CORD (D8W-87) - FURNISHED WITH CONSOLE
OCTOPUS CABLE CABLE (WP90780) - PEC 2720-05P
CORD D6AP-87 - FURNISHED WITH CONSOLE
KS22911 POWER SUPPLY, - FURNISHED WITH CONSOLE
23A1 DXS CONSOLE - PEC 62509
4 PAIR INSIDE WIRING CABLE*
FURNISHED BY INSTALLER
RANGE : WITHIN 2000 FEET OF SYSTEM CABINET
(LOCAL POWER REQUIRED >1000 FEET)
Figure 2-7. Attendant Direct Extension Selector Console Connections
2-28
ATTENDANT DISPLAY (V2; SLAC Only)
Description
Provides displays of call-related information on a 16-character screen at the top of the
Switched Loop Attendant Console (SLAC). The following types of data are automatically
presented for calls handled at the console:
●
The extension number and name of an inside party called from the console
●
The extension number and name of an inside party calling the console
●
Called extension number and trunk identification on incoming trunk calls
●
Digits dialed on outgoing trunk calls
●
Calling and called party identification on internal coverage calls
●
Call type and reason for return or redirection
●
Number of calls waiting in the common queue to be serviced.
A secondary, or Inspection, mode allows the attendant to make the console temporarily
unavailable to incoming calls and get information about held calls, trunk availability,
numbers assigned to Repertory Dialing, Automatic Intercom, DSS, or Flex DSS buttons, and
the number currently stored on the Last Number Dialed button. The display module
provides clock/timer functions in the Local mode.
Call Information Display
The console display can generate up to two screens of information. Screen 1 is the normal
display presented for all calls. No action by the attendant is required to access this screen.
Screen 2, activated by pressing the SCROLL button, contains support information about a
call, such as the calling party on a coverage call. If Screen 2 does not appear after SCROLL
is pressed, then no additional information is available for viewing.
●
Extension number and calling/called party name
●
Redirection descriptors:
> for a coverage call
} for a returning call
& for a call that is being extended.
●
Call type indicators that track call status or inform the covering attendant why the
user did not answer:
s for Send All Calls
b for call to busy station
n for no answer
d for DGC call
f for Call Following
2-29
p for Call Park
o for not logged-in FPDC.
●
Number of calls waiting in the common queue (l-9 or ! for 10 or more)
●
Trunk identification for incoming outside calls
●
Conference information.
Considerations
The attendant display provides pertinent information about the currently active call. The
following examples show some typical screens.
Incoming DID call:
2085 DID
3
The first four digits are the DID number assigned to the attendant. The digit 3 is the
number of calls waiting in the queue.
Incoming call from a system user:
1463 Pearson, M
The first four digits are the call ing user’s extension number. No calls are in the queue.
Incoming coverage call:
>1566 Davis, T 5
A call to T. Davis on extension 1566 is directed to the common queue for coverage. The
attendant can determine the caller’s identity by pressing SCROLL. The following display
appears on screen 2.
2381 Harmon, B s
The call type indicator s means that Harmon’s call was sent to coverage because Davis
activated the Send All Calls feature.
2-30
Returning parked call:
I
4
}801 Park
The call parked on extension 801 at the Selector Console is returning to the attendant.
Call from console to system user:
1728 Burns, R !
The extension number and name of the called party are presented. The ! symbol indicates
that 10 or more calls are waiting in the queue.
Outgoing trunk call:
912126378888
The display shows the digits (up to 14) of the dialed outside number. Digits in excess of 14
are displayed on screen 2.
Call from inside station extended to another inside extension:
2344 Carter, M 7
The attendant answers the incoming call, and the caller, Carter, asks to be connected to
another inside party. When the attendant presses START, screens 1 and 2 provide the
following information:
7
&
2344 Carter, M &
2-31
The call from Carter has been split while the call is extended. After the attendant dials the
desired PDC, screen 1 displays called party identification, and screen 2 remains unchanged:
&1397 Phelps, T 7
2344 Carter, M &
If the attendant now presses RELEASE, both screens go blank. If the attendant presses
JOIN, the two parties and the console are connected together; screen 1 displays conference
information:
CONFERENCE 3 7
Conference set up by attendant:
After the attendant has called all the conferees and connected them together on a LOOP
button, screen l displays the type of call and the number of parties, including the console:
CONFERENCE 4 7
If the attendant releases, the screen goes blank.
Inspection Mode
Pressing the INSPECT button puts the console in the inspection mode. In this condition the
console is unavailable to incoming calls and the attendant can get information about held
calls, trunk availability, and numbers assigned to certain buttons. Pressing the INSPECT
button a second time will restore the console to normal operation, as will a change of
switchhook state (on-hook to off-hook, or vice versa), selection of an idle loop, or 15 seconds
of inactivity (timeout).
While the inspection mode is active, the attendant can press any of the following buttons to
provide special information on display screen 1.
. Loop button where a call is ringing or on hold–The original message that appeared
when the call first arrived at the console is displayed; for a multiparty call,
“CONFERENCE N” N=number of conferees) is shown.
. Direct Facility Access button–The display shows the number of trunks that are busy
within the selected trunk group; the format of the message is “XX of YY BUSY. ”
. Repertory Dial, Automatic Intercom, DSS, Flex DSS, or Last Number Dialed–The
number stored on the button is displayed.
2-32
P r e s s i n g a n y b u t t o n o t h e r t h a n t h o s e listed above causes a message such as “NO
INFORMATION” to be displayed.
Timer Functions
The LOCAL button allows the attendant to have access to the Timer controls on the left side
of the display module. In the local mode, the screen can provide clock, calendar, timer, and
alarm functions. The local mode is canceled when the LOCAL button is pressed again, a new
call arrives, or a call is placed. Refer to the customer instruction booklet packed with each
console for detailed information on using the Timer controls.
Interactions
The display provides a display for each call placed or answered at the console.
Administration Requirements
Directory type information must be administered in the form of user and group names
corresponding to extension numbers and identifiers for incoming trunks. Although system
administration allows up to eleven characters for display IDs, only the first nine characters
are shown on the SLAC display.
Hardware Requirements
None
2-33
ATTENDANT FORCED RELEASE (V2; SLAC Only)
Description
Drops all active parties from a call in which the attendant and one or more other parties are
connected together. The attendant uses the feature by pressing the FORCED RELEASE
button while connected to other callers in a conference-type call. The other parties will be
disconnected from the console and from each other. After Forced Release has taken place,
the attendant can receive anew call from the console queue or place a call.
Considerations
Forced Release differs from Release in an important respect. Simple Release separates the
attendant from an extended call or a conference call, but leaves the other parties connected
together; Forced Release completely disconnects all parties.
When the attendant is connected to only one other party, Forced Release has the same result
as Release (that is, the call is ended). If the attendant has already Released, Forced Release
has no additional effect.
Interactions
Refer to the Attendant Feature descriptions for information on related features (Table 2-E).
Administration Requirements
None
Hardware Requirements
None; the FORCED RELEASE button is a standard fixed feature button on the SLAC.
2-34
ATTENDANT JOIN (V2; SLAC Only)
Description
Allows the attendant, while extending an incoming call, to connect together the calling
party, the called party, and the console in a 3-way call. All parties can talk to each other.
The attendant activates the feature by pressing the JOIN button while still connected to one
of the other parties.
Considerations
The join feature cancels a split condition.
The attendant can use the Join feature to stay on unextended call and give assistance to the
other parties. A joined call can be expanded into a conference call by adding more parties.
Interactions
Once the JOIN button has been pressed, there is no way to “unjoin” the calling and called
parties (that is, separate them back into SOURCE and DEST).
Refer to the Attendant Feature descriptions for information on related features (Table 2-E).
Administration Requirements
None
Hardware Requirements
None; the JOIN button is a standard fixed feature button on the SLAC.
2-35
ATTENDANT MESSAGE WAITING
Description
Allows the attendant to remotely control the status of Message LEDs on user stations.
Considerations
This feature allows the attendant to notify stations that a message is available for them.
The attendant can activate the Message LED of the station while either (1) ringing, (2)
receiving Busy Tone, or (3) talking to a station. The status of the Message LED of the called
party is reflected by the green status LED of the Attendant Message Waiting button in any
of these cases.
Note: The Attendant Message Waiting button on the Direct Trunk Attendant Console
(DTAC) is labeled ATT MSG. On the Switched Loop Attendant Console (SLAC), the
name is completely spelled out.
To activate (light) a user’s Message LED in any of these cases, the attendant presses the
Attendant Message Waiting button. (If the signaled voice terminal is not equipped with a
Message LED, the attendant’s LED will remain dark.)
If the attendant presses the button a second (or third) time before hanging up, the user’s
Message LED will turn Off (and back On), etc.
The red I-Use LED associated with the Attendant Message Waiting button on the DTAC is
inoperative.
The attendant can turn On or turn Off a user’s Message LED without disturbing the user by
going off-hook on a System Access or Loop button, pressing the Attendant Message Waiting
button, and then dialing the station. Confirmation tone is returned, and the user’s Message
LED will turn on. To turn it off, press the button again.
This feature is not the same as the Station-To-Station Message Waiting or the Call Coverage
Message Waiting features. Refer to the “Messaging Services” feature description for a
summary of all system Messaging Services.
Interactions
●
Conference: Pressing ATT MSG while on a conference call will be ignored.
●
Coverage Calls: The attendant can light the Message LED of the covered station
when receiving a coverage call for the station.
●
Hands-Free Answer On Intercom: If the attendant lights the Message LED on a
voice terminal with AUTO ANS button active, the auto-answer function will turn off,
allowing subsequent calls to receive coverage as assigned.
Administration Requirements
Attendant Position (Station Port):
●
Assign ATT MSG button (defaulted).
Hardware Requirements
Stations must have a Message indicator (not assignable).
2-36
ATTENDANT POSITION BUSY
Description
Allows an Attendant Console to be placed in an inactive mode.
Systems with Direct Trunk Attendant Console(s) (DTAC)
There must be two Attendant Consoles in the system before this feature can be activated. A
Position Busy (POS BUSY) button can be assigned on each of the consoles. Pressing POS
BUSY at one of two active consoles causes the POS BUSY status LED to light and the
console to be placed in the inactive mode. Pressing POS BUSY a second time causes the
LED to go dark and the console to be reactivated. Pressing POS BUSY when only one
Attendant Console is active is ignored (i.e., only one console is allowed to be inactive at a
time. )
When a console is in the inactive mode, ringing is disabled on facility appearances on the
two rightmost button columns only. The (green) status LEDs will continue to operate
normally. Calls to floating PDCs not logged in, DID calls, and dial “O” calls will be
transferred to the active console. Internal calls to the PDC of the inactive console will still
be directed to that console.
Incoming calls on lines that normally ring at only the inactive console will now ring at the
active console if they have an appearance there.
All buttons on the inactive console will continue to function normally, including the Selector
Console buttons. Calls can be originated by the inactive console. Call appearances in the
leftmost two columns of buttons on the inactive console are not affected by the Position Busy
feature.
The attendant can press a Direct Station Selection (DSS), Automatic Intercom (AUTO
ICOM), or a Direct Facility Access (FACILITY) button and then receive busy-to-idle
reminder when the facility becomes idle.
All dial “O” calls, calls to FPDCs not logged in, calls to unassigned DID numbers, and calls
to facilities in the rightmost two columns of buttons of the console that appear at both
consoles will be directed to the active console.
Note that if a personal trunk appears on only one DTAC, incoming calls on those trunks will
not receive service when the console is inactive. For this reason, it is strongly recommended
that each DTAC attendant be assigned a Call Coverage-Individual (COVER-IND) button for
the other console so that these calls can be covered. Also, be sure to make the Attendant
Console the principal station (owner) on all trunks that are to receive coverage by the other
attendant.
Systems with Switched Loop Attendant Console(s) (SLAC)
In order for the Position Busy feature to be operational, the system must have either two
attendant positions or one position plus a multiline voice terminal administered as a
“backup.” If the system has two consoles, pressing the Position Busy button on one will
make it inactive and cause most calls in the common console queue to be directed to the
active console. Each attendant covers for the other. Only one console can be in Position
Busy condition at a time. If the system has one console with an administered backup voice
terminal, pressing the POS BUSY button will make the SLAC inactive, and most calls from
the common queue will be directed to the backup terminal.
2-37
A console in Position Busy mode can receive attendant PDC, DID, and DGC calls, and
outgoing calls can still be placed. Local functions can be activated.
Considerations
Position Busy allows one of two attendant positions to be made inactive when not required.
This is useful in situations where calling traffic requires only one console operator.
Interactions
●
Attendant Call Extending: Unanswered calls extended by an inactive console will
return to the active console on the Return-On-Don’t-Answer (RTN-DA) button
(DTAC only) or on a Loop button (SLAC only).
●
Attendant Camp-On: Calls Camped-On by an inactive console will return to the
active console when Camp-On timeout occurs.
●
Attendant Message Waiting: An inactive attendant is permitted to control voice
terminal Message LEDs.
●
Automatic Intercom: The inactive attendant is permitted to place Automatic
Intercom calls. DTAC only: Automatic Intercom calls to the inactive attendant will
not ring at the console or be transferred to the active attendant when the AUTO
ICOM button is located in one of the two rightmost button columns
●
Call Coverage:
DTAC only: If the active attendant is a coverage receiver for the inactive attendant,
coverage is invoked and calls will appear at the active attendant’s Cover button. If
the inactive attendant is a coverage receiver for the active attendant, coverage, when
activated, is invoked at all coverage stations including the inactive attendant.
However, if the Cover button is located in one of the two rightmost button columns,
coverage calls will not ring at these buttons.
SLAC only: All Calls covered by the common queue will be directed to the active
console.
●
Call Park: A call parked on the inactive attendant console will return to the
inactive console if the call times out; calls parked via the Selector Console will return
to the active console.
●
Direct Group Calling: If the inactive attendant is a member of a DGC Group, calls
directed to the group will be routed to the inactive attendant. The attendant must
dial *4 (activate DGC Group “Make Busy”) to busy out from the group. Dialing *6
deactivates the “Make Busy” function.
●
Direct Inward Dialing: All DID calls to unassigned DID numbers will be
transferred to the active attendant.
●
Night Service: An inactive attendant that is a Directed Night Service receiver will
receive Night Service calls.
●
Personal Dial Codes: If a PDC/FPDC is logged in at an inactive attendant
console, then calls to this PDC/FPDC will go to the active attendant (SLAC) or to
the inactive attendant (DTAC only). All calls to floating PDCS not logged in will be
transferred to the active attendant.
2-38
. Personal
Lines (Trunk Appearances) on DTACs : All calls to trunks having an
appearance in either of the two leftmost button columns will ring normally at the
inactive console. All calls to trunks having appearances in either of the two
rightmost button columns will not ring. If these trunks also have an appearance at
the active console, they will ring there even if they do not normally.
. Program: The Program feature remains active at the inactive console.
. Programmable Buttons: All DSS, Flex DSS, and REP DIAL buttons
remain
active on the inactive console.
Administration Requirements
Voice Terminal: (Station Port)
●
●
Assign Position Busy button on DTAC. A Position Busy button is defaulted to the
SLAC; it can be assigned to another feature if desired.
Assign COVER-IND buttons between consoles (DTAC only).
Hardware Requirements
None
2-39
ATTENDANT RELEASE
Description
Releases the attendant from an extended call. There are two forms of Attendant Release:
Manual and Automatic. This feature applies to the Direct Trunk Attendant Console (DTAC)
and the Switched Loop Attendant Console (SLAC).
Manual Release:
Pressing RELEASE releases the attendant from an extended call and completes the
associated call transfer. The status LED of the original calling facility will change from
hold to busy for direct trunk terminations and from hold to idle for other call facilities (e.g.,
Loop, Return- On-Busy, Return-On-Don’t-Answer, Cover, Automatic Intercom, DSS, and
System Access).
Calls cannot be released to Reorder or Dial Tone.
Pressing CANCEL terminates the destination call and reconnects the attendant to the
calling party. If the attendant goes on-hook without first releasing a call, the call extending
operation will be terminated (the calling party remains on hold). In this case, the attendant
can go off-hook and press the held call appearance button to reconnect to the calling party.
Automatic Release:
This feature simplifies the attendant procedures by eliminating the need for the attendant to
press RELEASE when releasing from one call to handle another. Selection of any new line
facility while active on the Start button will automatically release the first call. At release,
the status LED of the first calling facility will change from hold to busy for direct trunk
terminations and from hold to idle for other call facilities (e.g., Loop, Return-On-Busy,
Return-On-Don’t-Answer, Cover, Automatic Intercom, DSS, and System Access).
The Automatic Hold feature can be administered for the SLAC as an alternative to
Automatic Release. If the attendant, active on a loop call, presses another loop button to
place a call or pick up a held call, the active call is put on hold—not released.
Considerations
Attendant Manual Release improves attendant efficiency in handling calls by allowing the
attendant to release an extended call without having to wait for the called station to answer.
Attendant Automatic Release enhances the attendant’s ability to handle many calls by
eliminating the Release operation when answering a second call.
The Release function is inhibited whenever the Start facility is connected to Reorder or Dial
Tone. Pressing CANCEL will terminate the destination call and reconnect the attendant to
the calling party.
Administering the Automatic Hold option instead of Automatic Release reduces the
occurrence of accidentally dropped calls.
2-40
Interactions
●
Attendant Camp-On: External calls that are released when Busy Tone is heard
will be camped on.
Administration Requirements
None.
Hardware Requirements
None.
2-41
ATTENDANT RETURN COVERAGE ON BUSY
Description
Allows a camped-on call at a busy station or DGC Group to be returned to the attendant for
service after a specified time period.
A camped-on call not answered within 1 to 120 seconds (administrable) after the attendant
releases the call will return to the console in one of the following ways:
●
On the Return-On-Busy (RTN-BUSY) button at a Direct Trunk Attendant Console
(DTAC).
●
On a LOOP button at a Switched Loop Attendant Console (SLAC; V2).
To answer a returned call at a DTAC, the attendant presses RTN-BUSY (if not selected by
Ringing Line Preference. ) A returned call can be reextended via the START button or a
Selector Console button. In either case, the Return-On-Busy button is idled as soon as the
attendant releases.
To answer a returned call at a SLAC, the attendant merely lifts the handset to be connected
to the ringing loop.
When the RTN-BUSY button is busy at a DTAC, the calling party will remain on-hold; if a
loop is not available at a SLAC, the returning call remains in the console queue. The system
will continue to attempt to ring the called station until the RTN-BUSY button is idle or a
loop is open. When Attendant Camp-On is not provided (Camp-On return time set to zero
seconds), calls released by the attendant to busy tone are returned to the console
immediately.
Considerations
Attendant Return-Coverage-On-Busy allows the attendant to service calls not answered
within specified time intervals. This provides the calling party better service and results in
fewer lost calls.
Interactions
●
Attendant Camp-On: External calls that are released when Busy Tone is heard
will be camped on.
●
Attendant Console, Direct Trunk: As long as an Attendant Console remains
active, the call will return to the attendant who transferred it.
●
Attendant Console, Switched Loop (V2): A returning call is directed from the
console queue to a LOOP button on any available console that is administered to
receive it.
●
Attendant Direct Extension Selection: If a call to a Floating PDC (FPDC) is
returned to the attendant, the FPDCs status LED on the Selector Console will flash
during ringing and go dark when the call is answered.
●
Direct Group Calling: External calls that are camped onto a DGC group that does
not have a delay announcement will return to the attendant console after the
specified number of rings.
2-42
.
Send All Calls (DTAC only): If Send Al Calls is activated, returning calls will
ring at the DTAC.
Administration Requirements
System:
.
Assign number of seconds before unanswered camped-on calls return to
Attendant Position (1-120 seconds, or O for No Camp-On) - Default = 30 seconds.
Hardware Requirements
None
2-43
the
ATTENDANT RETURN COVERAGE ON DON’T ANSWER
Description
Allows unanswered calls extended by the attendant to be returned to the attendant for
additional service.
Calls that are not answered after a specified number of rings will transfer ringing to the
Return-On-Don’t-Answer (RTN-DA) button on a Direct Trunk Attendant Console (DTAC) or
to a LOOP button on a Switched L O O P Attendant Console (SLAC). If the called voice
terminal has call coverage, the timing for return begins only after the coverage station
begins ringing.
When the RTN-DA button is busy, calls will continue to ring at the called station until the
button is idle. If a LOOP button is not available, the returning call stays in the common
queue until it can be serviced.
To answer a returned call at a DTAC, the attendant presses RTN-DA (if not selected by
Ringing Line Preference.) The call can be reextended via the START button or Selector
Console. In either case the button is RTN-DA button is idled as soon as the attendant
releases.
To answer a returned call at a SLAC, the attendant merely lifts the handset to select the
line.
Considerations
Attendant Return-Coverage-On-Don’t-Answer allows the attendant to service calls not
answered within a specified number of rings. This provides the calling party better service
and results in fewer lost calls.
Interactions
●
Attendant Console, Direct Trunk: As long as an Attendant Console remains
active, the call will return to the attendant who extended it.
●
Attendant Console, Switched Loop: A returning call is directed from the
common queue to a LOOP button on any available console that is administered to
receive it.
●
Call Coverage:
DTAC only: Whenever a DTAC attendant is a call coverage receiver for a particular
call coverage group and a call is placed from the attendant position via the Start
button or the Selector Console to a voice terminal in that group, the Call CoverageGroup (COVER-GRP) button on the Attendant Console will not track the call
(COVER-GRP button status LED will not flash). If the call remains unanswered, it
will return to the Attendant Console on the RTN-DA button rather than the
COVER-GRP button.
SLAC only: Whenever the common queue is a receiver for a particular call coverage
group and a call is placed from the attendant position via the START button or the
Selector Console to a voice terminal in that group, an unanswered call will return to
the attendant queue as a Return-On- Don’t-Answer call (instead of as a coverage
2-44
call).
●
Send All Calls (DTAC only): If Send All Calls is activated, returning calls will
ring at the DTAC.
Administration Requirements
System:
●
Assign number of rings before call return to the Attendant Position (1-31) - Default
= 5 Rings.
Hardware Requirements
None
2-45
ATTENDANT SOURCE AND DESTINATION (V2; SLAC Only)
Description
Allows the attendant, while extending a call, to switch back and forth between the calling
party (the source) and the called party (the destination) before connecting them together.
Pressing the SOURCE button on the SLAC after the called party has been reached has these
results:
●
The called party (the destination) is put on hold.
●
The attendant is reconnected to the calling party (the source).
●
●
The green status lamp of the DEST button starts winking to indicate that the
destination is on hold.
The green status lamp of the SOURCE button goes from winking to dark.
Pressing the DEST button after the source has been reconnected has these results:
●
The source is put on hold (again).
●
The attendant is reconnected to the destination.
●
●
The green status lamp of the SOURCE button starts winking to indicate that the
source is on hold.
The green status lamp of the DEST button goes from winking to dark.
Considerations
The Source/Destination feature is useful when the attendant needs to talk to each party
privately before connecting them.
Interactions
The Source/Destination feature can only be activated before the two parties are connected
together.
When the attendant presses JOIN, the other parties and the attendant are joined in a 3-way
connection.
When the attendant presses RELEASE, the other parties are connected, the call is separated
from the console, and the attendant is free to handle other calls.
After the source and destination parties are connected together, the SOURCE and DEST
status lamps go dark.
Administration Requirements
None
Hardware Requirements
None; the SOURCE and DEST buttons are standard fixed feature buttons
2-46
on the SLAC.
ATTENDANT SPLITTING ONE-WAY AUTOMATIC
Description
Allows the attendant to converse privately with a called party while the calling party is split
away on hold.
When the attendant presses START (or a DXS button) to extend an incoming call to a called
party, the calling party is automatically split away from the connection and placed on hold.
This allows the attendant to talk privately with the called party before extending the call.
The attendant can then press RELEASE to complete the transfer or CANCEL to drop the
called station and return to the incoming call.
If the console is a SLAC, the attendant can also use the Join and Source/Destination
features while in the Start mode. Refer to the descriptions of these features for details.
Considerations
Attendant Splitting One-Way Automatic allows the attendant to (1) announce a call, (2)
determine privately whether the called party is available to receive the call, and (3) obtain
information if necessary to redirect the call or take a message.
Interactions
.
.
Attendant Source/Destination (V2; SLAC only): This feature can be used after
reaching the called party. It allows the attendant to speak privately to both the
calling party and the called party before connecting them together.
Music-On-Hold: Music-on-hold is not provided to the calling party while the call is
split from the console.
Administration Requirements
None
Hardware Requirements
None
2-47
ATTENDANT SYSTEM ALARM INDICATION
Description
Provides an Alarm on the Attendant Console to alert the attendant to problems detected by
the system software. The ALARM LED on the Attendant Console will flash whenever a
detected fault persists longer than four minutes, or if more than five transient faults per
hour are detected. The alarm indication should be reportd immediately to your AT&T
Systems Technician.
The alarm type that causes an alarm indication is referred to as a Permanent System
Alarm. These alarms are faults that can cause degradation of service and require immediate
attention.
If a flashing ALARM button is pressed, the LED will change from flashing to steadily lit. A
new trouble situation will cause a steady ALARM LED to start flashing again. Only when
the trouble has been corrected will the LED turn off.
Considerations
The ALARM LED on the Attendant Console provides a warning as soon as the fault is
detected. This permits a quick response to system detected faults.
In a two-attendant system, both consoles track problems.
Interactions
None
Administration Requirements
None
Hardware Requirements
None
2-48
AUTOMATIC INTERCOM
Description
Allows a multiline voice terminal user (or attendant) to place and answer calls to and from
another station by use of a dedicated button appearance.
Automatic Intercom provides a private path between two designated multiline voice
terminals. To place an Automatic Intercom call, the calling party presses the Automatic
Intercom (AUTO ICOM) button and goes off-hook. The calling party hears ringback tone
and the called party receives standard ringing. The status LED associated with the button
is steadily lit at the calling voice terminal and flashing at the called voice terminal. To
answer an Automatic Intercom call, the called party presses AUTO ICOM (not necessary
with Ringing Line Preference ) and goes off-hook.
The AUTO ICOM status LED lights steadily whenever the other party is off-hook. This
provides each party with a station-busy indication for the other. To activate the busy-to-idle
reminder, the user can press AUTO ICOM (remaining on-hook). A short burst of tone is
provided when the other party goes on-hook. The user can then go off-hook, and the call will
be placed; the user does not press the AUTO ICOM button again.
Pressing AUTO ICOM to invoke the busy-to-idle reminder overrides Prime Line Preference.
Once activated, the feature can only be canceled by preelection of another button or
answering an incoming call. See the “Busy-to-Idle Reminder” feature description for more
information.
At a Switched Loop Attendant Console, operation of an AUTO ICOM button seizes an idle
loop button for the outgoing call. An incoming intercom call arrives on a loop button and
does not flash at the AUTO ICOM button of the console.
Considerations
The intercom feature should not be confused with ordinary station-to-station calling inside
the system using dialed PDCs. With Automatic Intercom, users who frequently call each
other can do so by pressing one button instead of dialing a PDC. In addition, the stationbusy indication and busy-to-idle reminder provide additional utility to users.
This feature is similar to Direct Station Selection (DSS), except that the buttons must
always be assigned in pairs (i. e., between two sets.) Hence, an AUTO ICOM button cannot
point to a single-line set. Also, Automatic Intercom calls arrive at the AUTO ICOM button,
thereby providing calling party ID; DSS calls arrive on System Access buttons.
Interactions
●
Attendant Position Busy: The inactive attendant is permitted to place Automatic
Intercom calls. For DTAC only: Automatic Intercom calls to the inactive attendant
where the AUTO ICOM button is located in one of the two rightmost button columns
will not ring at the console, nor can they be covered by the active attendant. For
SLAC only: Automatic Intercom calls to the inactive attendant will ring at the
inactive SLAC.
●
Call Coverage: Automatic Intercom calls are considered private and do not receive
call coverage.
2-49
●
Call Pickup: When an Automatic Intercom call is picked up via Call Pickup, the
AUTO ICOM status LED on the called voice terminal lights steadily. The called
party can press AUTO ICOM to enter the call at any time.
●
Direct Group Calling: Automatic Intercom calls cannot be directed to DGC groups.
●
Exclusion: Any attempt to engage Exclusion while active on an Automatic Intercom
call will drop the other party.
●
Last Number Dialed (V2): Numbers called using an AUTO ICOM button are not
saved by the Last Number Dialed feature.
●
Line Selection (Prime Line Preference): When the Automatic Intercom line is
assigned Prime Line status, the AUTO ICOM button must be pressed to activate the
busy-to-idle reminder even though the I-use LED is already lighted steadily.
Administration Requirements
Voice Terminal: (Station Port)
●
Assign AUTO ICOM buttons to voice terminals in pairs. Voice terminals can have
several AUTO ICOM buttons assigned for direct access to multiple stations.
Hardware Requirements
None
2-50
AUTOMATIC ROUTE SELECTION (ARS)
Description
Provides for the routing of calls over the telecommunications network based on preferred
routes (normally the least expensive route available at the time the call is placed.)
An ARS pattern can be composed of two subpatterns (time of day determines which
subpattern is selected), each consisting of up to three routes, associated Facility Restriction
Level (FRL) codes (described below), and CO overflow flags. A route is identified by
specifying a Facility Access Code for the pooled facility (trunk group); for V2 systems only,
a route may also be identified by specifying a Virtual Facility code.
A trunk group (or virtual facility, for V2) can be used in more than one ARS pattern and
more than once within a pattern.
Call routing can be specified by as many as eight routing patterns. Each pattern contains a
sequential list of routes (for example, trunk groups) the system can use to complete a call.
Number translations (deletion and addition of dialed digits) necessary to route the call is
determined on a trunk group basis. Overflow to the local CO when all trunks in a pattern
are busy or the route FRL is too high is optional. If all trunks in a pattern are busy
(including CO trunks if overflow is allowed), the call queues on the first route in the pattern.
All calls placed using the ARS access code (default = 9) are routed via the feature. The
dialed numbers that follow the ARS access code are generally seven or ten digit DDD
numbers preceded by a “l” if required by the serving Central Office. Numbers preceded by a
“0” are routed over the local CO pooled facility.
Typically, a dialed 7-digit number consists of a CO code and exchange number in the form
NXX-YYYY where N = 2-9, X = 0-9, and Y = 0-9. A 10-digit number consists of an area
code, CO code, and exchange number in the form NPA-NXX-YYYY where N = 2-9, P = 0-1,
A = 1-9, X = 0-9, and Y = 0-9.
Each route in a pattern has an associated FRL (0-3). This FRL may differ each time the
facility is specified as a route. A facility with a FRL of “0” is least restricted to callers; a
FRL of “3” is the most restricted. Similarly, each station in the system is assigned an FRL
(0-3). A terminal assigned an FRL of “0” has the least ARS privileges (i.e., routes with
FRLs of 1-3 are restricted); a FRL of “3” provides the most privileges. A station’s FRL
must be equal to or greater than the routes FRL to use the route.
The ARS feature, when accessed, selects a pattern as follows:
●
Emergency Number Calls (routed via the local CO facility)
●
International Calls (routed via the administered international pattern)
●
Calls made to specified COs or seven digit telephone numbers within the Home
Number Plan Area (HNPA). These calls are routed as specified in the HNPA
Exception Lists, or else via the NPA Routing Table or (by default if not otherwise
specified) the local CO facility.
●
Calls made to NPAs outside the HNPA, sometimes referred to as Foreign NPAs
(FNPAs). The route selected depends on the type of call, as follows:
— FNPA special number calls (includes all “800”, “900”, and Telex 510, 610, 710,
and 810 numbers). For V2 systems, each FNPA of the form N00 and N10 may
be assigned to a routing pattern. For V1 systems, these calls are routed via
the local CO facility.
2-51
— FNPA calls made to numbers specified in the FNPA Exception List.
— All other FNPA calls.
ARS Flow Chart
Figure 2-8 provides a simplified ARS flow chart. Bracketed numbers (e.g., [401], [601])
provide a link between ARS administrable action numbers and the associated item on the
flow chart. Certain readers may find this reference useful when reading the following
description in association with the System 25 Administration Manual. Administrable System,
Station, Toll Allowed, and Trunk action numbers are also noted where applicable.
The ARS feature is accessed when a user dials the ARS access code. As shown on Figure 28, the number dialed is first checked against the Emergency Numbers List. This list consists
of special service codes (911) and up to three customer-defined seven digit numbers. If the
number dialed matches one of the numbers on the list, the call is immediately routed via the
local CO facility. All user call restrictions are disregarded.
If the number dialed is not on this list, a check is made to determine if the terminal is
allowed to originate outside calls. If the terminal is outward restricted, the caller receives
Reorder Tone; otherwise, the dialed number is checked against any toll restrictions that
apply.
Terminals may be assigned a Toll Restriction Class (1-4), or be unrestricted (Class 0).
Terminals assigned Toll Restriction Class 1 have the most privileges, those assigned Class 4
have the least privileges. There are four associated Toll Call Allowed Lists (l-4) in the
system. Up to 64 3-digit CO codes and 6-digit NPA plus CO codes may be divided among the
four lists.
Numbers dialed from voice terminals assigned Toll Restriction Class 1 are checked against
all four Toll Call Allowed (TCA) Lists; numbers dialed from Class 2 terminals are checked
against TCA Lists 2-4; numbers dialed from Class 3 terminals are checked against TCA Lists
3-4; and numbers dialed from Class 4 terminals are checked against List 4 only. If the
number dialed does not appear on a list, the user receives Reorder Tone. Calls originated at
unrestricted (Class 0) terminals are not screened.
Calls are checked to determine if they are international calls or operator calls. Dialed
numbers “01” or “011” signify international calls, “0” plus a number other than “l” signify
operator calls. If the call is an international call, the international routing pattern is
selected and the call routed accordingly. Operator calls are routed via the local CO facility.
Calls within the HNPA are checked against the HNPA Exception Lists. There may be up to
four of these lists, each with an associated ARS Routing Pattern. Up to 800 (64 in R1V1) 3digit office codes may be divided among the four lists (eight entries may be 7-digit numbers.)
If a match is found, the call is routed via the associated ARS Routing Pattern. If no match
is found the dialed number is routed via the HNPA pattern (specified in the NPA Routing
Table ), or if none is specified, via the local CO facility.
If a number is entered more than once in the exception list, the pattern used will be the
pattern associated with the more specific number.
The NPA Routing Table is simply a listing of North American Plan NPAs and Special
Number NPAs (V2), each having an associated ARS Routing Pattern (all North American
NPAs are assigned routing pattern 1 by default). A dialed NPA that is listed in the table is
routed using the associated Pattern. Calls to NPAs not listed are routed via the local CO
facility.
2-52
V1 Systems Only: For calls outside the HNPA (FNPA calls), a check is made against a
non-administrable Special Numbers List. The numbers in the list include all “800” and
“900” numbers and Telex codes 510, 610, 710, and 810. If a match is found, the call is routed
via the local CO facility.
For FNPA calls not on the Special Numbers List (Vi), the dialed numbers are checked
against the FNPA Exception List (Vl and V2). Up to 32 entries may be assigned to the list.
Each entry must consist of a 3-digit NPA code, 3-digit CO code, and two additional digits
(for a total of 8 digits). The last two digits may be “.”, which match any digit. Each entry
has an associated ARS Routing Pattern. If a match is found, the call is routed using this
pattern. If no match is found, the call is then checked against the NPA Routing Table. A
dialed NPA that is listed in the table is routed using the associated Pattern. Numbers that
don’t match are routed via the local CO facility.
ARS Routing Pattern Table
Figure 2-9 provides a block diagram of an ARS Routing Pattern. Up to eight of these
patterns may be administered in the system. Each pattern consists of two subpatterns that
may be chosen based on the time of day. Each subpattern (A and B) can contain up to three
allowed routes. If all routes in a subpattern are busy, a CO overflow flag (when set) allows
the call to be routed via the local CO facility; otherwise, the call will queue on the first route
in the subpattern.
Administrable Start and Stop times (Hour and Minute) for Routing Subpattern A specify
when Subpattern A should be used to route calls. Subpattern B is used to route calls at all
other times.
Each route is specified by its trunk group facility access code (or Virtual Facility code in V2)
and an associated FRL.
An FRL is typically lower for the first route in a subpattern and increases with each
additional route in the pattern. A terminal’s FRL must be equal to or greater than the route
FRL for the route to be selected. The system first checks the Route #1 for an available
trunk on which to route the call. If the route is busy, Route #2 is checked, then Route #3, if
required. If all routes in the subpattern are busy and the CO overflow flag is set, the voice
terminal FRL is checked against an associated Overflow FRL before routing the call. For V1
only: If the voice terminal is allowed access to the local CO trunk pool, the system, on
overflow, will attempt to place the call regardless of the associated FRLs.
If all routes in a subpattern are busy and the CO Overflow flag is not set, or all CO trunks
are busy, the call returns to the first route in the subpattern and is queued (if the station
FRL permits access to the first route). The caller is placed in an off-hook queue indicated by
five short bursts of tone (Preferred Routing Tone). The order of service in the queue is First
In First Out (FIFO). An ARS user placed in an off-hook queued state always has precedence
over users with a Direct Facility Access buttons. There are 16 slots available in the ARS
queue. A route #1 must be specified in the subpattern for a call to queue. If it is not, the
caller receives Reorder Tone instead of being queued.
Once a route has been selected, the entries in a Digit Translations Table associated with the
selected route’s trunk group (or Virtual Facility in V2) is checked. Based on an associated
NPA and the NPA dialed, the system can remove up to 10 digits and then add a pattern of
up to 5 digits as specified to route the call.
2-53
STATION USER
DIALS ARS
ACCESS CODE
[401]
YES
CALL ROUTED
VIA LOCAL
CO FACILITY
NO
NO
I
[601-603]
VOICE TERMINAL
ASSIGNED TOLL
RESTRICTION
CLASS 1-4
VOICE TERMINALS
“RESTRICT
OUTWARD
CALLS” = YES
NUMBER DIALED
ON EMERGENCY
NUMBERS LIST
(911, AND
THREE CUSTOMER
DEFINED 7-DIGIT
NUMBERS
NUMBER DIALED
INTERNATIONAL,
OPERATOR, OR
OTHER
NO
OTHER
(SHEET 2)
STATIONS
[14]
STATIONS
[13]
YES
CALLING USER
RECEIVES SYSTEM
REORDER TONE
YES
OPERATOR
NUMBER DIALED ON TOLL CALL
ALLOWED LIST? 64 CODES MAY
BE DIVIDED INTO 4 LISTS OF YES
ANY LENGTH
TOLL ALLOWED
[1-3]
INTERNATIONAL
NO
STATION USER
RECEIVES SYSTEM
REORDER TONE
ROUTE CALL
VIA LOCAL
CO FACILITY
ROUTE CALL
VIA
INTERNATIONAL
ROUTING
PATTERN
[ 402]
Figure 2-8. Automatic Route Selection Flow Chart (Sheet l of 2)
2-54
FROM
SHEET 1
NUMBER DIALED
WITHIN HOME
NPA?
V1 ONLY: NUMBER
DIALED ON SPECIAL
NUMBERS LIST
(ALL 800,900,
AND TELEX 510,
610, 710, AND
810 CODES)
NUMBER DIALED
ON FNPA
EXCEPTION
YES
LIST
NO
CALL ROUTED
VIA ASSOCIATED
ROUTING PATTERN
NO
SYSTEM [30]
YES
YES
CALL ROUTED
VIA LOCAL
CO FACILITY
NUMBER DIALED
ON HOME NPA
NO
EXCEPTION
LISTS
[500]
HOME AREA CODE
ON NPA ROUTING
TABLE [300]
AREA CODE ON
NO
NPA ROUTING
TABLE [300]
YES
YES
CALL ROUTED
VIA ASSOCIATED
ROUTING PATTERN
CALL ROUTED
VIA ASSOCIATED
ROUTING PATTERN
NO
YES
CALL ROUTED
CALL ROUTED
VIA
ASSOCIATED
ROUTING
PATTERN
VIA LOCAL
CO FACILITY
Figure 2-8. Automatic Route Selection Flow Chart (Sheet 2 0f 2)
2-55
The following tones are associated with ARS:
●
Confirmation–Indicates that a queued call is being serviced (trunk available to route
call )
●
Busy—Indicates that the called number is busy
●
Reorder–Indicates that all trunks are busy, the ARS queue is full, or that ARS
calling is denied.
●
Preferred Route Tone—Five very short tones that indicate that your call has been
queued for the preferred route.
Considerations
With ARS, users do not have tc worry about accessing a particular pooled facility to make a
long distance call. The user simply dials the ARS access code and the desired number. The
system then routes the call via the facility best suited for that call.
The following provides a summary of the ARS call routing controls provided by the feature:
●
Emergency Numbers L i s t : 9 1 1 , a n d u p t o t h r e e c u s t o m e r - d e f i n e d , 7 - d i g i t
numbers.
●
Toll Call Allowed Lists: 1-4 lists, 64 entries maximum between all lists. Entries
may be 3-digit CO codes or 6-digit NPA plus CO codes. (Administrable for uses
besides ARS)
●
Station Toll Restriction Class: l-4 Classes (Administrable for users besides ARS)
●
HNPA Exception List: 1-4 Lists, each with an associated ARS Routing Pattern.
800 (64 in Vl) 3-digit CO codes entries maximum between all lists (eight of the
entries may be 7-digit numbers.)
●
NPA Routing Table: Entries may include every North American NPA (and Special
Number NPAs in V2), each with an associated ARS Routing Pattern. All NPAs are
assigned Routing Pattern #1 by default.
●
FNPA Special Numbers List (VI only): Routes all “800”, “900”, and Telex 510,
610, 710, and 810 numbers via the local CO trunk group (not administerable)
●
FNPA Exception List: One List with up to 32 eight-digit numbers. Each entry has
an associated ARS Routing Pattern.
●
Digit Translations Tables: One per trunk group (Vl and V2) or Virtual Facility
(V2 only).
A system can have up to eight ARS Routing Patterns assigned. Each pattern can contain up
to six routes (three per subpattern).
Interactions
●
Calling R e s t r i c t i o n s : Outward Restriction and To1l Restriction, when
administered, prevent calls from routing via ARS. Pooled facility access restrictions
do not apply. In fact, the recommended arrangement to ensure that users make
maximum use of ARS is to block dial access to most trunk groups, so that users must
dial the ARS access code to place calls.
2-56
●
Direct Facility Access Button: Multiline voice terminal users who have pressed
FACILITY to activate the busy-to-idle reminder must wait until all queued ARS
users have been serviced.
●
Virtual Facilities (V2 only): Virtual Facilities may be used in place of trunk
groups in ARS routing patterns. When used with ARS, a digit translation scheme
may be associated with each virtual facility. See the “Virtual Facilities” feature
description for more information.
Administration Requirements
System:
●
Specify your area code (HNPA )
●
ARS Access Code (1-9999) - Default = 9
ARS
●
International ARS Routing Pattern Number (1-8) - No default
●
Three Emergency Numbers Lists (7-digit numbers) - No default
●
NPA Routing Table (NPA code and associated pattern number 1-8) - Default = 1
●
HNPA Exception List:
— List Number (l-4)
— Pattern Number (1-8)
— Exception Numbers (NXX or NXX-YYYY). Last three digits may be “.”
●
FNPA Exception Telephone List
— Pattern Number (1-8)
— Telephone Number (8-digits in the form NPA-NXX-YY). Last two digits may
be “. ”
●
Digit
Translations
Table
— Trunk Group Facility Access Codes (1-9999) or, in V2, Virtual Facility Codes
(#190-#199)
— Associated NPA (NPA)
— Number of digits to remove for calls within associated NPA (1-10 digits,
none)
— Digit pattern to add for calls within associated NPA (maximum of 5 digits) Default = none
— Number of digits to remove for calls not in associated NPA (1-10, none) Default = O
— Digit pattern to add for calls not in associated NPA (maximum of 5 digits) Default = none
●
ARS
Routing
Pattern:
Pattern Number (l-8)
2-57
— Subpattern A Start and End Time (Hour: Minute)
— Subpattern A and B–Route 1, 2, and 3 Facility Access Codes (1-9999) or, in
V2, Virtual Facility Codes (#190-#199)
Subpattern A and B—Route 1, 2, and 3 FRLs (0-3)
— Subpattern A and B Overflows to CO facility (Yes, No)
Subpattern A and B Overflow FRL (0-3).
Terminal (Station Port)
●
ARS FRL Level (0-3).
Hardware Requirements
None
2-58
ROUTING SUBPATTERNS (RSP) A AND B
ROUTING
PATTERNS
(1-8)
RSP A ROUTE 1
ROUTE 2
ROUTE 3
POOLED
FACILITY
ACCESS
CODE
POOLED
FACILITY
ACCESS
CODE
POOLED
FACILITY
ACCESS
CODE
CO
OVERFLOW
FLAG
[110]
[120]
[130]
[140]
FRL (0-3)
[111]
FRL (0-3)
[121]
FRL (0-3)
[131]
FRL (0-3)
[141]
START/
STOP
TIME
[101],
[102]
[100]
RSP B
[ 200]
ROUTE 1
ROUTE 2
ROUTE 3
POOLED
FACILITY
ACCESS
CODE
POOLED
FACILITY
ACCESS
CODE
POOLED
FACILITY
ACCESS
CODE
CO
OVERFLOW
FLAG
[210]
[ 220]
[230]
[ 240]
FRL (0-3)
[211]
FRL (0-3)
[221]
FRL (0-3)
[221]
FRL (0-3)
[241]
ROUTE
SELECTED
ALL ROUTES
BUSY
NO
YES
ROUTE CALL
VIA DDD
NETWORK
DIGIT TRANSLATIONS
TABLE ASSOCIATED
WITH POOLED FACILITY
(OR VIRTUAL FACILITY, V2
ONLY) DELETE DIGITS
(0-10) ADD DIGITS
(MAX 5) [700S)
Figure 2-9. Automatic Route Selection Routing Pattern
2-59
QUEUE
ON
ROUTE 1
BUSY-TO-IDLE REMINDER
Description
Enables a multiline voice terminal user to be alerted by a single ring as soon as a busy
internal station, DGC group, or facility (trunk group) becomes available. The feature can be
activated only for stations, DGC groups, and trunk groups represented on the terminal by
DSS, FLEX DSS, AUTO ICOM, and FACILITY buttons.
Before making a call to a station, the multiline voice terminal user can check the green
status LED of the station button. If it is lit, the station party is off-hook. To be alerted
when the party hangs up and is available again, the user (while on-hook), presses the button
of the station. The red I-use LED lights, indicating that Busy-to-Idle Reminder is in effect.
When the other party hangs up, the user’s terminal rings once. The user simply goes offhook, and the station is called; the user does not have to press the button again.
If the user calls a station by pressing a FLEX DSS, DSS, or AUTO ICOM button and
receives busy tone, the user must hang up before activating Busy-to-Idle Reminder.
When all the trunks in a pool represented by a FACILITY button are busy, the green status
LED is lighted. The user can activate Busy-to-Idle Reminder in the same way as for a
station call, by pressing the FACILITY button while on-hook. When a trunk becomes idle,
the terminal rings once. The user goes off-hook and is automatically connected to the trunk.
To complete the call the user dials the desired outside number.
Considerations
Busy-to-Idle Reminder gives the multiline voice terminal user a way to get quick access to a
station or trunk group that has just become available after being busy. Access to the station
or trunk is not reserved for the user who activates this feature; any other user has equal
access to the idle facilities.
On some multiline voice terminals, FLEX DSS, DSS, AUTO ICOM, and FACILITY buttons
do not have status and I-use LEDs. This makes Busy-to-Idle Reminder less convenient to
use because the user must first call the facility to determine if it is available. If it is not,
the user activates the feature by hanging up and pressing the button again. When the
reminder ring sounds, the user must then remember which button was used to initiate the
call.
Interactions
●
Busy-to-Idle Reminder cannot be used with the Last Number Dialed, Personal Line,
or Repertory Dialing features.
Administration Requirements
None
Hardware Requirements
None
2-60
CALL ACCOUNTABILITY
Description
Allows system users to charge outside calls made from other users’ stations to their own
PDCs.
To charge a call to his or her own PDC, the user dials ## followed by the PDC immediately
upon receiving first dial tone to place a call. When second dial tone is returned, the user
dials the desired number in the normal way.
After completion of the call, the SMDR record will reflect the “accountable” PDC (that is,
the caller) in the “PDC’’ field, and the PDC of the voice terminal used in the “STN’’ field.
Considerations
Call Accountability, if used consistently, helps to ensure that calling costs are attributed
accurately to the personnel who incur the costs. Users do not use this feature when calling
from their own stations or when making inside calls from any station.
Interactions
All of the following conditions apply only when a user is calling from another user's station.
.
.
.
.
.
A c c o u n t C o d e E n t r y : This feature can be used on the same call with Call
Accountability.
Call Accounting: The caller’s PDC that is entered by the Call Accountability
procedure is integrated into the reports generated by Call Accounting systems.
Direct Facility Access: When a call is made using a Facility button, ##PDC must
be dialed before pressing the button.
Repertory Dialing: When a call is made using a Rep Dial button, ##PDC must be
dialed before pressing the button.
Speed Dialing: When a call is made using Speed Dialing, ##PDC must be dialed
before dialing the Speed Dialing code.
Administration Requirements
None
Hardware Requirements
None
2-61
CALL ACCOUNTING
Call Accounting is the collecting, processing, and use of information about all trunk calls
placed from and received by System 25. It is intended to help customers control telephone
use and manage associated costs.
Detailed call data is available at Interface Port 3 of the Digital Switch’s Call Processor
Circuit Pack. This data can be fed to one of the following peripheral equipment systems:
. Station
Message Detail Recording (SMDR) equipment for printing a standard call
report.
●
A Call Accounting System (CAS) for preparing a variety of cost estimate reports and
for providing management and directory type services.
Two station features of System 25 are also related to Call Accounting and are covered in
separate subsections of this manual. Account Code Entry allows individual voice terminal
users to associate specific account codes with their calls, when necessary. Call
Accountability provides users with the means to properly identify calls they make from
stations other than their own. The information gathered from these two features is part of
the data output from the processor to the SMDR or CAS.
The remainder of this subsection is dedicated to coverage of SMDR and CAS.
Station Message Detail Recording (SMDR)
Description
SMDR records detailed call information on all incoming and outgoing (external) calls and
sends this information to an (optional) output device. Data on inside calls is not collected.
The call records can be used to compute costs, allocate charges, and analyze calling patterns.
The output device can be any serial RS-232 compatible DTE device capable of receiving the
data (must supply DTR on pin 20) and either printing the call records or storing and
analyzing them. (80 character ASCII records are sent to the output device.)
The SMDR RS-232 port interface is provided by a DUART driver (68681). It is a one-way
port transmitting data to the output device. No characters are read by the port interface,
and no flow control mechanisms are provided. The standard data transmit rate is 1200 bps.
(Also operates at 300 bps.)
Call Records
The call records provide detailed information concerning both incoming and outgoing calls.
Call detail records are generated during call processing and are sent to the SMDR output
device in ASCII format. SMDR records are provided for:
. Voice Records: The system prints call records for incoming calls and for outgoing
calls that exceed a specified duration. For special types of calls such as conference
or transferred calls, one call record is reported for each trunk seized, regardless of
the number of parties connected to the call. The call’s duration is from the time the
last digit was dialed until the last person hangs up. No indication is provided that
trunks have been bridged together.
2-62
. Data
Records: The system prints call records for incoming and outgoing (external)
data calls. Calls are considered data calls if they involve a data extension.
The following list describes the SMDR data collected for each call and the number of
characters in each field. All information is right justified in its field, unless otherwise
indicated. The record is provided in a standard 80-column format. The headings for each
record item are noted in bold type. These headings are printed across the top of each page.
Page advance is determined by counting lines based on a fixed page length. Each record is
followed by a carriage return and a line feed.
The system can provide for the storage of up to 100 SMDR records. If more than 100 records
are received while the printer is disconnected, a message “Calls Lost Due To Call Record
Overflow” is provided when a printer is re-attached.
The SMDR call detail (Figure 2-10) contains the following information for each call record:
.
.
.
TYPE (Column 1)
All voice calls are labeled C, data calls are labeled D. (“TYPE” is not printed as a
column heading)
Blank (Column 2)
DATE (Columns 3-10)
The date the call is originated.
. Blank (Column 11)
. TIME (Columns 12-16)
The time the trunk is seized is listed using a 24-hour clock. For example, 2:01 PM is
listed as 14:01. Seconds are truncated.
. Blank (Column 17)
. CALLED NUMBER (Columns 18-35)
For outgoing calls, up to 15 digits may be recorded, excluding the ARS or facility
access code but including the O or 1 prefix (to identify local and toll calls) and 9501 0XX and 10 XXX interconnect access codes. Space is allotted for three dashes, one
between the fourth and fifth digits from the right, one between the seventh and eight
digits from the right, and the other between the tenth and eleventh digits from the
right. Numbers longer than 15 digits will be truncated. For Repertory Dial and
Speed Dial numbers, the facility will be extracted from the stored number and
reported under the FAC heading; the number remaining after the facility is extracted
will be reported as the called number.
.
.
.
An incoming call is identified by the word IN.
Error Character - Question Mark or Blank (Column 36). Indicates number dialed
exceeded 15 digits.
Blank (Column 37)
DUR (Columns 38-45- Duration)
For incoming calls, this provides the time between trunk seizure and disconnect,
rounded to the nearest second. For outgoing calls, it provides the time between the
last digit dialed until the last station on the call hangs up, less an estimated time for
call setup (15 seconds), rounded to the nearest second. A call transferred between a
2-63
.
.
number of voice terminals will reflect the total call duration. The maximum time
that can be reported is 95 hours, 59 minutes, and 59 seconds.
Blank (Column 46)
FAC (Columns 47-51- Facility)
Indicates the facility used to place the call. For outgoing calls including speed dialed
numbers, the pooled facility selected by ARS or the facility access code that was
dialed (or that corresponds to the facility button that was pressed) is identified. For
incoming calls and Personal Line calls, the trunk number is identified.
For V2 only: If a virtual facility was used to complete the call, the applicable Virtual
Facility Code (#190-#199) is identified in this field.
. Blank (Column 52)
. STN (Column 53-56)
Identifies the voice or data terminal responsible for the call. If an account code is
entered, the voice terminal where the code is entered is reported. If no account code
is entered, the terminal originating an outgoing call is identified, or the last terminal
connected to an incoming call is identified.
For an incoming call to a DGC group that is connected to an announcement but is
never answered, 0 will be recorded in the STN field. If the call is answered by a
station after receiving announcement, the station answering the call will be recorded.
For V2 only: If an outgoing call is originated by a tandem tie trunk, the tandem
trunk’s Facility Access Code (FAC) is recorded in this field. If no FAC exists for this
trunk, then the 4-digit trunk number (9 X X X) will appear.
.
.
.
.
For V2 only: If an outgoing call is originated by a DID trunk, 0000 is recorded in this
field.
Blank (Columns 57 ,58)
ACCOUNT (Columns 59-73)
Lists the Account Code associated with the call, if one was entered. On conference
and transferred calls, the first account code entered is recorded and subsequent
account code entries are ignored.
Blank (Column 74)
PDC (Columns 75-78)
Identifies the user responsible for outgoing calls. The user is identified by the call
accountability login (##PDC) entered at the originating voice terminal. If no call
accountability is entered, the PDC field is blank.
Figure 2-11 and Figure 2-12 summarize the Call Record and Call Record Header formats.
Considerations
SMDR provides detailed call information on incoming and outgoing calls. This information
can be used to facilitate cost allocation, traffic analysis, and detection of unauthorized calls.
2-64
Interactions
. Account Code Entry: Allows users to have an account code or project number
associated with each call record.
. Direct Group Calling (DGC): For an incoming call to a DGC group that is
connected to an announcement and never answered, “O” will be reported in the
“STN” field of the call record. If the call is answered by a station after receiving the
announcement, that station will be listed in the “STN” field.
. Modem Pooling: SMDR records do not reflect modem pool resource usage.
. Tandem Trunking (V2 only): If an outgoing call is originated by a tandem tie
trunk, the tandem trunk’s FAC is recorded in the STN field. If no FAC exists for
this trunk, then the 4-digit trunk number (9XXX) will appear. No other SMDR fields
are affected.
Administration Requirements
System:
.
.
Send SMDR records To SMDR Port (Yes, No) - Default = Yes
Minimum length (seconds) of calls that are reported by SMDR (10-255) - Default =
40.
Hardware Requirements
An AT&T Model 475 printer or any standard RS-232 serial 80-column ASCII printer is
required for printing the SMDR output. The printer must be dedicated to SMDR to ensure
that all calls are recorded. An AT&T Call Accounting System may also be used as the
SMDR output device (see below).
The printer can be directly connected to Port 2 of the Call Processor ZTN82 (V1) or ZTN128
(V2), or switched access (either on- or off-premises) can be provided. Connection is the same
as described for the SAT.
Detailed connection information is provided in Figures 2-13 through 2-17
Maximum cabling distances are provided in Section 5, “Technical Specifications.”
Call Accounting System (CAS)
Two types of CASs can be used with System 25:
CAS Model 100—A discrete microprocessor unit with
. and
a built-in power supply.
Model 200, 300, 500, or 2000 Software Package
. CAS
Personal Computer (PC) 6300.
2-65
cartridge-packaged
software
associated with an AT&T
Description
CAS Model 100:
The CAS 100, when connected to a serial printer and the System 25 SMDR port, calculates
the cost of calls made to outside numbers, stores the cost information, and generates
chronological and summary reports.
After each incoming or outgoing call is completed, a call record is printed on the
Chronological Report. Each call record includes the date and time of the call, the Personal
Dial Code (PDC) of the station involved, the call’s duration (in minutes), the call type (voice
or data), the calculated cost, and the number of the facility on which the call was made. If
the call is outgoing, the call record also includes the number dialed, a two-character
abbreviation for the area to which the call was made, an account code, and the PDC claiming
accountability for the call.
When a call record with a new date is received by the CAS 100, the previous day’s
Chronological Report is ended and reports containing summary information are printed.
Four summary reports are produced:
.
.
.
.
Calls by Hour of the Day
Calls by Facility Used
Calls by Extension Number
Calls by Account Code.
These reports summarize the information presented in the Chronological Report and list the
total number of calls, their total duration, and the total cost.
Switches inside the CAS Model 100 cabinet allow the System Administrator to control the
system options that determine report parameters.
CAS Models 200, 300, 500, and 2000:
The System 25 SMDR interface provides direct output to an AT&T PC 6300 Personal
Computer equipped with MS-DOS (V2.11 or later) and a CAS software package (i.e., Model
200, 300, 500, and 2000) that calculates the cost of calls and provides basic and sophisticated
call reports.
After a telephone call is completed, System 25 sends a call record to the AT&T PC 6300 via
the SMDR interface channel. The PC must be equipped with CAS software. Call records are
collected by the PC and held in a buffer until they are processed. When a call record is
processed, a cost is calculated and assigned to it. That cost, along with other call record
information, is then stored on a hard disk for subsequent retrieval.
Two modes of operation are available for PC 6300 operation:
. Dedicated Mode: The PC is dedicated to one and only one task–processing call
records.
. Multi-Function Mode: Allows the user to print reports, edit files, and run other
PC-based programs while the CAS continues to collect and buffer call records in the
background. The user must enter the Dedicated Mode to process calls and generate
reports.
2-66
The CAS performs three main functions; (l) call record processing, (2) report generation, and
(3) CAS system management. In addition, a limited directory lookup and message center is
provided. The followings a brief description of each function:
1. Process Calls: Involves screening call records, calculating the cost of valid calls,
anti storing the call records.
2. Generate Reports: Allows the user to print the stored call record information
organized in one of several different ways. Users can select a report or set up their
own special combination of reports from the following:
. Summary
Reports—A collection of reports that condense and summarize
call record information by total number of calls, duration, and cost. The
reports can be organized by department, call type, cost center, trunk,
extension, cost, duration, time of day, date, and account code.
Detail Report–A detailed report of each call record in the
. Organization
system, sorted by department, cost center, and extension.
. Selection Report—This report can contain at a user’s option, summary or
detailed information based on any combination of the following items: time
of day, date, cost, duration, extension, access code, account code, dialed
number, call type, department, or cost center.
Code Detailed Report—A detailed report on call records sorted by
. Account
account code. This report can be used for billing clients for calls made in
their behalf.
Preselected Reports—Allows up to five predefine reports, which can
. include
any of the above mentioned reports. These reports can run upon
request or at a specified time and date.
3. System Management: Allows the user to perform several functions. These
include editing the table of departments, cost centers, and extensions; setting up
account codes; defining preselected reports; and keeping call rate information up to
date. System configuration may be changed. This allows the user to inform the
CAS of changes in System 25 (e.g., dial access codes, trunks) or changes in charge
rates. System housekeeping may also be performed. This includes establishing
passwords, deleting call records, determining call processing options, and
performing various disk maintenance operations.
4. Directory Lookup and Message Center: Allows the user to look up anyone by
last name, first name, or extension. Messages can be recorded for individuals and
can be printed or displayed.
2-67
The following table summarizes CAS station and account code capacities.
AT&T CALL ACCOUNTING SYSTEMS
CAS Model
Stations
Account Codes
200
100
5000
300
150
5000
500
500
5000
2000
2000
15000
Refer to the CAS documentation supplied with the software package for additional
information.
Considerations
The CAS provides customers with an efficient tool to control and manage their telephone
usage and costs. The information available can be used to facilitate cost allocation, traffic
analysis, and abuse control.
Interactions
None
Administration Requirements
System:
. Send SMDR Records To SMDR Port (Yes, No) - Default = Yes
length (seconds) of calls that are reported by SMDR (10-255) - Default =
. Minimum
40.
Hardware Requirements
CAS Model 100 requires a CAS Microprocessor and a Custom Cartridge containing rate
information specific to your location.
CAS Model 200, 300, 500, or 2000 applications software must be run on an AT&T PC 6300
equipped with MS-DOS (V2.11 or later).
Refer to Figures 2-13 and 2-14 for typical CAS connection information.
2-68
C
C
C
D
C
C
C
D
D
C
C
C
C
C
C
C
C
C
D
C
C
C
C
C
DATE
TIME
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
10/08/85
11:14
11:14
11:15
11:17
11:19
11:20
11:34
11:44
11:50
12:14
12:24
13:27
13:14
13:23
13:28
13:33
13:45
14:14
14:24
14:34
14:43
15:14
15:16
15:19
CALLED NUMBER
1-232-566-1321
IN
1-322-564-1376
1-222-564-2171
IN
IN
1-242-563-1324
555-4541
1-252-514-3176
IN
1-222-566-2544
1-333-513-1376
1-244-564-3121
IN
IN
1-222-516-1176
555-2541
1-222-563-4321
1-343-516-2574
555-3141
IN
1-343-564-1321
1-222-566-1321
IN
DUR
FAC STN
00:15:41
00:09:05
00:29:50
00:10:45
00:05:32
00:29:45
00:19:00
00:05:35
00:19:45
00:25:42
00:10:35
00:15:05
00:09:40
00:15:45
00:19:35
00:19:40
00:09:05
00:20:42
00:10:05
00:09:45
00:19:32
00:20:45
00:19:45
00:19:45
9
2145
9
9
3214
2342
9
9
9
2145
9
9
9
3414
3421
9
9
9
9
9
3214
9
9
2342
1794
1324
1744
2001
1744
3455
1677
2312
3455
1492
1244
3566
2001
1566
3421
1492
3655
4321
1244
4633
2351
1794
1794
1794
ACCOUNT
PDC
123489764321341
4271
766544
3254
76322
323489764321341
3422
3422
123489764321341
4271
123489764321341
4271
763444
3465
1234893764321341
4271
Figure 2-10. Typical SMDR Call Detail Report
2-69
ASCII CHARACTER
POSITION
(Column Number)
01
DESCRIPTION
Call Type
VALID
CHARACTERS
C or D
Space
Date: Month
/
0-9
/
06-07
08
Date: Day
/
0-9
/
09-10
11
Date:Year
0-9
Space
Time:Hour
0-9
15-16
Time: Minute
0-9
17
18-35
Space
Dialed Number
Error Character
02
03-04
05
12-13
14
36
37
0-9, Space, -, IN
?, or Space
Space
Duration:Hour
0-9
41-42
43
Duration: Minute
0-9
44-45
46
Duration: Second
0-9
38-39
40
47-51
52
53-56
57-58
59-73
74
75-78
79
Space
Facility
Space, 0-9, #
Space
Station
0-9, Space
Space
Account Code
0-9, Space, #, ?
Space
Personal Dial Code
0-9, Space
Carriage Return
Line Feed
80
Figure 2-11. SMDR Call Record Format
2-70
ASCII CHARACTER
POSITION
00
01-03
04-07
DESCRIPTION
(top of form)
Space
DATE
08-12
13-16
Space
TIME
17-23
24-29
Space
CALLED
30
31-36
37-38
39-41
42-48
49-51
52-54
55-57
58-62
63-69
70-76
77-79
80
81
Space
NUMBER
Space
DUR
Space
FAC
Space
STN
Space
ACCOUNT
Space
PDC
(Carriage Return)
(Line Feed)
Figure 2-12. SMDR Call Record Header Format
2-71
SYSTEM 25
CABINET
.
PART OF
OCTOPUS
CABLE
—
ZTN82
OR ZTN128
CALL
PROCESSOR
—
—
—
—
—
— —
C2
PART OF
SIP
Z21OA
ADAPT .
— —
—
C1
355A/AF
SMDR OUTPUT
DEVICE/CALL
ACCOUNTING
SYSTEM
—
LEGEND:
C1 -MODULAR CORD (D8W-87) - PEC 2725-07G
C2 -OCTOPUS CABLE (WP90780) - PEC 2720-05P
355A ADAPTER RS 232 PLUG TO MODULAR JACK - PEC 2750-A24
355AF ADAPTER RS 232 RECEPTACLE TO NODULAR JACK - PEC 2750-A25
Figure 2-13. SMDR Output Equipment or Call Accounting System—On-Premises
Direct Connections (Sharing Same AC Outlet)
2-72
SYSTEM 25
CABINET
ZTN82
OR ZTN128
CALL
PROCESSOR
SMDR OUTPUT
DEVICE
OR CALL
ACCOUNTING
SYSTEM
PART OF
OCTOPUS
CABLE
C2
PART OF
SIP
Z210A
ADAPT.
C1
355AF
C3
ADAPT.
Z3A4
ADU
C1
C4
Z3A1
OR Z3A4
ADU
(NOTE)
C1
B1
W1
C7
2012D
TRANS.
248B
ADAPT.
LEGEND:
B1 - TYPICAL-103A CONNECTION BLOCK*
C1 - MODULAR CORD (D8W-87) - PEC 2725-07G
C2 - OCTOPUS CABLE (WP90780) - PEC 2720-05P
C3 - EIA CROSS OVER CABLE (M7U-87) - PEC 2724-30C
C4 - ADU CROSS-OVER CABLE (D8AM-87) - PEC 2724-38X
W1 - 4 PAIR INSIDE WIRING CABLE*
355AF ADAPTER RS-232 RECEPTACLE TO MODULAR JACK - PEC 2750-A25
Z3A1 ADU – EQUIPPED WITH A THREE FOOT PLUG-ENDED EIA CORD - PEC 2169-001
Z3A4 ADU – EQUIPPED WITH A 3 FOOT RECEPTACLE ENDED EIA CORD
C7 - MODULAR POWER CORD (D6AP-87)
248B ADAPTER – MODULARIZES 2012D TRANSFORMER
PEC 21691
400B2 ADAPTER – POWER ADAPTER
20210 TRANSFORMER – 15-18V AC TRANSFORMER
NOTE: CAS MODELS 100 AND 200 REQUIRE Z3A4 ADU.
* - FURNISHED BY INSTALLER
Figure 2-14. SMDR Output Equipment or Call Accounting System—On-Premises
Direct Connections (Greater Than 50 Feet From System Cabinet)
2-73
SYSTEM 25
CABINET
ZTN82
OR ZTN128
CALL
PROCESSOR
(PORT X)
TDM
BUS
TN726
DATA
LINE
PART OF
OCTOPUS
PART OF
CABLE
SIP
— — —
C2
Z210A
C1
355AF
ADAPT.
ADAPT.
— — —
PART OF
SIP
— — —
C2
400B2
Z210A
ADAPT.
ADAPT.
— — —
.
Z3A4
ADU
.
C1
C7
2012D 2 4 8 b
TRANS. ADAPT.
TN726
DATA
LINE
C2
PART OF
SIP
SIP
ADAPT .
W1
B1
C1
Z3A1/A4
ADU
SMDR OUTPUT
DEVICE
(NOTE)
LEGEND:
B1 – TYPICAL-103A CONNECTING BLOCK*
C1 – MODULAR CORD (D8W-87) - PEC 2725-07G
C 2- OCTOPUS CABLE (WP90780) - PEC 2720-05P
C3 – EIA CROSS OVER CABLE (M7U-87) - PEC 2724-30C
W 1– 4 PAIR INSIDE WIRING CABLE*
3 5 5 A F -ADAPTER (RS-232 RECEPTACLE TO MODULAR JACK) - PEC 2750-A25
Z3A1 ADU – EQUIPPED WITH A 3 FOOT PLUG-ENDED EIA CORD - PEC 2169-001
Z3A4 ADU – EQUIPPED WITH 3 FOOT RECEPTACLE ENDED EIA CORD - PEC 2169-004
248B ADAPTER– MODULARIZES 2012D TRANSFORMER \
400B2 ADAPTER – POWER ADAPTER
PEC 21691
2021D TRANSFORMER -15-18V AC TRANSFORMER
C 7 - MODULAR CORD (D6AP-87)
/
* – FURNISHED BY INSTALLER
NOTE: OUTPUT DEVICE OR MULTILINE VOICE
TERMINAL WITH DATA BUTTON DIALS PORT X DDC TO
ESTABLISH DATA CONNECTION TO ZTN-82.
Figure 2-15. SMDR Output Equipment —On-Premises Switched Connections
2-74
SYSTEM 25
CABINET
.
ZTN82
OR ZTN128
CALL
PROCESSOR
CO OR PRIVATE
LINE CIRCUIT
PART OF
OCTOPUS
CABLE
— — —
— — —
C2
— — —
PART OF
SIP
>
CONNECT VIA TAE
Z21OA
ADAPT.
TYPICAL
MODEM
C1
— — —
C3
TELCO
CENTRAL
OFFICE
355AF
SMDR
OUTPUT
DEVICE
TYPICAL
MODEM
LEGEND :
C 1C2 C 3355AF ADAPTER
MODULAR CORD (D8W-87) - PEC 2725-07G
OCTOPUS CABLE (WP90780) - PEC 2720-05P
EIA CROSS OVER CORD (M7U-87) - PEC 2724-30C
- (RS-232 RECEPTACLE TO MODULAR JACK) - PEC 2750-A25
Figure 2-16. SMDR Output Equipment—Off-Premises Direct Connections
2-75
SYSTEM 25
CABINET
ZTN82
OR ZTN128
CALL
PROCESSOR
PART OF
OCTOPUS
CABLE
— — —
C2
.
TN726
DATA
LINE
TN742
ANALOG
LINE
CO
TRUNK
FACILITY
Z210A
ADAPT.
Z3A4
ADU
355AF
ADAPT.
C1
— — —
TN758
POOLED
MODEM
TDM
BUS
PART OF
SIP
C2
PART OF
SIP
—
—
—
— — — —
— — —
400B2
Z210A
ADAPT.
ADAPT.
— — — — — — — — — —
.
C7
2012D 248B
TRANS. ADAPT.
CONNECTED
AS OPS
OR CO
FACILITY
CO CABLE
I
CENTRAL
OFFICE
I
— — — — — — — — — — —
SMDR
OUTPUT DEVICE
TYPICAL
MODEM
(212 TYPE)
OPS OR CO TRUNK
— — — — — — — — — — —
LEGEND :
C1 C2 C3 OPS 355AF ADAPTER Z3A4 ADU 248B ADAPTER 400B2 ADAPTER 20210 TRANSFORMER C7 -
*
MODULAR CORD (D8W-87) - PEC 2725-07G
OCTOPUS CABLE” (WP90780) - PEC 2720-05P
EIA CROSS OVER CABLE (M7U-87) - PEC 2724-30G
OFF PREMISES STATION
RS-23Z RECEPTACLE TO MODULAR JACK PEC 2750-A25*
EQUIPPED WITH 3 FOOT RECEPTACLE - PEC 2169-004
MODULARIZES 2012D TRANSFORMER
POWER ADAPTER
PEC 21691
15-18V AC TRANSFORMER
MODULAR POWER CORD (D6AP-87)
- FURNISHED BY INSTALLER
Figure 2-17. SMDR Output Equipment—Off-Premises Switched Connections
2-76
CALL COVERAGE—GROUP
Description
Allows calls to covered stations to be redirected to a group of covering stations.
A total of 32 standard Call Coverage Groups may be assigned; for V2 systems, an additional
32 “DGC” call coverage groups maybe assigned. Each standard group may be covered by up
to eight call coverage receivers (buttons). There is no limit on the number of covered users
(senders) that each Call Coverage Group can include, but a covered user can be assigned to
only one Call Coverage Group. Each call coverage receiver must have a multiline set
equipped with a Cover (COVER-GRP) button, except as noted below. A covering set may be
assigned more than one COVER-GRP button for the same or different groups.
In R1V2 Systems equipped with a Switched L OOP Attendant Consoles (SLAC), the console
queue can serve as a standard coverage group receiver. The consoles cannot have COVERGRP buttons, so the queue directs coverage calls to LOOP buttons.
Direct Group Calling (DGC) Groups may be designated as Call Coverage Group receivers in
V2 Systems. This provides the capability for System 25 to support “non-integrated” voice
mail systems as well as allow the formation of coverage pools.
Senders may be either single-line or multiline voice terminals. In V2 systems, receivers may
be single-line voice terminals if part of a DGC Call Coverage Group. Multiline voice
terminals may always be used as receivers
Standard Group Coverage
When a call arrives at a voice terminal that has group coverage, the COVER-GRP or LOOP
button status LED at the covering voice terminals will flash. Covering voice terminals will
begin to ring after a specified number of rings at idle covered voice terminals. If there is no
idle system access button at the covered station (sender), the call receives coverage
treatment, and the call immediately rings at the covering terminal(s). If no idle cover
button is available at the covering terminal(s), the calling party receives a busy signal. In
V1 Systems, the call will not receive coverage treatment if both system access buttons on the
sender’s station are busy; the calling party receives busy signal.
Ringing may be turned off at standard receiver stations for each covering button, as desired
(not recommended). If this option is selected, a flashing status LED will be the only
indication received at the covering station. In addition, Call Coverage ringing may be turned
off on internal calls (if desired) on a system-wide basis.
A member of a standard receiver group can use the Line Selection (Preelection) feature to
answer covered calls even before any audible alerting has begun at the covering user’s
terminal. This is useful if the user knows that the covered party is unavailable.
A covered voice terminal may elect to have calls covered while it is busy on another call.
Calls directed to an idle button on a busy covered multiline voice terminal will start ringing
at the covering terminal after a single burst of ringing at the busy covered voice terminal.
If there is no idle Cover button on the covering voice terminal, the call will queue and will
ring at the first available coverage receiver (V2 only); in a V1 system, the call will not
receive coverage treatment. Calls directed to a busy single-line voice terminal will start
ringing immediately at the covering terminal. If there is no idle Cover button on any
covering terminal, either individual or group, the call will not receive coverage and the
calling party will receive Busy Tone.
2-77
Calls sent to coverage will continue to ring at single-line sender terminals but will cease
ringing at multiline sender terminals. In the latter case, the calls will remain on the
incoming call appearance button, and that button’s status LED continues to reflect the
status of the call. In particular, covered calls to multiline sets remain available and can be
entered by the called (sender) station.
If a covered station activates the SEND ALL CALLS button, calls will be directed to
coverage immediately, with or without a single-ring reminder, as administered.
A station can provide (or receive) Individual Call Coverage (see Call Coverage-Individual)
and also be a member of a Call Coverage Group (sender or receiver). Unanswered calls to a
station, provided both Individual and Group Call Coverage, will first ring at the Individual
Coverage station and then, after a second delay cycle and still unanswered, will ring at the
Group Coverage station.
Calls from a covering station to a covered station will not be covered unless the covered
station has additional coverage. This is an important consideration when the attendant
provides coverage.
DGC Group Coverage (V2 only)
Calls proceeding to the DGC–Call Coverage Receiver Group hunt in a circular fashion for
the first idle station, starting from the last station to receive a call. If all DGC members are
busy, both internal and external calls continue to ring and/or flash at the covered station
and any individual coverage receiver’s station(s) until a DGC station becomes idle. If a DGC
group is used for both DGC calls and group coverage, trunk calls into a DGC group have
priority over coverage calls. Calls sent by coverage to a DGC–Call Coverage Group member
station do not receive additional call coverage.
DGC groups cannot be call coverage senders to another DGC–Call Coverage Group.
However, calls made directly to a DGC member can be covered by another DGC–Call
Coverage Group.
Once a call has been redirected to a DGC–Call Coverage Group member, the call is
transferred to the covering station. The call continues ringing until answered or dropped.
The call is not accessible at the covered station nor any individual coverage receiver once it
is redirected to an idle DGC station. If all DGC members are busy, the call remains
accessible at the covered station until a member is available.
DGC–Call Coverage Groups count against the system specified maximum number of DGC
groups, but not against the number of Call Coverage Groups. The limit of eight receiving
stations per Call Coverage Group does not apply when administering a DGC group as a Call
Coverage Group. A maximum of ten stations per DGC–Call Coverage Receiver Group is
allowed.
Considerations
Call Coverage provides a way to redirect calls to alternate answering positions. The feature
is versatile enough to permit suitable alternate answering arrangements for virtually every
level of employee. Special functions, such as the Send All Calls feature, accommodate the
day-to-day variations that occur in an employee’s work schedule.
2-78
Interactions (Standard Group Coverage)
●
Attendant Console, Direct Trunk: If the Direct Trunk attendant is a receiver for
a Call Coverage Group and extends a call (using the Start button or Selector Console)
that is unanswered/busy to a member of the group, the call will return On the
Return-On-Don’t-Answer (RTN-DA) or Return-On-Busy (RTN-BUSY) button, not on
the attendant’s COVER-GRP button.
●
Automatic Intercom: Auto-Intercom calls do not receive call coverage.
●
Call Coverage— I n d i v i d u a l : Unanswered calls to a station, provided both
Individual and Group Call Coverage, will first ring at the Individual Coverage station
and then, after a second delay cycle and still unanswered, will ring at the Group
Coverage station.
●
Call Coverage/Station Hunting: A call to a busy single-line voice terminal that is
both a member of a Station Hunting group and a Call Coverage sender will first hunt
for an idle station to service the call. If none is available, the call will be sent to
coverage.
●
Call Pickup: Call Pickup is independent of call coverage. When a call is answered
via Call Pickup, all Cover buttons associated with the called party go idle.
●
Direct Group Calling (DGC): A call to a DGC group member will receive coverage
if the member is also a Call Coverage sender. Calls to a busy DGC group do not
receive call coverage. Instead, after a predefined number of rings, the call will be
transferred to delay announcement (if provided), or ringing will be transferred to all
button appearances of the line.
●
Exclusion: If a call coverage receiver invokes Exclusion after answering a coverage
call, all other terminals (including the attendant and the covered station) are
excluded from the call. The covered user cannot enter the call until EXCLUSION is
pressed a second time by the covering user.
●
Hands-Free-Answer On Intercom: An incoming (inside) call will not receive call
coverage if auto-answer is activated, since the set will answer the call (whether the
user is present or not.).
●
Night Service: Directed Night Service calls do not receive call coverage.
●
Personal Dial Codes: Calls directed to a station because another non-floating P D C
is logged in there do not receive the coverage treatment of the logged-in station.
Such calls return to their home station and receive that station’s coverage
(immediately upon return). Calls to logged-in floating PDCs, on the other hand,
receive the same coverage treatment as any other calls to the logged-into station.
They, of course, have no home station to return to.
●
Personal Lines: Personal line calls receive the coverage of the principal (owner)
station for that line. Other line appearances (even if administered to ring) will not
receive coverage.
●
Tie Trunks: Tie Trunk calls directed at a user with call coverage receive normal call
coverage treatment.
2-79
Interactions (DGC Group Coverage; V2 Only)
●
Attendant Console, Switched Loop: If a SLAC is a member of a DGC-Call
Coverage Receiver Group, any DGC—Group Coverage call sent to this attendant will
enter the common queue and be treated as a coverage call, not as an Attendant—
DGC call. Thus, the call will be handled by whichever attendant is administered to
receive coverage calls.
●
Attendant Direct Extension Selector Console: The Selector Console can be used
to transfer and place calls to a DGC—Call Coverage Group provided the DGC group
access code appears on the console. The status LED of the DXS button lights
steadily whenever all stations in the DGC—Call Coverage Group are busy.
●
Automatic Intercom: Auto Intercom calls do not receive call coverage.
●
Call Pickup: A DGC—Call Coverage Group member can also be in a call pickup
group.
●
DSS: A DSS button can be assigned to a DGC—Call Coverage Group. The button
lights whenever all DGC members are busy.
●
Flex DSS: The Flex DSS button can be assigned to a DGC—Call Coverage Group.
●
Night Service: Directed Night Service calls do not receive call coverage.
●
Personal Line Access: All outside lines directed to a DGC—Call Coverage Group
can be assigned to button appearances in addition to the DGC—Group Coverage
assignment. If the outside lines appear at stations that also have DGC coverage by
the same group, then the operation is as follows:
When an incoming call is ringing in the DGC group, the status LEDs on the
appearance buttons light steadily, indicating that the line is busy. If the call
goes unanswered after a system-specified number of rings, then a delay
announcement is provided. The caller is subsequently put on hold and receives
music if available. If the system is not equipped with a delay announcement,
the call begins to ring at all line appearances after the system-specified
number of rings.
If the outside lines are not directed to a DGC group, but are provided DGC
Group Coverage, the feature operation is the same as for incoming calls on SA
keys except that the call appearance remains accessible at the covered station
after being directed to a DGC—Call Coverage Group member.
●
Station Hunting: Calls directed to a DGC—Call Coverage Group will not hunt.
●
Trunk Groups: Trunks can be directly assigned to DGC groups that also act as call
coverage groups. Among tie trunks, only automatic incoming tie trunks can be
translated as directed to a DGC Group.
Administration Requirements (Standard Group Coverage)
System:
●
Provide Call Coverage ringing on internal calls (Yes, No); Default = Yes.
If No, covered calls will flash but not ring at covering stations on internal calls.
2-80
.
Number of rings before call coverage ringing starts on no answer (0-31); Default = 2.
The status LEDs on Group Coverage buttons at covering stations begin flashing
immediately in all cases. Ringing, in addition to flashing, is always sent on external
calls, though it may not be accepted at the covering stations.
Voice Terminal: (Station Port)
.
.
.
.
Coverage Sender group number (l-32) - Default = 1
Provide Call Coverage ringing on no answer (Yes, No); Default = Yes.
If No, flashing LED is the only indication received at the covering station.
The calling party always hears ringing.
Provide Call Coverage ringing on busy (Yes, No); Default = Yes.
If No, flashing LED is the only indication received at the covering station.
If the covered station is a busy single-line set or a multiline set with one System
Access appearance busy, the call is sent to coverage and the calling party hears
ringing. If the covered station is a R1V1 multiline set with both System Access
appearances busy, the call cannot go to coverage and the calling party hears busy
tone; in R1V2 the call goes to coverage.
Call Coverage Receiver button
— Group Number (1-32)
— Allow Ring At Destination (Yes, No); Default = Yes.
If No, flashing LED is only indication received at the covering station.
Administration Requirements (DGC Group Coverage; V2 Only)
The parameter to assign a call coverage sender group has been expanded to include DGC
groups as coverage group receivers. DGC receiver groups are first set up as regular DGC
groups, numbered 1-32. DGC coverage groups can then be specified by using coverage group
numbers 101-132, where coverage group 101 has DGC group 1 as its receiver group, coverage
group 102 has DGC group 2 as its receiver group, etc. No button assignments are required.
Send ringing options (on busy, on no answer, system-wide for internal calls) have no effect
for DGC group coverage; all calls sent to coverage.
The system search of group call coverage sender stations has been expanded to allow the
system administrator to enter a DGC—Call Coverage Group and list all its sender stations
by PDC.
Hardware Requirements
None
2-81
CALL COVERAGE—INDIVIDUAL
Description
This feature is very similar to Call Coverage–Group. The primary difference is that this is
a one-on-one type coverage.
An Individual Cover (COVER IND) button can be assigned on multiline voice terminals to
cover calls to a specific (single) voice terminal. The covering station can answer covered
calls by pressing COVER-IND. Each button can be programmed to ring or not to ring. If
ringing is selected, the covering station will begin ringing after a specified number of rings
at the covered station. When the specified number of rings has occurred, multiline voice
terminals will stop ringing. Single-line voice terminals continue to ring until the call is
answered at a covering terminal. When the call is answered at the covering station, the call
remains accessible at the call appearance button of multiline voice terminals, but is no
longer accessible at single-line voice terminals.
Covered calls will appear on the COVER-IND button; all calls except Automatic Intercom
and Directed Night Service calls are covered.
Each Cover button at a covering station represents one covered voice terminal. If more than
one voice terminal is to be covered, multiple buttons are required, one for each station
covered. A covering voice terminal may be assigned multiple COVER-IND buttons for a
particular station to cover multiple simultaneous calls to that station. The first button will
track the first call, the second button, the second call, etc.
Up to eight COVER-IND buttons can be assigned for each covered station.
For VI systems, up to 31 stations may receive Individual Call Coverage. There is no
limitation for V2 systems.
A voice terminal can receive both Individual Call Coverage and Group Call Coverage.
Refer to Messaging Services for a description of Call Coverage—Message Waiting service,
which allows the covering station to control the status of the covered user’s Message LED.
Calls from a covering station to a covered station will not be covered unless the covered
station has additional coverage. This is an important consideration when the attendant
provides coverage.
Considerations
Call Coverage provides a way to redirect calls to alternate answering positions. The feature
is versatile enough to permit suitable alternate answering arrangements for virtually every
level of employee. Special functions, such as the Send All Calls feature, accommodate the
day-to-day variations that occur in an employee’s work schedule.
The Call Coverage–Individual feature is not administrable on the Switched Loop Attendant
Console.
For V2 only: COVER-IND buttons can be used to provide the ability to receive more than
two incoming calls at a time. Up to eight COVER-IND buttons can be administered on a
multiline voice terminal, providing “individual call coverage” for itself. When both System
Access buttons are busy, subsequent incoming calls will be directed to available COVER-IND
buttons on the set, thus simulating additional System Access buttons (for incoming calls
only).
2-82
Interactions
●
●
Attendant Console, Direct Trunk: When a coverage call rings at a busy
Attendant Console, the attendant receives a single burst of ringing. If the call is still
unanswered when the attendant hangs up, the Attendant Console will resume
ringing.
Call Coverage —Group: Unanswered calls to a station, provided both Individual
and Group Call Coverage, will first ring at the Individual Coverage station and then,
after a second delay cycle and still unanswered, will ring at the Group Coverage
station.
●
Direct Station Selection (DSS): Calls placed via a DSS button to a user with
Individual Call Coverage will receive call coverage. When a DSS button is used to
activate the busy-to-idle reminder for the user, the reminder is returned only when
the user becomes idle, not when an associated coverage user becomes idle.
●
Exclusion: If a covering station answers a coverage call and then invokes Exclusion,
all stations including the covered one, are excluded from the call.
●
Hold: May be used to place a coverage call on hold. The COVER-IND button’s
status LED winks at the covering station and the call appearance’s status LED winks
at the covered terminal. The held call will automatically leave the coverage terminal
if picked up by the covered user. The covering station will be unable to reenter the
call.
●
Transfer: When a covering station transfers a covered call to another station, the
call will no longer appear at the covering station’s Cover button, though it still will
appear at covered multiline stations.
Administration Requirements
Voice Terminal (Station Port):
●
Individual Call Coverage button (button function # 12)
— Individual Coverage PDC (l-9999)
— Individual Coverage Ring (Yes, No).
Hardware Requirements
None
2-83
CALL FOLLOWING (FORWARDING)
Description
Allows users who are away from their phone to receive calls at another phone. The feature
is functionally equivalent to Call Forwarding.
Users can log in their PDC at another voice terminal and receive their calls at that terminal.
I.egging in a PDC at a terminal automatically logs the PDC out at any other terminal.
PDCs will always have an associated home terminal. If a PDC is not logged in anywhere,
calls to the PDC will ring at the home terminal.
Logging in a FPDC automatically logs the FPDC out at any other voice terminal. However,
logging out a FPDC does not log the FPDC in at another terminal. Calls to FPDCs not
logged in at a terminal may be directed to the attendant (administrable and strongly
recommended).
When a PDC is logged in at another terminal (away terminal), dialing the PDC from the
away terminal directs the call to the PDC’s home terminal.
To log in a PDC, the user goes off-hook at the “away” terminal, enters “**” and their PDC
twice. The PDC logout procedure is similar; after going off-hook and receiving dial tone, the
user enters “**” followed by their PDC and then by a “0.” A “**0” entered at a voice
terminal will log out all PDCs and FPDCs logged in at that voice terminal. Confirmation
Tone is returned to a user who correctly completes one of these procedures.
The following call types or features are voice terminal oriented (associated with stations
rather than PDCs) and do not follow a user who logs in at another terminal.
.
.
.
.
.
.
.
Automatic Intercom Calls
Directed Night Service calls
DGC Group Calls
Manual Signaling
Message Waiting
Outward/Toll Restriction.
Personal Line Calls
In R1V1 systems, Direct Station Selection (DSS)/Flex DSS calls do not receive call following
treatment. This limitation does not apply to R1V2 systems.
Considerations
Call Following provides maximum flexibility to system users who are away from their voice
terminals. In addition, visitors can receive calls by logging in an assigned FPDC.
For more information, see the “Personal Dial Code (PDC)” feature description.
2-84
Interactions
●
Call Coverage: Calls to a logged-in FPDC receive the call coverage of that
terminal. [Unanswered calls to a PDC at an away terminal return to the home
terminal and receive the home terminal’s call coverage treatment; they do not receive
the away terminal’s call coverage.
Administration Requirements
None
Hardware Requirements
None
2-85
CALL PARK
Description
Allows a user to place a call on hold and then pick up the call from any voice terminal in the
system. It is used in three typical applications:
.
.
.
Call Park: A user places a call on hold and then picks it up at another voice
terminal
Meet-Me-Conference: A conference member places the conference on hold and
pages another employee to join the conference.
Transfer: A user places a call on hold and then pages another employee to pickup
the call
Call Park allows a user to move a held call to a “parked” position by dialing the Call Park
code (*5). The call can subsequently be retrieved from any voice terminal by dialing the Call
Park retrieval code (*8) and the PDC of the parking station. In addition, any user active in
a conference involving fewer than five members may park the conference, so that another
user may join the conference.
A multiline voice terminal user may invoke Call Park by pressing HOLD to place a call or
conference on hold, then pressing an idle System Access button and dialing *5. A single-line
voice terminal user may invoke the feature by pressing the switchhook to place the call or
conference on hold, then dialing *5. If the call is successfully parked, the user receives
Confirmation Tone and then Recall Dial Tone. If the call cannot be parked, Reorder Tone is
received. In the latter case, to return to the held call, press the held call button (multiline
sets) or flash the switchhook (single-line sets).
The parking station may return to a parked call or conference without affecting the park
state. The multiline voice terminal user may return by pressing the held call button. The
single-line user may return by pressing the switchhook.
When the single-line user goes on-hook, the parked call is removed from the terminal and
cannot be reentered.
To retrieve a parked call, a user must obtain system dial tone, dial *8 and then dial the PDC
of the station that parked the call. If the call is not retrieved within 2 minutes the call will
return to the user that parked the call.
A call may be parked by the attendant using the same procedure as a multiline voice
terminal. In addition, if the Attendant Position is equipped with a Selector Console, up to
eight additional calls may be parked by dedicating any eight of the console DXS buttons to
the call park function. A call may then be parked using the Selector Console by pressing one
of these dedicated Call Park buttons. The status LED of the parked line on the Attendant
Console winks and the status LED of the Call Park button on the Selector Console lights
steadily. The call can subsequently be retrieved from any voice terminal by dialing the Call
Park retrieval code (*8) and the PDC of the DXS button on which the call is parked.
A call parked by the attendant using the same procedure as a multiline voice terminal will
return to the Attendant Console’s System Access (DTAC) or LOOP (SLAC) button if it is not
picked up within 2 minutes. A call parked with the Selector Console that is not picked up
within 2 minutes will return to the RTN-DA button (DTAC) or a LOOP button (SLAC), in
the same manner as any other unanswered call.
2-86
A call parked with the Selector Console is retrieved by dialing *8 and the access code
assigned to the dedicated Selector Console Call Park button.
Each voice terminal user (except the attendant) can only park one call at a time and a
maximum of 24 calls can be parked in the system at one time. A call is no longer parked
when it is answered, returns to the parking terminal, or is abandoned by the caller.
Considerations
Call Park can be used whenever a user engaged on a call needs to go elsewhere, and wishes
to complete the call from another terminal. Call Park also allows users to answer a call
from any voice terminal when paged.
Interactions
●
Attendant Direct Extension Selection: Station-To-Station calls cannot be parked
via the Call Park buttons on the Attendant Selector Console.
●
Attendant Position Busy: If a call is parked on an attendant console and the
attendant console enters Position Busy mode, the parked call will return to the
inactive console if not answered within two minutes.
If a call is parked on the Selector Console by a Switched Loop attendant (V2 only)
and the SLAC is placed in the Position Busy mode, the parked call will return to the
other active attendant console, if not answered within two minutes.
●
Conference: Parked conference calls do not return to the parking voice terminal.
They remain parked. Call Park may be used to place a conference on hold if it
contains fewer than five parties.
●
Exclusion: A call cannot be parked, and a parked call cannot be answered, if the
Exclusion feature is invoked on that call.
●
Intercept Treatment With Reorder Tone: An unsuccessful attempt to park a
call due to misdialing or attempting to park more than one call at a voice terminal
results in Reorder Tone.
●
Music-On-Hold: Parked calls (except conferences) receive music.
●
Outward Restriction (see “Calling Restrictions”): If the parking station is
outward restricted, the Recall Dial tone following a successful parked call cannot be
used to avoid restriction.
●
Personal Line: A parked Personal Line is bridgeable by any user with a button
appearance of that line. Bridging on to the connection does not answer the parked
call. The parked call will not return to the parking voice terminal user in this case.
●
Toll Restriction: (see “Calling Restrictions”) If the parking voice terminal user
is toll restricted, the Recall Dial tone following a successful park cannot be used to
avoid the restriction.
2-87
Administration Requirements
Attendant Selector Console:
●
Assign Call Park buttons.
Hardware Requirements
None
2-88
CALL PICKUP
Description
Allows a user to answer a call ringing at another voice terminal. There are two forms of
Call Pickup; (l) Directed and (2) Group.
Directed Call Pickup:
Directed Call Pickup allows calls to most other voice terminals, including Automatic
Intercom calls and calls ringing at coverage buttons, to be “picked up” by dialing the Call
Pickup code (*7) and the ringing terminal’s PDC. Picked-up calls remain accessible at the
call appearance button of multiline terminals, but are no longer available at single-line
terminals.
A ringing call can be answered at a busy- single-line voice terminal by pressing the
switch hook, which will place the current call on hold, dialing *7, and the ringing voice
terminal’s PDC.
This is a standard feature available at every voice terminal. No administration is required.
Also, this feature cannot be turned off or restricted.
Group Call Pickup:
Group Call Pickup permits calls to another terminal in the Call Pickup group to be
answered. Any call, including Automatic Intercom calls, is eligible for Call Pickup.
A member of a Call Pickup Group can answer any call to any other member of the group by
dialing the Group Call Pickup code (*70).
Up to 16 groups (with up to 16 voice terminals in each group) can be set up. Each Call
Pickup group can have a maximum of two simultaneous ringing calls eligible for Call Pickup
treatment at a time, and the calls are picked up in order of arrival. A user can be assigned
to only one Call Pickup Group.
If the picked-up call was to a multiline terminal, the called terminal can still enter the call.
If the called terminal was a single-line terminal, it cannot enter the call once it is picked up.
Call Pickup cannot be invoked after the call has been answered.
If no eligible calls are ringing, attempting a call pickup results in Reorder Tone.
Considerations
With Call Pickup, users do not have to leave their own voice terminal to answer a call at a
nearby voice terminal. Instead, a user simply lifts the handset and dials an access code.
This allows calls that may go unanswered to be handled quickly and efficiently.
The call must be administered to ring at the voice terminal for which Call Pickup is
attempted. Otherwise the attempt will be blocked.
Interactions
.
Call Coverage: When a call is directed to a coverage station and the call is
answered via Call Pickup, all Cover buttons associated with the call go idle.
2-89
●
Personal Line: After a call is picked up from a Personal Line (PERS LINE) button,
the called terminal can still enter the call.
Administration Requirements
Voice Terminal (Station Port):
●
Assign Call Pickup Group Number (1-16, None) - Default = None.
Hardware Requirements
None
2-90
CALL PROGRESS TONES
Call Progress tones provide audible feedback on the status of calls during call set-up.
●
●
Busy Tone: A slow pulsed tone indicating that all facilities for answering the call
are in use.
Call Waiting (Camp-On) Tone: A single short tone to a busy terminal indicating
that another call is waiting (has been “camped on” by the attendant).
●
Confirmation Tone: Three short tones indicating that the system has accepted the
instruction entered or that your ARS queued call is being placed.
●
Dial Tone: A steady tone indicating that dialing or feature activation can begin.
●
Dialing Feedback: Indicates that a digit has been dialed.
●
Preferred Route (Queuing) Tone (ARS): Five short tones. Indicates no facility
is currently available to place your call. If you remain off-hook, your call will be
placed as soon as a facility becomes available.
●
Reorder Tone: A fast pulsed tone indicating that all trunks are busy, a dialing
error has occurred, or the terminal is restricted from making this call.
●
Ringback Tone: In general, indicates that a called terminal is ringing.
For additional information, refer to “Tones” in Section 5.
2-91
CALLING RESTRICTIONS
Description
Designated voice and data terminals can be restricted from making certain types of calls.
Available restrictions are:
●
Outward
Restriction
●
Toll Restriction
●
Facility Access Restriction
●
ARS Restrictions.
Note: Each of these restrictions is voice terminal oriented, not PDC oriented.
Outward Restriction:
When outward restricted, a station will be unable to place any outside calls. The station will
be able to answer incoming calls and place and receive inside calls. A station that is
outward restricted will be unable to use Automatic Route Selection to place external calls
except to the emergency numbers.
Toll Restriction:
Allows calls by restricted terminals to be made based on as many as the first six digits of
the number called (after the facility access code). Toll restricted users can make outgoing
calls only to those numbers that are on the Toll Call Allowed (TCA ) Lists to which it has
access. TCA entries must be in the form NXX or NPA-NXX (exactly three or six
characters. ) The system administrator can establish up to 4 individual lists. A list can
contain from 1 to 64 entries provided that the total of all four lists does not exceed 64. One
character “.” can be specified as a wild card character in place of the last 1, 2, or 3-digits
(e.g., “NX.”, “N..”, or “...” ) of the NXX code, but not in the NPA code. When this character
is used, any character in the dialed number appearing in that position is acceptable. Those
stations assigned Toll Restriction Class 1 have access to all four TCA Lists; Class 2 stations
just lists 2 through 4, Class 3 stations just lists 3 and 4, Class 4 stations just list 4. The entry
of an area code followed by “...” on a TCA List allows access to all office codes in that area.
To allow calls within a customer’s local area, individual office codes are entered; this allows
the customer to restrict toll calls within the local calling area. NPA-NXX entries allow
specific office codes to be called within an area. Note, NPA only entries are not permitted
(use NPA-... ).
Note that stations that are toll restricted are only toll restricted on CO trunks (type 701 and
801 ) or when they use the ARS feature. They will not be toll restricted when they dial
access (or button access) any other type of trunk (e.g., FX, WATS, or Tie trunks).
Facility Access Restriction:
Any station may be denied dial access to the local CO and/or to all other pooled facilities (as
a group). A station so restricted may only dial access those facilities via the Automatic
Route Selection (ARS) feature.
In addition, each trunk (VI and V2) and Virtual Facility (V2 only) can be administered to
allow or restrict dial access. If dial access is restricted, the trunk or Virtual Facility (V2)
may only be dial accessed via ARS.
2-92
ARS Restriction:
Special restrictions on each station may be imposed when the call is routed by the ARS
feature. Facility Restriction Levels (FRLs) are used to restrict access to trunk groups. An
FRL is a single digit (0, 1, 2, 3). A terminal assigned an FRL of 0 has the least privileges, a
terminal assigned an FRI. of 3, the most. An FRL is also assigned to each route in each
ARS routing pattern. The terminal’s FRI. must be equal to or greater than the route’s FRL
in order to use that facility.
Considerations
Restrictions are used whenever it is necessary to restrict certain users from accessing
designated facilities. A typical application is to deny most stations dial access to all trunk
groups. This forces callers to use the ARS feature, which should result in reduced toll
charges.
Interactions
●
Automatic Route Selection: The use of the ARS feature will not allow users to
avoid restrictions. Outward Restriction and Toll Restriction, when administered, can
prevent calls originating at associated voice terminals from routing via ARS.
Facility access restrictions, however, are circumvented.
●
Direct Facility Access: Toll restricted stations receive standard toll restriction
treatment on all Direct Facility Access (FACILITY) buttons.
●
Personal Lines: Personal Lines are subject to the toll restriction options of the
stations on which they appear.
●
Repertory Dialing: A user cannot use Repertory Dialing to access a number that
he/she is restricted from dialing.
●
Speed Dialing: A user cannot use Speed Dialing to access a numberthathe/she is
restricted from dialing.
●
Transfer: A non-restricted user (typically the attendant) can transfer a CO trunk to
an outward restricted or toll restricted station, giving the station outward service.
The toll restriction class of the transferring station will apply for calls placed over a
transferred trunk.
Administration Requirements
Terminal (Station Port):
●
Restrict access to CO trunk pool (Yes, No) - Default = No
●
Restrict access to all other trunk pools (Yes, No) - Default = No
●
Restrict outward calls (Yes, No) - Default = No
●
ARS Facility Restriction Level (Level Number 0-3) - Default = 3
●
Specify Toll Restriction Class (Class Number 1-4, None) - Default = None (not
restricted).
2-93
Automatic Route Selection:
●
Route
Facility
Restriction
Levels
Trunk Port:
●
Allow dial access to this trunk (Yes, No) - Default = Yes
System (V2 only):
●
Allow dial access to this virtual facility (Yes, No) - Default = No
Hardware Requirements
None
2-94
COMMAND MODE
Allows data terminal users to originate data and voice calls and change (or view) their data
port options.
(Refer also to the general description of the system’s data features in “Data Services
Overview”; users who are familiar with Command Mode options should read the “Expert
Mode” subsection. )
Command Mode supports digital data endpoints connected via Asynchronous Data Units
(ADUs) to ports on a Data Line CP (TN726). Command Mode also supports, with certain
restrictions, users of the STARLAN Interface Circuit Pack, ZTN84 (V2 only).
Command Mode is invoked from a data terminal in the idle (on-hook) mode by:
Terminal Optioned For Autobaud
Terminal Not Optioned For Autobaud
Enter Break followed by Return
Enter Break
The terminal then displays the Command Mode menu:
<Data call>
<Voice call>
<place Data call>
<Options>
<Options>
<Hangup> (V2 only)
<Hangup> (V1 only)
Menu items are chosen either by positioning the cursor under the desired item (by typing
“space” characters) and entering a Return, or by typing the upper-case character in the
menu field (e.g., type “D” to choose data terminal dialing or “O” to move to the options submenu). Once a user has entered Command Mode, the terminal is considered off-hook and
busy to incoming calls until it returns to the idle mode.
<Data call> or <place Data call>:
Refer to the “Data Terminal Dialing” feature description for information on how to dial
from your terminal. Once a data call has been set up, either the “Disconnect/Recall
Sequence” (see the Permissible Options, Table 2-F) must be sent or <Hangup> selected from
the above menu to terminate the data call and return to idle mode. If a data call is not
answered, the caller must disconnect by sending a Break.
<Voice call> (V2 only):
The data terminal user can originate a call for an on-premises source voice or data terminal
to a remote terminal by selecting <Voice call> and dialing the required digits. Refer to the
“Third-Party Call Setup” feature description for a complete description of this feature.
2-95
<Options>:
If the Command Mode menu item <Options> is chosen, the terminal displays the data port’s
administered options. System default values for each option are also shown. The display
shown below is similar to what is actually presented on the screen. (See Tables 2-F and 2-G)
for additional information on options. )
OPTIONS
CURRENT
DEFAULT
Speed (highest)
Parity
Mismatch
Local Echo
Answer Text
Connect Indication
Recall Sequence
9600
Even
Yes
Yes
Yes
Yes
Br-Br
19200 (Auto)
Even
No
Yes
Yes
Yes
Br-Br
<eXit>
<Change options>
<View options>
In V1 systems, only the System Administrator can change options. In V2 systems, the
System Administrator can still perform the task or can authorize data terminal users to
change their own options. Selection of <Change options> from the sub-menu shown above
allows the data terminal user to change the values in the CURRENT column. Refer to the
“User Changeable Options” subsection for complete information on this feature.
<Hangup>:
The <Hangup> option can be used to terminate a connection to the data port. For V2
systems only, this option is needed for AT&T STARLAN NETWORK endpoints to disconnect
from a STARLAN Interface CP through Command Mode. Accessing <Hangup> provides the
user with these options:
●
<eXit>— Do not hang up. Return to the top level of Command Mode.
●
<All>—Hang up (disconnected)
2-96
TABLE 2-F. Partial List of Permissible Data Port (TN726) Options
DEFINITION
OPTION
Speed (61-68)†
Autobaud, Low *, 300, 1200, 2400, 4800, 9600, 19200
Parity (69)
Odd, Even, 0, 1. The 0 and 1 choices are not shown
on the user’s display.
Enable Command Mode (70)
Yes or No. Must be On for Command Mode
(i.e., Command Mode Menu display). Not shown on
user’s options display.
V2 only: Allow user to
change data port options? (71 )
Yes or No. Used to enable/disable User Changeable
Options feature. Not shown on user’s options display.
Permit Mismatch (72)
Yes or No. Allows two data endpoints to
communicate at different rates.
Local Echo (73)
Yes or No. Determines whether characters from the
data equipment will be echoed by System 25 during
Command Mode.
Answer Text (75)
Yes or No. Enables call progress messages to be
displayed at the called data endpoint.
Connected Indication (77)
Yes or No. Yes indicates that users who have
Command Mode enabled will receive the
“CONNECTED” message when a connection has been
established. If Command Mode is disabled, the Data
Line port control lead will be “raised” when a
connection is established. Usually set to “No” for
host computer endpoints.
Disconnect/Recall Sequence
(74 )
One Long Break or Two Short Breaks; the sequence
used to disconnect a data call.
* A terminal whose baud rate is low cannot use the Command Mode feature. Call
origination at this terminal must be via Transfer To Data.
† Numbers in () indicate the action numbers used to administer data port options.
2-97
TABLE 2-G. Typical Option Profiles for Different Types of Data Port Endpoints
Options
Speed (highest)
Host
Data Term,
Modem
Modem
or PC
Computer
(users)
(computer)
19200 (Auto) 9600 *
Modem Speeds Modem Speeds
Parity
Even
Enable Command Mode Yes
Even
Even
Even
Yes
Yes
Yes
Permit Mismatch
Yes
No
Yes
No
Local Echo
Yes
No
Yes
No
Answer Text
Yes
No
Yes
No
Connected Indication
Yes
No
Yes
No
Disconnect/Recall
Sequence
Br-Br
Br-Br
Br-Br
Br-Br
* or 9600( Auto) if not not used primarily for Host-Host communications
2-98
COMMUNICATIONS ACCESS MANAGER (CAM) (V2)
The CAM program facilitates communication between workstations on the AT&T STARLAN
NETWORK (STARLAN NETWORK) and workstations on System 25. Detailed procedures
for using CAM can be found in the CAM User Guide. The material here provides a brief
overview of CAM capabilities.
CAM is an MS-DOS application program that provides an enhanced calling interface and
terminal emulation for PCs. CAM, combined with System 25’s Third-Party Call Setup
feature, provides the capabilities of an integrated voice /data workstation, specifically:
●
A 200-entry directory for automatic dialing of voice and data calls
●
VT100 terminal emulation with:
— file transfer with error checking
— unattended remote access operation with mail
●
On-line HELP that is accessible from almost anywhere within the program.
CAM runs on the AT&T PC6300 or compatible PC with at least 384K bytes of memory,
running MS-DOS Version 2.0 or later (when connected to System 25) or MS-DOS Version 3.1
or later (when connected to the STARLAN NETWORK).
The PC running CAM can be connected to System 25 in one of two ways (Figure 2-18):
1. By the PC’s RS-232 COM port to the System 25 via an ADU/DLC connection
2. As a STARLAN NETWORK workstation to the System 25 via the STARLAN
Interface CP.
CAM interfaces with System 25’s Command Mode to provide call control. The Third-Party
Call Setup feature provides voice call origination.
I
COMMUNICATIONS
ACCESS
MANAGER
EX-RS232C
DRIVER
RS-232-C
PORT
NAUCOM
DRIVER
STARLAN
NAU PORT
PERSONAL COMPUTER
Figure 2-18. Communications Access Manager Architecture
2-99
The STARLAN NETWORK communication driver (NAUCOM) is used before CAM is run on
a STARLAN NETWORK workstation. The Extended Device driver (CAM232) is used when
CAM is run on a PC connected to a System 25 DLC port.
The default screen presented when the user accesses CAM is the phone directory screen. The
phone screen is divided into five partitions:
●
Call Appearance area–provides call appearance for voice lines and data lines for
each extension shown. A call timer for each line is also displayed.
●
Feature Selection area—allows the user to select the voice or data line to be used,
initiate the call, and start the timer by function keys. Additional function keys may
be assigned to repertory dialing numbers.
●
Personal Directory area–holds a maximum of 200 entries, displayed 10 entries at a
time. Each screen is arranged alphabetically.
●
Message and Status area—contains prompts and messages for the user for the action
being executed.
●
Command Line area—contains commands available to the user for the area being
worked in. Commands are executed when the user presses the <ALT> key and the
first letter of the command.
The user may access the following commands:
— Data mode—provides the user with the terminal emulation screen.
— Edit—provides the user with the directory edit screen. Allows the user to
add, modify, and erase directory entries, group names, and feature functions.
Directory entries contain name, number with auto login script, comment,
group, and voice/data fields. Data entries also have parameter setup, a
screen with fields for speed, parity, permit mismatch, and number of bits.
The parameter setup allows speeds of 2400, 4800, 9600, and Autobaud.
Find–allows the user to search directory entries by name or group ID.
— Restore–displays the first 10 entries of the directory after a Find.
— Print–prints the entire contents of the directory on device LPT1.
— Setup—provides the user with the setup screen. Allows the user to view or
change the following options: communications port, printer port, speed,
parity, character size, return key code, autotimer, flow control, extension
numbers, remote access enable, remote access password, and remote greeting.
2-100
CONFERENCE
Description
Allows up to five parties, including the conference originator, to participate in a conference
call. Any voice terminal user, including operators at Direct Trunk Attendant Consoles and
Switched Loop Attendant Consoles (V2), can set up conferences. Refer to the description of
“(’conference Drop” for additional information on conferencing.
Multiline Voice Terminals:
Multiline voice terminal users can add another (external or internal) party to an existing call
by pressing the CONFERENCE button. This places the first party on Special Hold
(indicated by a broken flutter on the line appearance button) and the system selects an idle
SYSTEM ACCESS or LOOP (Switched Loop Console) button and provides system dial tone.
The user may dial the desired number or select another facility to dial the party to be
conferenced-in. Subsequently, pressing the held line button completes the conference. If the
facility to be added is busy or has invoked Exclusion, the conference will be denied.
Users can conference up to two outside facilities (trunks), and up to five parties in all. Any
attempt to add a sixth party will be denied, and the sixth party will be dropped. This limit is
for the conference as a whole. Other conference inside stations are also prohibited from
adding a third outside party or sixth party.
Single-Line Voice Terminals:
The single-line voice terminal user can establish a conference by momentarily pressing the
switchhook, which puts the first party on hold, receiving Recall Dial Tone, and then dialing a
second party. After connection to the second party, another press of the switchhook
establishes the conference. A third press of the switchhook will drop the second party,
restoring the original call. The user cannot put a conference that he/she has established on
hold. Other internal conferees (multiline or single-line) may then add additional parties to
the conference up to the five party/two outside line maximum.
Considerations
The Conference feature allows any voice terminal user to set up conference calls. Nonattendant users do not need the assistance of the attendant.
Waiting for an added party to answer and announcing the purpose of the call before adding
the party to the conference is good operating practice.
Interactions
●
Account Code Entry: If more than one user attempts to associate an account code
with a Conference Call, the first to activate the feature will prevail.
●
Attendant Message Waiting: Pressing the Attendant Message Waiting (ATT
MSG) button while on a conference call will be ignored.
2-101
●
Call Park: Call park may be used to place a conference on hold. Parked conference
calls do not return to the parking station (they remain parked).
If a 5-person conference is parked, the conferee who parked the conference will be
dropped when someone picks up the parked conference.
●
Exclusion: Exclusion may be invoked before establishing a conference. If it is
invoked after the conference is established, all internal conferees will be dropped.
●
Extended
Stations:
An Extended Station counts as one of the two allowable
outside lines on a conference call.
●
M u s i c - O n - H o l d : For Vl systems, an outside line placed on hold when
CONFERENCE is pressed will hear Music-On-Hold, if provided. For V2 systems,
Music-On-Hold may be enabled or disabled for “Special Hold” through a new System
Administration item. However, if the outside line is already part of a conference,
music is not heard.
●
Off-Premises Stations (OPS): For conference purposes, an OPS counts as one of
the two allowable outside lines.
●
Paging System Access: A paging zone may not be conference.
●
Trunk-To-Trunk Transfer: Trunk-to-trunk transfers may be set up using the
Conference feature. The conference must include an incoming trunk call on either a
ground start, DID, or tie trunk if it is to continue after all inside stations have
dropped off.
Administration Requirements
None
Hardware Requirements
None
2-102
CONFERENCE DROP
Description
Allows a voice terminal user, except for the attendant at a Switched Loop Attendant Console
(SLAC, V2 only), to selectively drop a previously added party from a conference call. At a
SLAC, the attendant can drop conferees only before they have been added to conference.
Multiline Voice Terminals (except SLAC):
On a multiline voice terminal, pressing the DROP button and then pressing the button
appearance of a conference party drops that party from the conference.
If a station called for a conference does not answer, the conferencing user should drop the
call by pressing and releasing the switchhook before returning to the conference. Otherwise,
the ringing line will be added to the conference.
Switched Loop Attendant Consoles:
Once a conference has been set up and all the parties can talk to each other, the SLAC
attendant cannot selectively drop a conferee. Individual members of the conference wishing
to drop out must hang up. However, while still setting up a conference, the attendant can
drop calls before they have been conferenced in, as follows:
●
A call to an inside party rings unanswered or returns busy tone-press DROP.
●
A call to an outside party rings unanswered or returns busy tone–press another
LOOP button or RELEASE or FORCED RELEASE.
●
A call to an inside or outside party is completed but the person cannot participate—
press another LOOP button or RELEASE or FORCED RELEASE.
It is good operating practice to wait for the called party to answer before adding the party to
a conference.
All Multiline Terminals:
If all System 25 stations hang up on a conference with two outside lines, the outside parties
will remain conference (until one of them hangs up) if at least one is on a ground start, tie,
or DID trunk. If not, the call will be terminated when the last inside user disconnects from
the conference.
Single-Line Voice Terminals:
A single-line terminal user, after having established a three-party conference, can drop the
second party and retain the first party by pressing the switchhook.
Considerations
Conference Drop allows users to conference lines appearing on their terminals and then
remove them from the conference when appropriate.
A user should only drop parties that they have added to a conference. If a user tries to drop
a party who previously added them to the conference, other parties may also be dropped.
2-103
Interactions
None
Administration Requirements
None
Hardware Requirements
None
2-104
DATA CALL SETUP
Allows a user to originate data calls from a data terminal. System 25 provides three methods
of data calling:
●
Dialing from a data terminal, which is described in the “Data Terminal Dialing”
feature description.
●
Setting up data calls from a voice terminal, which is described in the “Transfer to
Data” feature description.
●
V2 Only: Setting up data calls (or voice calls) for another terminal from a data
terminal, which is described in the “Third-Party Call Setup” feature description.
2-105
DATA SERVICES OVERVIEW
System 25’s data features provide switched data transmission at up to 19,200 bps (RS-232
interface ), and a 212A modern compatible conversion resource capable of handling data at
300 and 1200 bps.
The system provides switched connections between data endpoints. These endpoints include
data terminals, personal computers, multiport computers, and modems. Data endpoints are
either digital data endpoints or analog data endpoints.
Analog endpoints are connected to System 25 voice terminal or trunk port circuits through a
modem in the traditional manner. Digital endpoints are connected to System 25 data port
circuits on the TN726 Data Line CP. A data module (specifically, an Asynchronous Data
Unit - ADU) is required in place of the modem used with analog endpoints. Section 4 of this
manual shows the connections supported and required connecting equipment.
Data C alls can be set up between data endpoints. Analog to analog and digital to digital
connections are straightforward; calls between analog and digital endpoints are possible
only if the system is equipped with a conversion resource (TN758 Pooled Modem Circuit Pack
or external modem pool). System 25 data calls from analog endpoints (including those to
digital endpoints) are set up in the traditional manner. The calling party should follow the
procedures supplied with his/her modem. However, a Modem Request Code must be dialed
when calling a digital endpoint.
Call set-up from digital endpoints is facilitated by several data features: Command Mode,
Expert Mode, Data Terrminal Dialing, Modem Pooling, Third-Party Call Setup, and Transfer
To Data.
In the discussion that follows, it is important to understand the difference between analog
voice terminology and data terminology. Refer to the “Glossary” (Section 9).
The following provides a definition of a data call in terms of its contextual components. The
components are (1) data endpoints, (2) data endpoint states, (3) data call processing modes,
(4) connecting configurations, and (5) controlling features.
Data Endpoints
Data endpoints are composed of data equipment, a data module or modem, and a connection
to the switch via an analog or data port. A digital data endpoint is addressed by its Data
Dial Code (DDC). Analog data endpoints are addressed like other voice terminals, by their
PDCs. For the remainder of this description, data endpoints will refer to digital data
endpoints unless stated otherwise.
Several different categories of data endpoints are supported. The categories have been
divided into two general groups, those having a DTE type interface, which encompasses
almost all of the data terminal devices, and a group of DCE interface devices (primarily
modems ). The groups have then been divided into categories based upon their functional
attributes. However, it must be noted that within each category, control interfaces may
vary. The following describes the categories and attributes of each:
1. DTE Devices
This group of data endpoints have one thing in common: their interface
configuration (although RS-232 control signal utilization varies significantly from
terminal to terminal). Some data equipment do not use any RS-232 control signals;
these require only BA (Transmitted Data Ready-Tx), BB (Received Data Ready-Rx)
2-106
and AB (Signal Ground) to function, while others require more RS-232 control
signals to operate. An ADU (Figure 2-19) can send Data Terminal Ready (DTR)
from the data terminal to the Data Line circuit and the Data Line circuit can send
a control signal to the data terminal. The signals Data Set Ready (DSR), Clear To
Send (CTS), and Received Line Signal Detector (DCD) are all connected to the
control signal from the DLC in the ADU and available if required by the data
terminal. Refer to Section 5 (Port Specifications) for additional information.
TO/FROM
RS-232C
DTE
TD
RD
DTR
DCD
DSR
CTS
ADU
(DATA
MODULE)
(4-WIRE
CONNECTION )
TO/FROM
TN-726
DATA LINE
CIRCUIT PACK
* CD CONTROL SIGNAL CONNECTED
IN ADU TO PROVIDE CTS,
DSR, AND DCD TO RS-232C
DEVICE
Figure 2-19. Asynchronous Data Unit Interface Signals
The following categories are part of the DTE data endpoint group:
a. Data Terminal Without ASCII Keyboard
This category includes such devices as Fax machines, EBCDIC or Baudot
terminals, and receive only devices such as printers.
Once connected to an ADU and turned on, these data endpoints appear
on-line, available, and ready to enter the Setup mode on auto-answered
calls (modes are described below). These endpoints will display or print
information received after a valid connection has been established
without additional RS-232 control from that endpoint. Note that since
these endpoints cannot establish calls for themselves, they must either be
called by other endpoints or have calls established for them via the
Transfer to Data or Third-Party Call Setup (V2) feature.
b. ASCII Data Terminal With Keyboard
This category can be subdivided into two classes: (1) basic terminals, and
(2) intelligent, programmable data equipment such as personal
computers. Basic terminals appear to the data port to be on-line and
available whenever they are turned on, thus ready to enter either the
2-107
Data Terminal Dialing mode, or to enter the Setup mode on calls
originating from a voice terminal or on auto-answered calls.
In the case of most personal computers, a communications program must
be executed in order for it to communicate with its own RS-232 port or
built-in modem. Once the communications software is running, further
operation will be similar to that of the basic terminal.
c. Host Computer Endpoint
A host computer endpoint is very similar to a data endpoint with
keyboard except that the host has many ports and the interface is usually
capable of supporting multiple speeds and more of the RS-232 control
signals. Front-end communication software running in the host is
typically supplied by the computer vendor and is not designed t O support
the Data Terminal Dialing feature. Such software typically supports call
origination through Automatic Calling Units (ACUs), which are not
compatible with Data Terminal Dialing. Thus, the primary means of
communicating with the host is by calling from data terminals or
personal computers. Groups of host ports with matching characteristics
may be members of hunt groups (referred to as host port groups).
d. Analog Data Endpoint
Data endpoints with modems are referred to as analog data endpoints.
Modems connected via tip ring lines use PDCs as extension numbers
rather than Data Dial Codes (DDCs). Station-to-station data calls to (or
from ) this endpoint from (or to) digital endpoints require a modem
conversion resource to convert the endpoint’s analog data to digital
format. Calls from a digital endpoint to an analog data endpoint (i.e.,
calls to a PDC), will automatically have a conversion resource inserted in
the calling path. If the called (analog) endpoint should then invoke
Transfer To Data, the conversion resource will be released. Data calls
originating from an analog data endpoint must first enter a Modem
Request Code before addressing a digital data endpoint. This is required
because the system assumes that a call originating from a voice terminal
will invoke Transfer To Data. If the analog data endpoint is not going to
transfer to data, they must indicate this so that a conversion resource
will be included in the connection.
2. DCE Devices
This group of data endpoints consists primarily of modems. The modems are
connected to a data port from their RS-232 side. The data module must be
configured as a DTE interface to provide connectivity between the modem and a
data port. It is possible to simulate a DTE interface from an ADU data module
with a cross-over (“null modem”) cable. This group of endpoints is important for
users who provide their own modems, connected to dedicated trunks or private
lines, for internal modem pooling.
2-108
Data Endpoint States
The data endpoint may take on three states: (l) off-line, (.2) on-line (on-hook), and (3) on-line
(off-hook). Off-line is when a data terminal is out of service (turned off, disconnected, etc,).
The on-line (on-hook) state occurs when the terminal is turned on, is available to answer a
call, but is not on a data call. Finally, the on-line (off-hook) state is when the data endpoint
is actively on a data call.
Data Call Processing Modes
Data calls differ both in signaling and call setup from voice calls. For this reason, a unique
set of data call processing modes have been defined to support data call operation in a
manner consistent with the characteristics of data terminals.
a. Off-Line Mode
The data endpoint is considered to be in the Off-Line Mode whenever the data
endpoint’s DTR signal is inactive (e.g., “turned off”). The endpoint is considered
unavailable and calls to this endpoint will receive the “RINGING” message or
Ringback (indefinitely).
b. Idle Mode
The Idle Mode indicates that the data endpoint is in its on-line, on-hook state.
While idle, call processing will allow the endpoint to:
— Enter either Data Terminal Dialing mode to originate a data call, or enter
the Setup mode after a call is originated from a voice terminal (Transfer To
Data) or other data terminal (Third-Party Call Setup; V2 only).
— Autoanswer a data call and go into Setup mode.
The data endpoint remains in the Idle mode while the user is establishing a data
call from a voice terminal until Transfer To Data is activated.
c. Command Mode
Command Mode enables the Data Terminal Dialing feature, allows the user to view
and change (V2 only) associated data port options, and provides access to the
Third-party Call Setup feature (V2). Command Mode may be entered by going online and pressing Break or Break-Return.
d . Expert Mode is an enhancement to the Command Mode feature that provides an
alternative method of performing the full range of Command Mode functions. By
eliminating the display of menus and allowing multiple commands to be entered on
a single line, Expert Mode lends itself to computer-driven instructions. Individual
users who are very familiar with Command Mode operations may also find it
useful.
e. Data Terminal Dialing Mode
Data Terminal Dialing is a data feature accessed via Command Mode. It provides a
procedure to establish data calls without the use of a voice terminal. Data
Terminal Dialing supports both on-premises and off-premises data calls (with the
support of the System 25 Modem Pooling feature). Dialed digits are entered from
the data terminal keyboard or host computer (using a program compatible with
Data Terminal Dialing protocol). Call progress text messages are sent to the
terminal in place of call progress tones. Upon completion of digit entry, Data Call
2-109
Setup mode is entered.
f. Data Call Setup Mode
Data Call Setup Mode is a transitional state entered after Transfer To Data, Data
Terminal Dialing, or during auto-answer; it exists during the handshake between
data ports.
If the endpoints are compatible and handshaking is successful, a data connection is
established. If handshake failure occurs, the user is notified and the data endpoint
returns to the Idle mode. Successful handshake must occur within 15 seconds of
answer at the called data endpoint. This implies that the voice terminal user must
invoke Transfer To Data within 15 seconds after far-end answer. Similarly, if an
originating voice user calls a voice terminal and both users transfer to data, both
ends must transfer within the 15 second time limit.
If the data endpoint is optioned for Command Mode permission, the data endpoint
will receive call progress text messages while in the Data Call Setup mode.
g. Data Mode
Data Mode is first entered after successful completion of Data Call Setup.
Transparent communication between connected endpoints is provided in Data
Mode.
Connecting Configurations
Refer to “Connectivity” in Section 4 for data equipment connections.
Controlling Features
It is possible to originate data calls from a voice terminal with a Transfer to Data button or
from data endpoints that support Command Mode (i.e., ASCII data terminals with keyboards
and host computers). Several controlling features are provided to allow data endpoints and
voice terminals to set up data calls. The following briefly describes the Data Service
features used in controlling data calls:
a. Command Mode/Expert Mode
Command Mode and Expert Mode provide an interface to the Data Terminal
Dialing feature, the Third-Party Call Setup feature (V2), and permits users to
display and change (V2 only) data port options.
b. Data Terminal Dialing
Data Terminal Dialing provides call setup from terminals and host computers.
c. Transfer To Data
Transfer To Data is the preferred method of data call origination from multiline
voice terminals equipped with Transfer to Data (DATA) buttons and associated
digital data endpoints. The DATA button is associated by DDC with a near end
data endpoint. A unique DATA button must be provided for each DDC that the
voice terminal is capable of controlling. Associated with each DATA button is an
LED that reflects the status of data endpoints as follows:
●
Dark–Data endpoint is idle
●
Winking–Data endpoint is reserved (preindicated)
2-110
●
●
Flashing–Data endpoint is being alerted of an incoming call
On Steady–Data endpoint is either in the on-line (off-hook) state or is
reserved for another user and busy.
Refer to the following feature descriptions for additional information:
●
Command Mode
●
Data Call Setup
●
Data Terminal Dialing
●
Expert Mode
●
Modem Pooling
●
Third-Party Call Setup.
●
Transfer To Data
2-111
DATA TERMINAL DIALING
Description
When the user makes a data call from a data terminal, voice terminal dialing and call
progress tones are replaced by keyboard dialing and call progress messages. The message
“DIAL:” prompts the user to enter the called number from the keyboard, and “RINGING” or
“DIALING . . . COMPLETED” informs the user that the dialed number is being called.
Table 2-H provides a list of the call progress messages.
Note: The following dialing procedures assume that “Command Mode” is active. Refer
to the subsection on “Expert Mode” for an alternative method of dialing.
Dialed Characters:
In addition to digits and the “#” and “*” characters on the touch-tone pad, the dialed
number may contain the following special characters:
●
The characters “(” “)” “-” and SPACE may be used to improve legibility. These
characters are ignored.
●
The “%” or “,” characters may be used to cause a 1.5-second pause in dialing.
Multiple pause characters can be used to produce longer pauses.
●
The “$” (mark) character indicates that the remaining digits are for end-to-end
signaling.
●
UNDERSCORE or BACKSPACE characters may be used to correct previously typed
characters on the same line.
●
The “@” character may be used to delete the entire line and start over with a new
DIAL: prompt.
Each line of dialing information may contain up to 27 characters. Note that all of the
dialing information, including pauses and ignored characters, must be typed on a single line
following the DIAL: prompt and terminated by Return.
Dialing Correction:
The backspace character (BS key or Ctrl-H keys) or underscore (“_’”) may be used to cancel
the previously entered character. More than one entered character may be deleted by using
multiple backspace or underscore characters. The “G” character may be used to delete the
entire line of entered characters.
Pause:
To assist the completion of off-premises calls, the pause characters “%” or “,” may be used.
A pause character may be used to help ensure the receipt of dial tone before continuing to
dial. Each “%” or “,” causes a fixed delay of one and one-half (1.5) seconds. Pause
characters may be used consecutively if a longer pause is required. Note that System 25
cannot detect tones, such as a second dial tone for end-to-end signaling.
2-112
End-to-End Signaling:
Data connections to off-premises destinations require that a conversion resource (pooled
modem) be inserted into the connection. Occasionally it is necessary to send additional tones
to the remote endpoint after the connection is established to signal the remote equipment. A
“mark” character must be included on the DIAL: line to indicate to call processing that the
remaining digits are to be sent to the far end prior to insertion of the conversion resource
into the connection. The “mark” character marks the boundary between the digits dialed to
reach a distant endpoint, and the digits used by that distant endpoint after it has answered.
Pause characters may and usually should follow a “mark” character. An example using a
“mark” character and several pause characters is shown below. Dashes are included for
readability.
Examples of dialing are as follows:
●
DIAL: 3478
●
DIAL: 9-1-(201 )-946 -8123,, $,5678
●
DIAL: 9%946-8123%%$%5678%137%110
Call Disposition:
Call progress messages corresponding to call progress tones provided to voice terminals are
listed in Table 2-H. The message supplied (indicating reorder, busy, ringback) depends on
the disposition of the call.
1. When ringback is received the displayed message is “RINGING” (internal calls
only). For outside calls, the corresponding call progress message is
“DIALING . . . .“.
2. If the endpoint answers, the displayed message is “ANSWERED” (internal calls
only). Then, if the handshake succeeds, a data connection is established. For
outside calls, when the system has finished dialing, the message “COMPLETED” is
displayed.
3. If the handshake fails because a connection cannot be established between
endpoints (e.g., a port optioned at 9600 baud attempts to talk to a conversion
resource t h a t c a n o n l y talk at 300 or 1200 baud), the user receives
“INCOMPATIBLE FAR END,” “DISCONNECTED,’” and the data endpoint goes
on-hook.
If the far end does not answer, the caller must press Break to terminate the call attempt.
If the disposition of the call is such that TRY AGAIN or BUSY (indicating reorder or
intercept and busy respectively) is received, the switch sends “DISCONNECTED” to the data
terminal that goes on-hook.
Answering Endpoint:
When the dialed end point is alerted, the user receives “INCOMING CALL-”. (The called
terminal will auto-answer if it is turned on. ) If the handshake succeeds, a data connection is
established and the “CONNECTED” message is displayed if so optioned. If the handshake
fails, the user receives “INCOMPATIBLE FAR END, DISCONNECTED” and the data
endpoint goes on-hook.
2-113
TABLE 2-H. Call Progress Messages for Data Terminal Dialing
Displayed
Message
DIAL:
Application
Placing a call
Meaning
Equivalent to dial tone. Enter any
required facility number followed by the
dialed number and a RETURN.
Equivalent to Ringback Tone. Called
Placing a call
RINGING
number (far-end) is being signaled.
Provided on internal calls only.
Placing a call
Equivalent to busy tone. Called number is
BUSY
in use, or out of service. Provided on
internal calls only.
Notifies calling and called users that call
ANSWERED
Placing or
receiving a call
has been answered. Provided on internal
calls only.
Placing a call
Equivalent to Reorder Tone. System
TRY AGAIN
facilities are currently not available or
invalid number.
Equivalent to ringing.
INCOMING CALL-* Receiving a call
PLEASE ANSPlacing a call from Originating voice terminal user has
transferred call to data terminal using
a voice terminal
Transfer to Data.
Call
or call attempt is disconnected.
DISCONNECTED*
Call is terminated
Notifies user that the call connection is
Call is connected
CONNECTED,
established and what the baud rate is.
SPEED = NNNN
[Provided that “Connection indication”
(Data Port Action 77) is enabled.]
Notifies user that the handshake between
INCOMPATIBLE
Placing a call
data end points has failed
FAR END
Indicates off-premises call is being dialed
Placing a call
DIALING . .
and that dialing is completed.
COMPLETED
Call queued.
Placing a call
PLEASE WAIT
Bad dialed number
Placing a call
BAD NUMBER
No modem available for a call that
Placing a call
NO MODEM
requires one.
Specifies the session number (1) of the
Placing or
(V2) SESSION 1
terminating a call data call to the calling party
* Bell sounds when message is displayed.
2-114
Considerations
Data Terminal Dialing allows users to place data calls from their terminals using the Data
Terminal Dialing feature and allows users to review the options administered for their data
ports.
Interactions
●
●
●
End-To-End Signaling: (See above text)
Modem Pooling: Data calls between analog and digital endpoints require that a
conversion resource (TN758) be available. If one is not, the “NO MODEM” followed
by “TRY AGAIN” message will be displayed.
Speed Dialing: System Speed Dial codes can be dialed from data terminals.
Administration Requirements
Data Port: See the table of Permissible Data Port Options in the “Command Mode” feature
description.
Hardware Requirements
TN726 Data Line CP to support each digital endpoint.
TN758 Pooled Modem CP to support data calls between digital and analog endpoints.
2-115
DIAL PLAN
The dialing plan for System 25 is based on the concept that, whenever possible, calls should
be placed to individuals rather than to voice terminals. To implement this concept,
individuals are assigned Personal Dial Codes (PDCs) and are allowed to log in those PDCs at
other voice terminals. There are two types of PDCs: assigned and floating. An assigned
PDC is associated with each voice terminal. Floating PDCs are administered at the SAT and
may (optionally) be associated with the attendant position when not logged in.
Data extensions on System 25 are assigned Data Dial Codes (DDCs).
Dial Code Assignments
System 25 dial codes are as follows:
Assignable System 25 dial codes may have 1, 2, 3, or 4 digits. These include voice terminal
PDCs, data terminal DDCs, Direct Group Calling (DGC) Groups, Paging Access, Attendant
Call Park, Night Service, Modem Request, Automatic Route Selection Access, Facility Access
(trunk group), and Dictation System A C c eSS codes.
System 25 fixed dial codes are:
●
0—Attendant access
●
*1, *2, *3—Reserved for maintenance calls
●
*4—Activate Make-Busy for DGC group member
●
*5—Call
●
*6—Deactivate Make-Busy for DGC group member
●
*7 + 0—Group Call Pickup Answer
●
*7 + PDC—Directed Call Pickup Answer
●
*8 + PDC–Call Park Retrieval
●
*9–Camped on Call Retrieval
●
*0—Account Code Entry from single-line voice terminals
Park
●
** PDCPDC— Login PDC (Call Following)
●
**PDC0—Logout PDC (Call Following)
●
●
●
**0—Logout all PDCs (Call Following)
## PDC-Call Accountability
*—PAUSE character used in programmed numbers
●
#100-#189— System Speed Dial Codes
●
#190-#199—Virtual Facility Codes (V2)
●
#20-#26—Personal Speed Dial Codes
●
#3—Start end-to-end signaling in programmed numbers (V2 only)
●
#4—Activate
Program
mode
2-116
●
#5–Insert dialed digits here (in Virtual Facility numbers; (V2 only)
●
#–End of dialing
The maximum number of dial codes available for a System 25 is 600. In R1Vl, each assigned
code is allocated as a ten number block. In R1V2, each assigned code is stored individually
in memory.
The dial codes assigned in the system must be completely unambiguous. For example, a
dialing plan that contains the number “20” cannot contain the numbers “2,” “200-209,’” or
“2000-2099.”
PDC to Voice Terminal Association
During installation, each voice terminal is assigned one PDC that serves as its extension
number. These are termed assigned PDCs, and the associated terminals are called home
stations. Additional PDCs may be assigned in a system. These PDCs are termed “floating”
PDCs (FPDCs). At the customer’s option, floating PDCs may have the attendant position
assigned as their home station (i.e., calls to FPDCs will be directed to the attendant when
they are not logged in elsewhere). A maximum of 200 assigned PDCs and 300 FPDCs may be
allocated in a system.
Data Dial Codes (DDCs)
At the time of installation each digital data endpoint will be assigned a Data Dial Code
(DDC) that serves as its extension number. A maximum of 104 DDCs may be allocated in a
system.
Direct Inward Dial (DID) Number Assignments
Each DID number is associated with a unique PDC (floating or assigned), a DGC access
code, a DDC, or a pooled facility access code. The code associated with a DID number is the
last 2, 3, or 4 digits of the DID number. For example, the code associated with the DID
number “NXX-2157” will be 57, 157, or 2157.
All dial codes in the system that are associated with DID numbers should have the same
number of digits. However, there is no requirement that all PDCs, DDCs, DGC access codes,
or facility access codes be associated with DID numbers.
Voice Terminal Directed Features
Directed Night Service, DGC calls, Personal Line Calls, Manual Signaling, Station Message
Waiting, Automatic Intercom, and Outward/Toll Restriction are associated with specific
terminals (stations), not with PDCs. This means that these features do not move with a
PDC when it is logged in at another voice terminal. In R1V1 systems, Direct Station
Selection (DSS or FLEX DSS) calls do not receive Call Following treatment.
2-117
DICTATION SYSTEM ACCESS
Description
Permits access to, and control of customer-owned dictation equipment by voice terminal
users. Dictation systems may be connected either via single-line voice terminal ports (the
preferred method) or via auxiliary trunk ports (if the dictation equipment requires a
separate contact closure for proper operation). The dictation system is accessed by dialing a
PDC oppressing a DSS button.
Considerations
Allows users to access and control shared dictation equipment.
Interactions
Direct Inward Dialing (DID): A DID number may match the dictation system
access code. This allows an outside caller to access the dictation equipment.
●
Administration Requirements
System:
●
Requires a port assignment on a ZTN78 Tip Ring Line or TN742 Analog Line CP. A
port on a TN763 Auxiliary Trunk CP must be used if the dictation equipment
requires a separate contact closure for proper operation.
Voice Terminal: (Station Port)
●
Assign DSS access buttons as desired.
Hardware Requirements
Customer provided dictation equipment
Port on a ZTN78, TN742, or TN763 CP, as required.
closure, the TN763 must be used.
If the equipment requires a contact
Detailed connection information is provided in Figure 2-20. Descriptions of the SIP (Station
Interconnect Panel), TAE (Trunk Access Equipment), and associated cables and adapters are
provided under the heading “Connectivity” in Section 4 of this manual.
2-118
SYSTEM 25
CABINET
TN742
OR
ZTN78
l
PART OF
OCTOPUS
CABLE
— — —
C2
—
—
—
—
—
—
PART OF
SIP
SIP
ADAPT .
W1
B1
C5
DICTATION
EQUIPMENT
(NOTE)
LEGEND:
TN742 - ANALOG LINE CP
ZTN78 - TIP RING LINE CP
B 1 - TYPICAL-103A CONNECTING BLOCK*
C 2- OCTOPUS CABLE (WP90780) PEC 2720-05P
C 5- MODULAR CORD (D4BU-87)
W1 - 4 PAIR INSIDE WIRING CABLE*
* - FURNISHED BY INSTALLER
NOTE : IF CUSTOMER DICTATION EQUIPMENT REQUIRES A CONTACT
CLOSURE, A TN763 AUXILIARY TRUNK CP MUST BE USED. REFER
TO “PAGING SYSTEM USING AUXILIARY TRUNK CP” FOR TYPICAL
CONNECTION INFORMATION.
Figure 2-20. Dictation System Connections (FCC Registered)
2-119
DIGITAL TAPE UNIT (DTU)
The Digital Tape Unit (Figure 2-21 ) is a RS-232 device used to record administration
translations. The DTU does not encode the translations data as it records, nor does it
require decoding circuitry when playing back (restoring) recorded data. Data is recorded
and transmitted at 1200 bps.
The DTU requires 115V commercial power from a 3-wire grounded outlet. It should be
located on a desk or table top. The recorder is approximately 5 inches wide, 2 inches high,
and 10 inches long.
As shown in Figure 2-22, the DTU must be directly connected to port #3 on the Call
Processor (ZTN82 or ZTN128) CP. Remote and switched connections are not supported.
Maximum cabling distances are provided in Section 5, “Technical Specifications.”
115V AC
POWER
CABLE
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EIA
RS-232C
CONNECTOR
3-WAY POWER
DIGITAL
COUNTER
RESET
❘
RECORD
❘ ❘
PLAY
❘ ❘
❘
REWIND
F FWD
STOP/EJECT
❑
o
(TOP VIEW)
Figure 2-21. Digital Tape Unit
2-120
❘
❘ ❘
X
PAUSE
❘ ❘
SYSTEM 25
CABINET
ZTN82
OR ZTN128
CALL
PROCESSOR
PART OF
OCTOPUS
CABLE
— — —
— — —
C2
●
PART OF
SIP
Z210A
ADAPT.
C1
355A/AF
DIGITAL
TAPE UNIT
— — —
— — —
LEGEND:
C1 - MODULAR CORD (D8W-87) - PEC 2725-07G
C 2 - OCTOPUS CABLE (WP90780) - PEC 2720-05P
355A ADAPTER RS 232 PLUG TO MODULAR JACK - PEC 2750-A24
355AF ADAPTER RS 232 RECEPTACLE TO MODULAR JACK - PEC 2750-A25
Figure 2-22. Digital Tape Unit—On-Premises Direct Connections (Sharing Same
AC Outlet)
2-121
DIRECT GROUP CALLING (DGC)
Description
AllO w S incoming calls to be directed to a specific group of terminals. Up to 32 DGC groups,
each including up to 10 members, may be set up. A terminal can be in only one DGC group.
Incoming calls on any trunk can be directed to a DGC group. These trunks can also be used
for outgoing calls. This feature is referred to as “Incoming Calls Group” in the User Guides
for System 25 voice terminals.
DGC groups can be administered as Call Coverage groups in R1V2 systems. Refer to the
“Call Coverage–Group” feature description for details.
Calls to a DGC group hunt in a circular manner, starting at the terminal following the last
one to receive ringing (whether answered or not), and will ring at the next idle terminal in
the group. On multiline voice terminals the calls arrive on a SYSTEM ACCESS button
(LOOP button on a Switched Loop Attendant Console; V2 only).
If all group members are busy (off-hook), an outside call is queued and the caller receives
ringback tone. If the system includes a delay announcement, it is played after a specified
number of rings. The caller is subsequently put on hold (in queue) and will receive MusicOn-Hold if available. If the system is not equipped with a delay announcement, the call will
begin to ring at all line appearances after the specified interval.
An inside caller dials a DGC access code to reach a DGC group. If all members of the group
are busy, the call will not go into a queue and the caller will receive Busy Tone.
Once the call begins to ring at a group member’s station, it will not receive announcement
service or ring at a line appearance. For this reason, it is important that DGC members log
out (as described below) when they will be away from their desks.
For R1V2 systems, the attendant can camp-on multiple outside (trunk) calls when all
members of the group are busy. Group members do not receive camp-on indication. The
camped-on calls will be queued, and are eligible for the DGC delay announcement. If no
delay announcement is available, the calls will return to the attendant console after a
specified number of rings.
For R1V1 systems, the attendant can camp-on only one outside (trunk) call when all
members of the group are busy (assuming Attendant Camp-On is activated). Group
members do not receive camp-on indication.
DGC group members may withdraw from the group by going off-hook and dialing *4. To
reenter the group, the member goes off-hook and dials *6.
An off-hook multiline terminal or attendant console (even if busy on only one SYSTEM
ACCESS or LOOP button) appears busy to DGC calls. However, terminals other than the
SLAC (V2) may receive other (non-DGC) calls while active on a DGC call.
Direct Group Calling groups may be used for data applications to access host ports and the
STARLAN Interface CP (V2). The System Administrator may disable queuing for data DGC
groups, if desired (V2). Delay announcements and music-on-hold are not provided for data
groups.
2-122
Considerations
DGC groups are particularly useful when the answering group receives a high volume of
calls. Call completion time is minimized and attendant assistance is not required.
Any number of outside trunks may be administered to feed into a DGC group. A trunk may
feed only one DGC group.
Interactions
●
Attendant Camp-On: For V1 systems only: if the attendant attempts to camp-on a
second call to a DGC group, it is immediately returned on the console.
●
Attendant Direct Extension Selection: When all stations in a DGC group are
busy, the status LED on the Selector Console lights steadily.
●
Call Coverage: When a call rings at DGC station that has Call Coverage, the call
will receive that station’s coverage. Calls directed to a busy DGC group do not
receive call coverage. Instead, after a predefined number of rings, the call will be
transferred to a delay announcement (if provided), or ringing will be transferred to
all button appearances of the line and the SLAC queue (if trunk has ringing enabled;
V2).
●
Call Pickup: A DGC group member can also be a member of a Call Pickup group.
●
Direct Group Calling Delay Announcement: Provides a recorded announcement
to an outside caller who has been placed in queue for a DGC group.
●
Direct Inward Dialing: An incoming DID call may match a DGC group access
code.
●
Direct Station Selection (DSS): A DSS button can be assigned to a DGC group.
The associated LED lights steadily when all stations in the group are busy.
●
Modem Pooling: Modem Pooling supports calls to data endpoints that are part of a
DGC group. While an incoming data call is in a DGC group queue, the caller receives
ringing. The conversion resource is inserted if the call is completed to a digital
endpoint.
●
Personal Lines: An outside line directed to a DGC group can be assigned button
appearances in addition to the DGC group assignment. When an incoming call is
ringing at a DGC group, the status LED on the voice terminal button appearance
lights steadily, indicating that the line is busy. If the call goes unanswered for a
pre-determined number of rings (and no delay announcement is provided), ringing
will be transferred to all button appearances of the line and the status LED will
flash.
●
Station Message Detail Recording (SMDR): For an incoming call to a DGC
group that was connected to an announcement but was never answered, “O” will be
reported in the “STN” field of the call record. If the call was answered by a station
after receiving the announcement, that station will be listed in the “STN” field.
●
Tie Trunks: Calls to a busy DGC group via auto-in tie trunks will be queued, but
will not receive a delay announcement. Calls to a busy DGC group via dial-in tie
trunks will not be queued; these callers will receive Busy Tone.
2-123
●
Transfer: For V2 systems only: internal stations can transfer outside (trunk) calls
to a busy DGC group. The transferred call will be treated as any other trunk call to
a busy DGC group.
Administration Requirements
●
Trunks - Assign trunks to DGC Group
●
Trunks (V2) - Assign trunks to ring in SLAC queue
●
System: Number of rings before DGC calls are transferred to announcement or begin
ringing at button appearances or SLAC queue (V2)
●
Direct Group Calling: Assign DGC access code, Add/delete DGC members,
Enable/disable queuing for data DGC groups (V2).
Hardware Requirements
None
2-124
DIRECT GROUP CALLING DELAY ANNOUNCEMENT
Description
Provides a recorded announcement to an outside (trunk) caller who has been placed in queue
for a DGC Group.
When all members in the group are busy (off-hook), the call will be queued for DGC service
and the calling party will receive ringback tone. Note that no incoming call indication
(ringing) is provided to the DGC group members at this point. After a specified number of
rings (administrable) a recorded announcement will be played to the calling party without
disturbing his or her position in queue. The caller is subsequently placed on hold and will
receive music if available.
Once a call begins to ring at a DGC station, the call is no longer eligible for delay
announcement service. The call will then ring until answered, covered, picked up or
abandoned.
Considerations
DGC Delay Announcements provide the calling party with a message that acknowledges
their call and assures them that their call will be handled in an orderly way.
Interactions
●
Tie Trunks: Calls to busy DGC groups via auto-in tie trunks w-ill he queued, but
will not receive the delay announcement. Calls to busy DGC groups via dial-in tie
trunks will not be queued (and, hence, will not receive the delay announcement).
Administration Requirements
The DGC announcement device requires a port assignment on a ZTN78 Tip Ring Line or
TN742 Analog Line CP. Only one DGC Delay Announcement may be assigned in the system.
Callers to all DGC groups receive the same message.
Hardware Requirements
The AT&T Answer-Record 2500 or Code-A-Phone 2540 may be used as the announcement
device. The announcement device must automatically hang up at the end of each call so that
the incoming call can be returned to the DGC queue.
The equipment requires a port on a ZTN78 Tip Ring Line (or TN742 Analog Line) CP. The
system supports one DGC delay announcement.
For Music-On-Hold hardware information, refer to the “Music-On-Hold” feature description.
Detailed connection information is provided in Figure 2-23.
Descriptions of the SIP (Station Interconnect Panel), TAE (Trunk Access Equipment), and
associated cables and adapters, as shown on the figures, are provided under the heading
“Connectivity” in Section 4.
2-125
SYSTEM 25
CABINET
TN742
OR
ZTN78
PART OF
OCTOPUS
CABLE
— — —
C2
PART OF
SIP
●
—
—
—
—
—
—
SIP
ADAPT .
W1
B1
C5
DELAY
ANNOUNCEMENT
EQUIPMENT
LEGEND:
TN742 - ANALOG LINE CP
ZTN78 - TIP RING CP
B 1- TYPICAL-103A CONNECTING BLOCK*
C 2- OCTOPUS CABLE (WP90780) PEC 2720-05P
C 5- MODULAR CORD (D4BU-87)
W1 - 4 PAIR INSIDE WIRING CABLE*
* - FURNISHED BY INSTALLER
Figure 2-23. Delay Announcement Equipment Connections (FCC Registered)
2-126
DIRECT INWARD DIALING (DID)
Description
Allows incoming calls to reach specific individuals or facilities in the system without
attendant assistance.
System 25 customers reserve blocks of DID numbers from the CO. The DID numbers may
correspond to a PDC FPDC, DGC access code, DDC, or any facility with an access code such,
as a pooled facility or a paging zone.
The system is capable of receiving either 1, 2, 3, or 4 digits over its DID trunks. The number
of digits received on a specific DID trunk will be constant for that trunk; however, different
DID trunks may receive different numbers of digits. The system is capable of receiving U P
to four digits and then ignoring leading digits as specified to match against system dial
codes. For example, the dial code matching DID number NXX-2157 can be 57, 157, or 2157.
If the System 25 is administered to match on more digits than are received from the Central
Office (CO), the additional leading digits are taken from the 4-digit trunk number. For
example: if a call comes in on DID trunk number 1234, the CO sends two digits (77) over this
trunk to identify the recipient, and System 25 is administered to match on three digits, then
the call will be routed to dial code 277.
Incoming DID numbers that don’t match any valid dial code may optionally be directed to
the Attendant Console or to Reorder Tone.
If the DID number received is a valid dial code, the caller is provided either Ringback Tone,
Busy Tone, or the tone from a pooled facility (e.g., Dial Tone) as soon as addressing is
completed. Busy Tone is provided if and only if the call cannot be completed to the intended
voice terminal and cannot be provided coverage.
DID calls appear at System Access buttons on multiline voice terminals (they do not have
other button appearances). These calls can be transferred to a covering station, answered
via Call Pickup, directed to a DGC Group, or given “call hunting” or “call following”
treatment. A DID call is not automatically covered on the Attendant Console.
DID trunks may utilize DID Immediate Start or Wink Start protocols. Refer to Section 9,
Glossary, for a brief description of each of these trunk types.
Considerations
Direct Inward Dialing frees the attendant from handling certain incoming calls.
Interactions
●
Attendant Camp-On: DID calls are not provided Attendant Camp-On treatment.
They will not appear on the Direct Trunk Attendant Console Return-On-Busy or
Return-On-Don’t-Answer buttons or on the Switched Loop Attendant Console Loop
buttons unless they are first answered at the attendant position and are subsequently
extended by the attendant.
●
Attendant Direct Extension Selection: Selector Console LEDs respond to DID
calls just as they do for other outside calls. When a user answers a DID call, the
associated LED on the Selector Console will light steadily. When a DID call arrives
at the attendant position for coverage, the LED associated with the call coverage
2-127
sender will flash and will then go dark when the call is answered. However, if the
call is placed directly to the attendant position or is forwarded to the position and
thereby arrives on a System Access button or a Loop button (e.g., if a DID PDC is
logged-in at the attendant position), then no LED indications on the Selector Console
will be provided. If a DID call is directed to the answering position and is
subsequently extended to a station, then the LED on the Selector Console associated
with the station will flash if the call returns to the answering position. The LED
will light steadily if the call is answered by the station.
●
Call Coverage: DID calls receive standard call coverage treatment.
●
Conference: For conference purposes, DID calls count as one of two allowable
outside parties.
●
Dictation System Access: A DID number may be associated with the dictation
system access code. This allows an outside caller to access the dictation equipment.
●
Direct Group Calling: A DID call will be directed to a DGC group if the DID
number matches the DGC group access code.
●
Night Service: DID calls do not receive Night Service treatment. A DID call will
ring at the appropriate station whether Night Service is activated or not.
●
Off-Premises Stations (OPS): DID calls can be directed to OPS.
●
Paging System Access: A DID call may access a paging zone. This allows the
user to dial in and utilize the Paging feature. Dial restricting the paging code will
block this interaction.
●
Personal Dial Codes: DID calls will be redirected to PDCs logged in at other
terminals in the system. DID calls to an unassigned PDC or a FPDC that is not
logged-in will be either redirected to the attendant or receive Reorder Tone.
●
Pooled Facility Access: Access to pooled facilities via DID is permitted. This
includes access to WATS, FX, tie trunks, private lines, dictation equipment, and
paging systems. This access is provided by selecting facility access codes so that they
will match DID numbers.
●
Station Message Detail Recording (SMDR): For V2 systems, only one SMDR
record is produced if an outgoing call is originated by a DID trunk. The STN field
will contain 0000, the FAC field will contain the facility access code of the trunk
group used to complete the call, and the CALLED NUMBER field will contain the
called number.
Administration Requirements
System:
●
Send misdirected DID calls to the Attendant Console (Yes, No) - Default = Yes.
●
Set number of DID digits matched against dial codes (2-4, None) - Default = 3.
Trunk Port:
●
DID trunk type (Immediate Dial, Wink Start)
●
Number of digits to be received from CO on this trunk - Default = 3.
2-128
Hardware Requirements:
Each DID trunk requires a port on a TN753 DID Trunks CP.
2-129
DIRECT STATION SELECTION (DSS)
Description
Allows one-button access to another voice terminal, a paging zone, or a DGC Group. This
feature requires a button assignment on a multiline voice terminal.
There are two types of DSS buttons. Numbers stored on DSS buttons (maximum of four
digits) are programmed at the SAT; numbers stored on Flexible DSS buttons (maximum of
four digits) are programmed at the voice terminal. The procedure for programming FLEX
DSS buttons is provided in the “Program’’ feature description.
To use DSS, the user presses DSS or FLEX DSS and goes off-hook. The caller hears
Ringback Tone. DSS calls to a multiline voice terminal are received on a System Access
button. The DSS status LED is lighted steadily at the calling station.
The DSS status LED is lighted whenever the pointed-to station is off-hook. The user may
press DSS and remain on-hook to receive the busy-to-idle reminder. The user’s voice
terminal will ring once when the other party hangs up; lifting the handset will automatically
place the call.
When Prime Line Preference is assigned to a DSS button, the button must be pressed to
invoke the busy-to-idle reminder, even though the I-Use LED is lighted.
Access to Paging Zones and DGC Groups:
Access is provided to an individual Paging Zone or to all paging zones or to a DGC group.
The status and busy-to-idle reminder indication described above also apply to DGC groups
with the understanding that they are busy if all the members are busy.
Considerations
Direct Station Selection differs from Automatic Intercom in that it provides one-button
access from one voice terminal to another (one-way only), while Automatic Intercom
provides similar access for each voice terminal (two-way) and must be assigned between two
multiline voice terminals. A DSS button may point to a single-line station; an Automatic
Intercom button may not. DSS calls receive call coverage, Automatic Intercom calls do not.
Interactions
●
Call Coverage: DSS calls placed to an individual with Call Coverage will receive
standard call coverage treatment.
●
Call Following: For Vl systems, DSS calls do not follow users who log in at other
voice terminals. For V2 systems, DSS calls do receive Call Forwarding treatment.
●
Direct Group Calling: A DSS button can be assigned to a DGC group. The
associated LED lights steadily when all stations in the group are busy.
●
Last Number Dialed (V2): Numbers called by pressing FLEX DSS or DSS buttons
are not saved by Last Number Dialed and cannot be recalled with that feature.
2-130
●
Line Selection (Prime Line Preference): When Prime Line Preference is
assigned to a DSS button, the button must be pressed to invoke the busy-to-idle
reminder, even though its red I-Use LED is lighted.
●
Personal Dial Code (PDC): An attempt to program a FPDC on a FLEX DSS
button (rather than a PDC) results in Reorder Tone.
●
Pooled Facilities: A pooled facility access code may be stored on a FLEX DSS
button (but not on a DSS button). If so, the button will function very much like a
Direct Facility Access button, with the capability of receiving a busy-to-idle reminder
for the pooled facility. However, this button will not allow access to a dial-restricted
facility.
Administration Requirements
Assign DSS and/or FLEX DSS buttons at voice terminal.
Hardware Requirements:
None
2-131
DISTINCTIVE RINGING
Description
Allows users to distinguish between different types of incoming calls.
Users can receive the following types of ringing:
●
A repeated two-burst tone indicates an outside call or a call extended by the
attendant. The two-burst tone pattern is: 0.4 seconds on, 0.2 seconds off, 0.6 seconds
on, and 4.0 seconds off.
●
A repeated one-burst pattern indicates a call from an internal user. The tone is one
second on and three seconds off for multiline voice terminals, and 1.2 seconds on and
4 seconds off for single-line voice terminals.
●
An
“abbreviated
alerting” signal on incoming calls to off-hook multiline voice
terminals. These calls will ring just once. The status LED associated with the
incoming call will continue to flash after the abbreviated ring. The user may place
their current call on hold and answer the incoming call if desired.
●
A single short beep at a voice terminal equipped with the Hands-Free Answer
feature indicates that an incoming inside call has been answered by the terminal.
Depending on the status of the terminal’s HFAI controls, the user can talk with the
caller without lifting the handset.
●
Priority ringing (V2) is a repeated pattern of two short rings followed by one long
ring. It indicates that a data terminal has used the Third-Party Call Setup feature to
originate a voice call from the voice terminal where this ringing is heard.
Considerations
Distinctive Ringing enables a user to handle each call in an appropriate manner.
Abbreviated alerting alerts the called party to an incoming call but does not provide the
continued distraction of ringing.
For V2 systems only: distinctive ringing is not available at Extended Stations; All incoming
calls are signaled by standard one-burst ringing, repeated.
Interactions
●
Call Coverage: Covering stations receive distinctive ringing, depending on the
origin of the call receiving coverage.
Administration Requirements
None
Hardware Requirements
None
2-132
END-TO-END SIGNALING
Description
Allows multiline voice terminals to send touch-tone (DTMF) signals over the DDD network
and allows single-line and multiline users to send touch-tones over dial pulse trunks.
The 7300H series voice terminals do not generate touch-tones when a dial pad button is
pressed. The End-To-End Signaling feature provides for the conversion of signals generated
by these terminals to touch-tones.
Dialed numbers are toned out for a default duration of 60 ms followed by 60 ms of silence
(administerable).
When using dial pulse trunks, End-To-End signaling is invoked by dialing “#” after the last
digit of the called number or waiting for about 10 seconds after dialing the last digit (see the
Interdigit Timeouts feature description). All subsequent dial pad button presses generate
touch-tones on the outside line.
Considerations
End-To-End Signaling permits stations to access network services that require touch-tone
signals.
Interactions
●
Command Mode And Data Terminal Dialing: Occasionally it is necessary to
send additional tones to the remote endpoint after a data connection has been
established. A mark character “$” is embedded in the dialing sequence to indicate to
call processing that additional tones must be sent prior to insertion of a conversion
resource (pooled modem) into the connection. The mark character “$” is used to
indicate that all the following digits are for end-to-end signaling. This character is
used to mark the boundary between the digits dialed to reach the distant endpoint
and the digits used by the distant endpoint after it answers.
●
Repertory Dialing: Repertory Dialing can be programmed on the 7300H series
voice terminals. End-To-End Signaling works properly with this feature.
●
Speed Dialing: (For V2 systems only) #3 must be stored to start End-to-End
Signaling.
●
Virtual Facilities (V2): #3 must be stored to start End-to-End Signaling.
Administration Requirements
None
Hardware Requirements
None
2-133
EXCLUSION
Description
Allows multiline voice terminal users to keep other users with appearances of the same
Personal Line from listening in on their calls. Exclusion allows users to exclude the
attendant and other stations from an existing or held outside call, or to drop other System
25 users from a call.
The Exclusion button status and I-use LEDs are lighted steadily when the feature is invoked.
When an excluded call is placed on hold, the Exclusion button’s I-Use LED goes dark and the
status LED winks with the LED of the held line.
Exclusion can be applied to only one line at a time. Once Exclusion is invoked on a call it
will remain active until the user either presses the button a second time or goes on-hook.
Considerations
Exclusion allows the sharing of a Personal Line by several users while retaining privacy.
Note that all inside calls are automatically private.
Interactions
●
Automatic Intercom: Any attempt to activate Exclusion while active on an
Automatic Intercom call will drop the other party.
●
Call Coverage: If a call coverage receiver invokes Exclusion after answering a
coverage call, all other terminals (including the attendant and the covered station)
are excluded. The covered user cannot enter the call until Exclusion is pressed a
second time by the covering user.
●
Conference: When Exclusion is invoked, all other inside parties will be dropped. If
a private conference including inside parties is desired, the user should activate
Exclusion first and then set up the conference.
●
Hold: A call can be placed on hold after Exclusion is invoked. The I-Use LED will go
dark; the status LED of the line appearance button and the Exclusion button will
wink.
Administration Requirements
Voice Terminal (Station Port):
. Assign Exclusion button.
Hardware Requirements
None
2-134
EXPERT MODE (V2)
Description
Expert Mode is an enhancement to the Command Mode feature that provides an alternative
method of performing the full range of Command Mode functions. By eliminating the
display of menus and allowing multiple commands to be entered on a single line, Expert
Mode lends itself to computer-driven instructions. Individual users who are very familiar
with Command Mode operations may also find it useful.
When Expert Mode is activated, a system administrable prompt is displayed that can consist
of up to nine characters (the quote character and RETURN are not allowed). Command: is
the system default prompt. As with dialing in Command Mode, the ASCII characters
backspace (BS or CTRL-H) or underscore (_) may be used to cancel a previously entered
character. When in Expert Mode, each line must be terminated with a keyboard RETURN.
Users of Expert Mode must follow the exact tree structure of Command Mode (both up and
down the menu tree) as shown in Figure 2-51. However, instead of moving one level at a
time, Expert Mode allows the user to move up or down several menu levels at once. This can
be accomplished by entering, on a single command line, the capitalized letters that define the
sequence of menu selections desired. For example, to change data port parity from the tree’s
entry level, the user types OCPE and presses RETURN. This requests that parity be set to
“even”, but does NOT enable the change. To enable this change (see Figure 2-51), the user
must now type XE and press RETURN.
Activating Expert Mode
A user can move back and forth between Command Mode and Expert Mode by typing “!”
(exclamation mark). For ports on a Data Line circuit pack (Data Line Card, DLC), either
Command Mode or Expert Mode is presented at the start of a new session, depending upon
the port’s setting at the termination of the previous session. Thus, if a data session ends in
Expert Mode, the next session will begin in Expert Mode. However, calls from an AT&T
STARLAN NETWORK to System 25 will always begin a new session in Command Mode.
An alternative command, “>”, can be used to guarantee entry into Expert Mode. Conversely,
guaranteed entry into Command Mode can be accomplished with the command “>!” followed
by RETURN. These commands are especially useful for computer-driven DLC endpoints
that might otherwise have difficulty detecting whether a new session had been started in
Expert Mode or Command Mode.
Making a Data Call
To make a data call from the entry level (see Figure 2-51), the user enters “D” following the
system prompt and then the data endpoint number. For example:
Command: D9,5553822
Note: “Command:” on the above line is the default system prompt while in Expert
Mode. The user enters all data following the prompt.
2-135
If the user enters “D” and then a RETURN, the system will prompt for the data endpoint
number as follows:
Command: D
DIAL:
The user must then enter the digits required to complete the call.
Activating the Third-Party Call Setup Feature
The following provides an abbreviated method of using the Third-Party Call Setup feature
while in the Expert Mode. A complete description of this feature is provided later in this
manual.
To activate the Third-Party Call Setup feature and place a call, the user enters numbers
using the following format:
Command: V{Destination} F {Source}
The V on the above command line provides access to <Voice call> from the Command
Mode entry level menu. The balance of the dialed number is composed of destination a n d
source numbers, as described in the Third-Party Call Setup feature description.
User Changeable Options
Refer to the User Changeable Options feature (discussed later) for a detailed description of
the feature. The menus selected in the following discussion are shown in Figure 2-51.
To view the current Options Table (starting at the Command Mode entry level), the user
simply enters “OV” following the system prompt, as follows:
Command: OV
To change the current Options Table (starting at the entry level), the user enters “OC” as
follows:
Command: OC
E n t e r i n g “ O C)’ places the user at the Change Options level. At this point the user may
change options by entering the appropriate letter to indicate the required option (S for
Speed, P for Parity, M for Mismatch, etc.) followed by the desired setting(s). Only one
Option is allowed per line. If more than one setting is selected for an Option that can only
accept one setting, call processing recognizes only the last entry.
Examples:
Command: S +1200 -300 +4800 Add 1200 and 4800 baud to the
available speeds, remove 3 0 0
baud
2-136
Command: PE
Change Parity to Even
Command: MY
Change Mismatch to Yes
If the user enters an invalid Option or setting, the system responds with I N V A L I D
OPTION and the entry is ignored.
Once all changes have been entered, the user enters XE to enable the options.
Considerations
Expert Mode is primarily for use by computer-driven endpoints that can store command
sequences for automated use. However, a user experienced in accessing Command Mode
menus may find Expert Mode to be a faster alternative when operating at slower speeds,
since the time required to display each menu and to input separate commands is essentially
deleted.
Interactions
●
Command Mode
Refer to the Command Mode feature description for a detailed description of
Command Mode and of the various menu items.
Administration Requirements
The data port associated with a data terminal can be administered to allow the user to
change options when in Command or Expert Mode. Otherwise, the user may view the
current options but not change them.
The default prompt for
administration.
E x p e r t M o d e ( C o m m a n d : ) may be changed via system
Hardware Requirements
None
2-137
EXTENDED
STATIONS
Description
Allows single-line voice terminals to be located at distances greater than 2000 feet from the
system cabinets.
Extended stations have the same feature capabilities other voice terminals. These stations
count as an outside party on conference calls.
Transmit and receive levels are increased at extended stations for conferencing.
Considerations
A single-line voice terminal must be administered as an extended station before this feature
is activated.
In V2, extended stations will always receive standard (that is, single) ring for calls; System
25 will not send distinctive ringing.
Interactions
●
Call Park: (For Vl only) unextended station cannot pick up parked calls.
●
Conference: An Extended Station counts as one of the two outside parties allowed
on conference calls.
Administration Requirements
Single-Line Voice Terminals (Station Port)
●
Assign port on Analog Line (TN742) CP
●
Make This An Extended Station (Yes, No) - Default = No.
Hardware Requirements
The Extended Station must be a single-line voice terminal. Requires a port on a TN742
Analog Line CP.
2-138
EXTERNAL ALERTS
Description
Provides standard station ringing at a location away from the called station. This feature
can be used to activate an external alerting device such as a bell.
External Alerts supports the Trunk-Answer-from-Any-Station (TAAS) form of Night
Service.
The feature can be used in conjunction with voice terminals located in noisy environments
and large areas such as warehouses, etc. The alerting device is activated whenever the
associated station is alerted.
A Supplemental Alert Adapter installed on a hybrid station allows the terminal user to
transfer incoming ringing to an alerting device located in some remote area. When the user
goes to the area, the alerting device rings for incoming calls to the user’s normal station.
Considerations
External Alerting enhances user ability to recognize incoming calls. Noisy environments,
large areas, and outside locations are candidates for external alerting devices.
Interactions
●
Manual Signaling: Manual Signaling will not activate an external alerting device.
●
Night Service: When the system is in Trunk-Answer-from-Any -Station (TAAS)
Night Service mode, an incoming attendant-seeking call will activate the Night
Service alerting device.
●
Power Failure Transfer: When the system is in the power failure transfer mode,
the external alerting devices are disabled.
Administration Requirements
Station Port
●
An external alert operating as the endpoint device on a station line requires a port
assignment on a ZTN78 Tip Ring Line or TN742 Analog Line CP. (Specify special
feature port type = 253. ) An xternal alerting device controlled from a Supplemental
Alert Adapter operates on the same line as the associated terminal and requires no
additional port assignment.
●
Specify the PDC of the associated station (or 0 if alert is used with TAAS Night
Service).
Hardware Requirements
Order line-activated alerting devices (e.g., bells) as required.
An alerting device operating on a line separate from a terminal requires a port interface on
a ZTN78 or TN742. Refer to Figure 2-24 for connection information.
2-139
Order the Supplemental Alert Adapter (PEC 2301-SAA) for controlling a remote alerting
device. A Supplemental Alert Adapter is installed in the line between the port CP and the
user’s MERLIN terminal. The line requires a ZTN79 ATL Line CP. Figure 2-25 contains
connection details.
Descriptions of the SIP (Station Interconnect Panel), TAE (Trunk Access Equipment), and
associated cables and adapters are provided under the heading “Connectivity” in Section 4.
SYSTEM 25
CABINET
PART OF
OCTOPUS
CABLE
— — —
C2
TN742
OR
ZTN78
●
PART OF
SIP
SIP
ADAPT .
—
—
—
—
—
—
W1
B1
C5
LEGEND:
TN742
ZTN78
B1
C2
C5
R1
WI
*
-
ANALOG LINE CP
TIP RING CP
TYPICAL-103A CONNECTING BLOCK*
OCTOPUS CABLE (WP90780) - PEC 2720-05P
MODULAR CORD (D4BU-87)
E1CM-50 RINGER OR EQUIVALENT PEC-31019
4 PAIR INSIDE WIRING CABLE*
FURNISHED BY INSTALLER
Figure 2-24. External Alert Connections
2-140
ALERTING
DEVICE R1
SYSTEM 25
CABINET
ZTN79
ATL
LINE CP
PART OF
OCTOPUS
CABLE
— — —
— — —
C2
●
PART OF
SIP
SIP
ADAPT .
W1
B1
C1
TERMINAL
T1
— — —
— — —
LEGEND:
A1 - SUPPLEMENTAL ALERT ADAPTER (2301-SAA)
B 1- TYPICAL – 103A CONNECTING BLOCK
C 1- MODULAR CORD (D8W-87)
C 2 - OCTOPUS CABLE (WP90780)
C8 - MODULAR CORD (D4BU-87)
S1 - EXTERNAL ALERT
T1 - HYBRID TYPE TERMINAL
W1 - 4 PAIR INSIDE WIRING CABLE
Figure 2-25. Supplemental Alert Adapter Connections
2-141
HANDS-FREE ANSWER ON INTERCOM (HFAI)
Description
Allows the following voice terminals to provide hands-free answer service on eligible
incoming calls; each terminal must have Automatic Answer (AUTO ANS) assigned to a
flexible button.
●
BIS (7305H03B and 7305H04C) and HFAI (7309H01A); these sets provide full service
without requiring adjuncts.
●
10-Button (7303H01B), 34-Button (7305H01B), and 34-Button Deluxe (7305H02B)
equipped with a Hands-Free Unit (HFU—a 102A Speakerphone); these arrangements
provide full HFAI service.
●
5-Button (7302H01C) and the 10- and 34-Button sets listed above, not equipped with
an HFU; these arrangements allow callers to “voice announce” their calls, but the
terminal user must use the handset to reply.
Calls Eligible for Hands-Free Service:
The following types of calls are eligible for HFAI service:
●
Inside calls (that is, calls from one System 25 set to another System 25 set using a
System Access, Loop, DSS, or Auto Intercom button).
●
V1 Only: Incoming calls extended by the attendant are eligible for HFAI service at
BIS sets but not at HFAI sets. However, HFAI sets may be translated as BIS sets
and will then provide HFAI service on attendant extended calls (see “Interactions,”
below, for more information).
For V2, no outside (trunk) calls extended by the attendant are eligible for HFAI
service.
●
Calls transferred from another System 25 set using the Transfer feature. The
transferring station may pass both inside and outside calls in this way. Note that
calls transferred by the attendant are indistinguishable from calls transferred by any
other station.
BIS and HFAI Voice Terminals
LEDs next to the AUTO ANS button and the HFAI/Mic (HFAI set) or HFAI (BIS set)
button indicate whether the HFAI feature is enabled. The LEDs are turned on and off by
pressing the adjacent buttons. When both the AUTO ANS and HFAI LEDs are on, the set
will auto-answer eligible calls.
The HFAI LED will wink (on HFAI sets) or light steadily (BIS sets) during HFAI calls.
The set’s response to HFAI-eligible calls depends on the status of the HFAI and AUTO ANS
buttons and LEDs, as follows:
●
If both HFAI and AUTO-ANS LEDs are on:
— The set generates a tone burst over its speaker to indicate an incoming call
(the calling party also hears this tone).
— The parties may converse. The called party can speak in a normal voice
toward the set. No other action by the called party is required.
2-142
— During the call, the called party can press the HFAI/Mic or MICROPHONE
button to mute the microphone temporarily and prevent the caller from
hearing. Pressing the button again turns the microphone on again.
— The HFAI/BIS user may press the SPEAKER (HFAI set) or the
SPEAKERPHONE (BIS set) button to end the call. For V2 systems only: if
the calling party hangs up first, this is not necessary.
●
If only the AUTO ANS LED is on:
— The set generates a tone burst over its speaker to indicate an incoming call
(the calling party also hears this tone).
— The set’s speaker turns on and the set “answers” the call.
— Call setup is complete. However, the called party can hear, but not respond
to, the calling party. To respond, the user must lift the handset or press the
HFAI/Mic button on an HFAI set or press the MICROPHONE button on a
BIS set.
— The HFAI/BIS user may press the SPEAKER (HFAI set) or the
SPEAKERPHONE (BIS set) button to end the call. For V2 systems only: if
the calling party hangs up first, this is not necessary.
●
If only the HFAI LED (or neither LED) is on:
— The HFAI feature is disabled. The call answering procedure is the same as
for a standard MERLIN set.
If, during an HFAI call, the user decides to pick up the handset, the HFAI/Mic or HFAI
LED will turn off. On an HFAI set, the user is not permitted to revert to hands-free
operation. (Pressing the HFAI button while using the handset will simply disable the HFAI
feature for subsequent calls.) A BIS set user may transfer a call from the handset to the
speakerphone by pressing the SPEAKERPHONE button and hanging up.
Voice Terminals with HFUs
These sets do not have an HFAI button. To turn on the HFAI feature the user simply
presses the AUTO ANS button; the green status LED lights.
After HFAI is activated, operation is exactly the same as for the BIS set except that the
SPEAKERPHONE and MICROPHONE buttons and LEDs are on the HFU.
Voice Terminals without HFUs
The HFAI feature is activated by pressing the AUTO ANS button. A beep signal announces
an incoming call and the SPEAKER LED lights. A one-way talking link is established from
the caller to the terminal; the user can hear the caller but cannot converse. Lifting the
handset connects the user to the caller.
Considerations
The user of a HFAI equipped station should always deactivate the HFAI feature when
leaving the work area. If this is not done, incoming calls will be “answered”, but the callers
will be talking to an unattended position.
2-143
Interactions
●
Attendant (V1 Only): The attendant may pass calls to HFAI sets using the
Transfer feature and button rather than extending these calls using the START
button or the Selector Console. Such calls are then eligible for HFAI service.
However, since unanswered transferred calls do not return to the attendant for
further service (as extended calls do), this practice is not recommended.
A better procedure is to mistranslate the HFAI sets as BIS sets (using the BIS set
code—305 at the System Administration Terminal). If this is done, attendant
extended calls are eligible for HFAI service at HFAI sets.
●
Call Coverage: When the HFAI feature is enabled at a set, calls eligible for HFAI
service will not receive call coverage because the set will answer them whether the
user is present or not. However, if the attendant uses the Attendant Message
Waiting feature to turn on the Message indicator at the set, the HFAI feature will be
disabled (the AUTO ANS LED turns off), allowing subsequent calls to receive
coverage.
●
Send All Calls: Activating Send All Calls will disable the HFAI feature (the AUTO
ANS LED turns off).
Administration Requirements
Voice Terminal (Station Port):
●
Assign AUTO ANS button.
Hardware Requirements:
This feature requires one of the voice terminals or combinations of terminal and HFU listed
in the Description.
2-144
HEADSET ADAPTER ADJUNCT
Description
The headset adapter adjunct provides an interface for connecting a headset to an associated
voice terminal. A headset plugged into the adapter is activated by switches on the adapter.
The terminal operator has the choice of using either the handset or the headset for handling
calls. Turning the headset on and off is equivalent to lifting and hanging up the handset.
Considerations
Use of a headset allows a voice terminal operator to carry on conversations with both hands
free for writing, typing, etc. It is valuable adjunct for high traffic positions such as
attendant consoles.
Use of a headset does not affect normal voice terminal operations in any way.
Interactions
●
Speakerphone Adjunct: A voice terminal cannot have both a headset and a
speakerphone. These adjuncts plug into the same jack on the voice terminal.
Administration Requirements
None
Hardware Requirements
500A/502A Headset Adapters:
The 500A adapter (Figure 2-26) is designed for use with the 12-Button (7203M) MET voice
terminal. The 502A adapter, which is identical in appearance, is designed for use with
MERLIN (7300H Series) terminals (with the exception of the 5-Button and HFAI sets).
Most standard commercial headsets can be used with the adapters.
Each adapter has an “ON/QUIET” button, an “OFF” button, a green indicator lamp, a jack
for a single headset, and two modular keyed jacks (4-wire and 8-wire). Each is equipped
with an 18-inch connecting cord. Optional cords are available in lengths of 4 and 14 feet.
The 500A Headset Adapter is powered locally by a 2012D Transformer, which plugs into a
115V ac receptacle. Power from the transformer is applied to the voice terminal mounting
cord via a 400B2 adapter at the wall jack and conducted to the 500A on its connecting cord.
Refer to “Voice Terminal Adjunct Power Supplies” in Section 4 for additional information.
The 502A Headset Adapter does not require supplemental power, except when used with a
34-Button Deluxe, BIS, or BIS with Display voice terminal.
Detailed headset adapter connection information is provided in the following figures:
●
Figure 2-27— Typical Headset Adapter Connections for 7300H Series Multiline Voice
Terminals (Except 34-Button Deluxe, BIS, or BIS with Display)
2-145
●
Figure 2-28 -Typical Headset Adapter Connections for 34-Button Deluxe, BIS, or BIS
with Display Voice Terminals
●
Figure 2-29 -Typical Headset Adapter Connections for 12-Button MET Sets
MET Headset Adapter:
Use of a headset with a 10-Button MET voice terminal requires a JS0180-3A Headset
Adapter (18 inch cord) or a JS0180-4A Headset Adapter (8 foot cord).
Figure 2-26. 500A/502A Headset Adapter
2-146
SYSTEM 25
CABINET
PART OF
OCTOPUS CABLE
ZTN79
HYBRID
LINE CP
—
●
—
—
PART OF SIP
SIP
ADAPT
C2
—
—
—
—
—
—
—
—
—
W1
B1
VOICE
TERMINAL
C1
T1
,
C8
502A HEADSET
ADAPTER PEC 3164-HFA
LEGEND:
B1C1C2C8 T 1-
TYPICAL-103A CONNECTING BLOCK*
MODULAR CORD (D8W-87) - FURNISHED WITH SET
OCTOPUS CABLE CABLE (WP90780) - PEC 2720-05P
SPECIAL CORD- FURNISHED WITH ADJUNCT
7300H SERIES VOICE TERMINAL EXCEPT 34-BUTTON DELUXE, BIS, OR
BIS WITH DISPLAY
W I - 4 PAIR INSIDE WIRING CABLE*
*
- FURNISHED BY INSTALLER
Figure 2-27. Typical Headset Adapter Connections For 7300H Series
Multiline Voice Terminals (Except 34-Button Deluxe, BIS, or
BIS with Display)
2-147
SYSTEM 25
CABINET
PART OF
OCTOPUS CABLE
— —
—
PART OF SIP
ZTN79
HYBRID
LINE CP
●
SIP
ADAPT
C2
— —
—
— —
—
— —
—
W1
C1
B1
VOICE
TERMINAL
T1
I
C7
PWR
SUPPLY
P1
C8
502A HEADSET
ADAPTER PEC 3164 - HFA
LEGEND :
B1 - TYPICAL-103A CONNECTING BLOCK*
C 1- MODULAR CORD (D8W-87) - FURNISHED WITH SET
C2 - OCTOPUS CABLE CABLE (WP90780) - PEC 2720-05P
C8 - SPECIAL CORD - FURNISHED WITH ADJUNCT
T1 - 7300H SERIES VOICE TERMINAL (34-BUTTON DELUXE, BIS, OR BIS WITH DISPLAY)
W1 - 4 PAIR INSIDE WIRING CABLE*
C7 - MODULAR CORD (D6AP-87)
PEC 62510
P1 -KS 22911 POWER SUPPLY
Z400F - ADAPTER
* _ FURNISHED BY INSTALLER
Figure 2-28. Typical Headset Adapter Connections For 34-Button Deluxe,
BIS, or BIS with Display Voice Terminals
2-148
SYSTEM 25
CABINET
TN735
MET
LINE CP
PART OF
OCTOPUS
CABLE
— — —
—
—
—
PART OF SIP
●
●
C2
SIP
ADAPT
W1
B1
400B2
ADAPT
C1
MET SET
T1
— — —
C7
— — —
—
—
—
—
—
—
248B
ADAPT
2012D
TRANS
C8
500A
HEADSET
ADAPTER PEC 31712
LEGEND:
B 1 - TYPICAL-103A CONNECTING BLOCK*
C1 - MODULAR CORD (D8W-87) - FURNISHED WITH SET
C2 - OCTOPUS CABLE (WP90780) PEC 2720-05P
C8 - SPECIAL CORD - FURNISHED WITH ADJUNCT
T1 - 7203M SET - 12-BUTTON MET SET
W 1- 4 PAIR INSIDE WIRING CABLE*
248B ADAPTER -MODULARIZES 2012D TRANSFORMER
400B2 ADAPTER - POWER ADAPTER
PEC 21691
2012D TRANSFORMER - 15-18V AC TRANSFORMER
C 7- MODULAR CORD (D6AP-87)
* - FURNISHED BY INSTALLER
Figure 2-29.
Typical Headset Adapter Connections For 12-Button MET
Sets
2-149
HOLD
Description
Allows users to temporarily disconnect from one call and either place or answer another call.
A single-line voice terminal user can place only one call on hold and must remain off-hook to
retain the held call. A multiline voice terminal user can place as many calls on hold as it
has lines and can hang up without losing held calls.
Single-line users can place a call on hold by flashing the switchhook (the user receives
Confirmation Tone). The user can then dial another party or return to the held call by
flashing the switchhook twice. The first switchhook flash sets up a conference call, the
second flash drops the third party; if System 25 Dial Tone, Busy Tone, or Reorder Tone (but
not Ringback Tone) was obtained when the third party was dialed, a single switchhook flash
will drop the tone and return the user to the held party.
Multiline voice terminal users can press HOLD and subsequently replace the handset or call
another party without losing the held call. The status LED associated with the held call
winks on all terminals with an appearance of the line, except in the case of a conference call.
In this case, the wink indication is given only to the party who invoked hold. To return to
the held call the multiline user either presses the line button associated with the held call
(VI and V2) or simply goes off-hook if the held call is on a button assigned prime line
preference (V2 only).
Considerations
The Hold feature allows voice terminal users to handle several calls simultaneously. For
single-line sets, placing a call on hold is the first step in transferring or conferencing the
call.
Interactions
●
Attendant Console: The Attendant does not receive hold indications for lines
(trunks) appearing on the Console unless he/she placed the call on hold.
●
Exclusion: A call can be placed on hold after Exclusion is invoked. The status LED
of the line appearance button and the Exclusion button will wink.
●
Music-On-Hold: A held party on an outside line will receive Music-On-Hold if
provided.
●
Personal Lines: A Personal Line cannot be placed on hold if any other stations are
also off-hook on that line.
Administration Requirements
None
Hardware Requirements
None
2-150
INTERCEPT TREATMENT WITH REORDER TONES
Description
Reorder tone (fast busy) is provided when an unassigned or toll-restricted number is dialed,
a dialing error occurs, or an attempt to park a call fails.
Any attempt to dial an unassigned code (PDC, DDC, feature or facility access code) or an
unsuccessful attempt to park a call will result in Reorder Tone being provided to the caller.
A DID call to an unassigned number will be routed to the attendant or will receive Reorder
Tone at the System Administrator’s option. Any attempt to dial a restricted call (toll or
access restricted) will be intercepted and routed to Reorder Tone.
Considerations
Intercept treatment provides a calling party with positive feedback of an error in dialing or
use of an incorrect code.
Interactions
●
Call Park: An unsuccessful attempt to park a call due to misdialing or attempting
to park more than one call at a voice terminal results in Reorder Tone.
Administration Requirements
None
Hardware Requirements
None
2-151
INTERDIGIT TIMEOUTS
Description
Allows an originating register to be made available for others if dialing is not completed
within a set time period.
Interdigit timeouts is 24 seconds until the first five digits have been dialed, 10 seconds until
the next five digits have been dialed, and five seconds thereafter.
After timeout, voice terminal dial pad button presses are interpreted as end-to-end signaling
requests and touch-tones are placed on the outside line.
Considerations
Interdigit timeouts also apply to data calls.
When a user dials out over a trunk set up for dial pulse rather that Touch-Tone service, the
interdigit timeout interval is involved. The caller cannot speak to the called party until the
timeout expires (even though the caller may be able to hear the called party). Because the
timeout interval decreases as more digits are dialed, the problem is often noticed on tie
trunks since the user may dial fewer (or no) digits when using these trunks. If the user
presses the # button after dialing the last digit this timeout ends and the caller may speak
immediately.
Interactions
None
Administration Requirements
None
Hardware Requirements
None
2-152
LAST NUMBER DIALED (V2)
Description
Automatically saves the last number dialed from a multiline voice terminal and allows the
user to place the call again without redialing the number. The feature is administered to a
button on the terminal. Both inside and outside calls can be made in this way. The original
call can be placed by manual dialing, by operation of a programmed button, or by speed
dialing.
To use the feature, the caller first gets dial tone, and then presses the Last Number Dialed
button. The I-use and status LEDs of the button selected for originating the call (for
example, System Access, Loop, or Personal Line) light steadily; if the Last Number Dialed
button has an LED, it lights momentarily. The call proceeds in the normal way. The number
associated with the Last Number Dialed button remains saved even if the called party
answers. Only the dialing of a new number changes the state of the Last Number Dialed
button; the old number is then erased and the new one stored.
Considerations
Last Number Dialed is a convenience feature that is especially valuable for recalling multidigit numbers that were first dialed manually from a terminal’s pushbutton dial. However,
the feature can also be used to recall numbers originally called by the following means:
●
Repertory Dial buttons.
●
Group Select and DXS buttons on a Selector Console.
This feature saves numbers with up to 16 digits.
The user must hear dial tone before pressing the Last Number Dialed button.
The Last Number Dialed feature cannot be activated by dialing an access code. It is not
available to users of single-line voice terminals.
If a dialed number does not complete a call, Last Number Dialed still stores the digits dialed.
If the user presses the Last Number Dialed button, then dials additional digits to complete
the call, both the currently stored digits and the dialed digits will be stored.
Interactions
●
Account Code Entry: Last Number Dialed does not save an Account Code dialed by
operation of an Account Code Entry button.
●
Attendant Display: When a call is placed at a Switched Loop Attendant Console
(V2), using the Last Number Dialed button, the call information display has the
normal format of an outgoing call. In the Inspect mode, pressing the Last Number
Dialed button displays the number currently stored on the button.
●
Automatic Intercom: Numbers called using an Automatic Intercom button are not
saved by the Last Number Dialed feature. The number currently stored by Last
Number Dialed is not changed by operations of the Automatic Intercom button.
2-153
●
Call Accountability: When a station user dials ##PDC to provide accountability for
a call and then dials the desired digits, the ##PDC is not saved by the Last Number
Dialed feature.
●
Conference: When a station user adds a party to a conference, the number dialed is
saved as the Last Number Dialed.
●
Direct Station Selection (DSS): Numbers called using an DSS or Flex DSS
button are not saved by the Last Number Dialed feature. The number currently
stored by Last Number Dialed is not changed by operations of these buttons.
●
Personal Lines: When a user originates a call from a Personal Line, only the digits
dialed after the line is accessed are saved by the Last Number Dialed feature. The
same type of line must be selected to get dial tone for placing another call using Last
Number Dialed. If a different type of line is used, the call may be directed to the
wrong destination.
●
●
●
Pooled Facility—Direct Access: When a user originates a call from a Direct
Facility Access button, only the digits dialed after the line is accessed are saved by
the Last Number Dialed feature. The same type of button should be selected to get
dial tone for placing another call using Last Number Dialed. If a different type of
button is used, the call may not be directed to the right destination.
Repertory Dialing: When using a Repertory Dial button to place a call, the
numbers dialed are saved by Last Number Dialed. When the call is redialed using
Last Number Dialed, the same type of button where dial tone was originally
accessed should be used again to ensure that the call is directed to the correct
destination.
Speed Dialing: When using a Personal or System Speed Dialing code to place a call,
the code is saved by Last Number Dialed. When the call is redialed using Last
Number Dialed, the same type of button where dial tone was originally accessed
should be used again to ensure that the call is directed to the correct destination.
●
System Access/System Access Originate Only Buttons: If a user originates a
call from one of these buttons, the same type of button should be selected for getting
dial tone to place a second call with the Last Number Dialed feature. Using another
type
of button, such as Personal Line or Direct Facility Access, to get dial tone may
.
prevent the call from completing properly.
●
Transfer: When a station user Transfers a call, the dialed number (of the party to
whom the call is transferred) is saved as the Last Number Dialed.
Administration Requirements
Last Number Dialed is a default feature on all multiline voice terminals in a V2 system.
One button is assigned to the feature at each set. The feature can be moved/removed by
administration.
Hardware Requirements
None
2-154
LINE SELECTION
Description
Multiline voice terminals may have many line (facility) appearances. There are three
methods by which a user may select a desired line: (1) Prime Line Preference, (2) Ringing
Line Preference, and (3) Preelection.
Prime Line Preference:
Automatically connects a multiline voice terminal to a specified line or facility designated as
preferred when the terminal goes off-hook. This feature may be assigned to System Access,
Loop, Automatic Intercom, DSS, Personal Line, and Direct Facility Access buttons.
On the Switched Loop Attendant Console (V2), the topmost Loop button has Prime Line
Preference by default. However, the feature can be assigned to any of the five Loop buttons.
The user may override this feature by preselecting another button (see below).
If Prime Line Preference is assigned to an Automatic Intercom (AUTO ICOM) or DSS
button, the called voice terminal will ring as soon as the terminal goes off hook.
When the Prime Line Preference feature is assigned to an AUTO ICOM, DSS, or Direct
Facility Access (FACILITY) button, the button must be pressed to activate the busy-to-idle
reminder even though the button’s I-Use LED is lighted steadily.
If Prime Line Preference is assigned to a Personal Line button, the user is connected to the
button upon going off-hook, even if the line is busy. For V1 systems only: if Prime Line
Preference is assigned to a button that has a call on hold, no connection will be made; to get
a line for placing a call, the user must press another button. In V2 systems, the user is
always connected to the button where Prime Line Preference is assigned, even if the line has
a call on hold; if the user wishes to use a different line, the line may be preselected before
going off-hook.
Ringing Line Preference:
Automatically connects a multiline voice terminal to an incoming call ringing at the
terminal.
Line access buttons that can be selected by Ringing Line Preference include System Access,
Automatic Intercom, Coverage, and Personal Lines.
If two or more lines on a multiline terminal or a Direct Trunk Attendant Console are
ringing simultaneously, the user is connected to the first line to start ringing. If the user
wishes to use a different line, the line must be preselected prior to going off-hook. If ringing
ceases while the user is on-hook, line preference reverts to whichever option is applicable
(Prime Line Preference or no preference).
If a line rings at a multiline terminal when the terminal is busy on another call, Ringing
Line Preference will not activate, even if the user goes on-hook during the ringing cycle.
However, Ringing Line Preference is not canceled at the Attendant Console while the
attendant is off-hook. If a line is ringing while the attendant is off-hook, the ringing line
will be selected as soon as the attendant goes on-hook.
2-155
Preselection:
Allows multiline voice terminal users to override the above line preference features.
Users may simply press a desired line access button before going off-hook. The user will be
connected to the facility selected unless the facility is busy and the party using it has
invoked Exclusion or is part of a conference call that is at maximum capacity.
When off-hook, a user can select a facility by pressing the associated button. (This will
terminate the call the user was on. )
A user may activate the busy-to-idle reminder on a busy AUTO ICOM, DSS, FLEX DSS, or
FACILITY button by pressing the button while on-hook. A burst of ringing is provided
when the facility becomes idle. Refer to the subsection on the Busy-to-Idle feature for
additional information.
Considerations
Prime Line Preference (on the topmost SYSTEM ACCESS or Loop button) and Ringing Line
Preference are assigned by default to all multiline voice terminals. While these assignments
may be changed, it is strongly recommended that Ringing Line Preference be retained.
It is recommended that Prime Line Preference not be assigned to a Direct Trunk Attendant
Console (DTAC).
Preselection allows users to override line preference features already administered for the
terminal and to activate the busy-to-idle reminder feature.
Interactions
●
Attendant Console, Direct Trunk: If a line rings at a DTAC while the attendant
is on another call, Ringing Line Preference will be invoked when the attendant hangs
up.
●
Ringing Line Preference:
Ringing line preference overrides Prime Line
Preference and Preselection when a call is ringing at an on-hook voice terminal.
●
Power Failure/Cold Start: On power-up, most multiline voice terminals will have
no I-use LED lit and will not draw dial tone until a button is pressed. After this, line
selection should work as described above.
Administration Requirements
Voice Terminal (Station Port):
●
Prime Line Preference:
Assign Prime Line Preference - Default = Top SYSTEM ACCESS button or top
LOOP button (Switched Loop Attendant Console only).
●
Ringing Line Preference:
Assign Ringing Line Preference (Yes, No) - Default = Yes.
Hardware Requirements
None
2-156
LINE STATUS AND l-USE INDICATIONS
Description
Provides users with a visual indication of the status of feature buttons and lines appearing
at a their multiline terminals. A green status LED and a red I-Use LED are provided for
each programmable button on most multiline voice terminals.
Table 2-I summarizes LED states and associated descriptions for line appearances.
TABLE 2-I. LED Indications
I-Use
(Red LED)
Line Status
(Green LED)
Meaning
Off
Off
Facility is idle.
On
On
If off-hook, facility is in use at this terminal.
If on-hook, busy-to-idle reminder is set.
Off
On
Facility is busy or Feature has been activated.
Off
Winking
Facility placed on hold.
On
Flashing
Facility ringing; call will be answered if user
goes off-hook.
Off
Flashing
Facility ringing; call will not be answered if
user goes off-hook.
On
Off
Facility that will be accessed upon going offhook.
Off
Broken Flutter
Facility is being transferred or conference.
Considerations
Line Status and I-Use indications provide the user with visible indications of the status of
the lines and features.
Interactions
●
Attendant Console, Direct Trunk: When a line that appears at both the
attendant position and a multiline voice terminal is placed on hold by the terminal
user, the green status LED winks at the terminal but lights steadily on the
Attendant Console. When the line is placed on hold by the attendant, the green
status LED winks on the console and on voice terminals on which it appears.
Trunk-to-trunk transfers will cause the affected PERS LINE buttons on the DTAC to
wink.
2-157
Administration Requirements
None
Hardware Requirements
None
2-158
MANUAL SIGNALING
Description
Allows a user to signal another voice terminal. The user may do this at any time, whether
on-hook or off-hook.
Multiline voice terminal users can signal another predesignated multiline voice terminal by
pressing an associated Manual Signaling (SIGNAL) button. A single tone burst is provided
at the signaled terminal. The signaling voice terminal also receives the tone and can use
this feature while in any call state. No LED indication is associated with the Manual
Signaling feature.
When the Manual Signaling feature is used while the called station is ringing on another
call, no audible signal is received by either the signaling or the called voice terminal.
The duration of the single burst of signaling will always be the same, regardless of how long
SIGNAL is pressed. The signal is repeated each time the button is pressed.
Considerations
Manual Signaling allows a user to signal another voice terminal without calling the
terminal. The meaning of the signal may be prearranged between the sending and the
receiving parties. Only multiline terminals may be signaled.
Interactions
●
External Alerts: Manual Signaling will not activate external alerting devices
associated with the signaled station.
Administration Requirements
Voice Terminal (Station Port):
●
Assign Manual Signaling button.
Hardware Requirements
None
2-159
MESSAGE CENTER-LIKE OPERATION (V2; SLAC Only)
Description
A System 25 Switched Loop Attendant Console (SLAC) can be made to function like a
message center through administration of call type translations. Certain specific types of
calls in the common queue will then be directed only to a console administered as a message
center. This arrangement involves no changes in equipment or in operating procedures. The
message center attendant answers incoming calls of the preselected types in the normal way.
No provision is made for storing messages, a capability often associated with full service
message centers.
Message Center Call Types:
The Message Center receives calls of the following types:
●
Returning parked calls that were originally parked from a Selector Console.
●
Returning
●
Returning calls that were extended (transferred) from an Attendant Position to a
busy station or a station that does not answer.
●
Coverage calls–incoming inside and outside calls (including DID calls) covered by
the common queue when the called party does not answer, is busy, or does not want
to be disturbed (Send All Calls).
●
Floating PDCs not logged in at a station and unassigned DID calls.
camped-on
calls.
The type of each incoming Message Center call will be identified by a call type descriptor on
the console’s 16-character display screen. Refer to the earlier Switched Loop Attendant
Console subsection for a list of descriptors.
Console Configurations:
Message Center-like operation applies only when the System 25 has two SLACs. In a oneconsole system, all calls are handled at the same position.
In the default condition, a SLAC is a combined Attendant Position/Message Center. It can
receive any type of call. In a one-console system, there is no division between attendant and
message center functions.
A dedicated Message Center is a console that is administered to receive only the specified
incoming call types. Dial O calls (attendant-seeking calls from inside the system) are not
directed to a Message Center, but the console has a unique PDC number that callers can use
to reach the attendant.
A dedicated Attendant Position is a console that is administered to answer all of the calls
not handled by the Message Center.
Message Center capability can be supported in any
configurations:
of the following two-console
●
One dedicated Attendant Position and one dedicated Message Center. A call
extended by the Attendant Position to a station that does not answer or is busy
returns to the Message Center.
●
One dedicated Attendant Position and one combined Attendant Position/Message
Center; the Attendant at the combined position also functions as the Message Center
2-160
operator for the entire system. A call extended by either Attendant to a station that
does not answer or is busy returns to the Message Center.
●
Two combined Attendant Position/Message Centers; the Attendants at each
combined position also function as Message Center operators. The special call types
answered by Message Centers can be divided between the two consoles, or both
consoles can answer all types. A call extended by either Attendant to a station that
does not answer or in busy returns to the Message Center specified in translation.
Three return options are provided: to the 1st attendant, to the 2nd attendant, or to
either attendant.
Interactions
All System 25 console features are accessible at a Message Center. Operating procedures are
exactly the same as those at a standard Attendant Position. The BIS and HFAI features can
be used to answer Message Center calls.
An auxiliary Direct Extension Selector Console can be used with a Message Center.
Outgoing calling, from the Basic Console or the Selector Console, is not affected by Message
Center administration.
Administration Requirements
A new item in administration allows selection of an alternative set of call type
making one of the attendant positions a “message center”. These defaults may be
for individual call types, if desired, to tailor the message center-like operation for
locations. For administration purposes, the Message Center call types are divided
following four groups:
●
Nonlogged-in floating PDC calls
●
Unassigned DID calls
●
Coverage calls
●
Returning
defaults,
modified
different
into the
calls.
With the standard (ie, non-message center) defaults, each of these groups is translated to be
directed to all consoles. When both positions are combined Attendant/Message Center
consoles, administration can direct specific types to one or both consoles.
Message Center calls are held in the same common queue as any other attendant-seeking
calls before being directed to the console.
Hardware Requirements
None.
2-161
MESSAGING SERVICES
Description
Lights an LED to indicate that another station (or the attendant) has a message for the
user.
The Messaging Services provide light activation/deactivation only. Users must call the
sender to receive their messages.
The system supports three types of Message Waiting service:
●
Attendant
Message
Waiting
●
Call Coverage Message Waiting
●
Station-To-Station Message Waiting
Attendant Message Waiting:
The Attendant can turn On (and
this indicator is lighted, users
terminals may be turned Off by
Message LED on most single-line
turn Off) the Message LED at other voice terminals. When
call the attendant for messages. The LED on multiline
the user (by pressing MESSAGE) or by the attendant. The
terminals can only be turned On or Off by the attendant.
Refer to the Attendant Message Waiting feature description for additional information.
Call Coverage Message Waiting:
Allows a user providing Individual Call Coverage to control the Message LED on covered
voice terminals. A Coverage Message button (COVER MSG) is used to display and control
the status of the covered user’s Message LED. The state of the COVER MSG LED reflects
the state of the covered station’s Message LED. The covering user can turn On or Off
(toggle) the covered party’s Message LED at any time during a coverage call by pressing
COVER MSG. To turn On the covered user’s Message LED when not on a coverage call, the
covering user may go off-hook on a System Access button, press COVER MSG and then dial
the covered user. The covered station’s Message LED turns On if Off and stays On if On. If
the covering station then presses COVER MSG a second time before hanging up, the
Message LED will turn Off.
A covered party must dial the covering party to retrieve their messages. Multiline voice
terminal users can press MESSAGE to turn Off their Message LED. Message indicators on
most single-line voice terminals can only be controlled by the covering party or the
attendant.
If a user tries to turn On the Message LED at a voice terminal for which they don’t provide
Individual Call Coverage, they receive Reorder Tone.
Refer to the Call Coverage-Individual feature description for additional information.
2-162
Station-To-Station Message Waiting:
Multiline voice terminals can be assigned (paired) Message Waiting (MSG WAIT) buttons
with associated status LEDs. When this indicator is lighted, the user calls the other user for
messages. The MSG WAIT LED can be controlled by the two associated terminals o n l y ;
either user can toggle the state of both LEDs (e.g., both LEDs go On or Off together) at any
time, whether on-hook or off-hook.
Refer to the “Station-To-Station Message Waiting” feature description for additional
information.
Considerations
The Attendant and Coverage Message Waiting features light the same “basic” Message
indicator on each set. The Station-To-Station feature may be assigned to programmable
(MSG WAIT) buttons between two sets; it lights the LED next to the button.
Interactions
None
Administration Requirements
Attendant Position (Station Port):
●
Assign ATT MSG button (defaulted)
Voice Terminals (Station Port):
●
●
●
Individual Call Coverage Message Waiting - assign Coverage Message (COVER-MSG)
button.
Assign Individual Call Coverage (COVER-IND) between sets.
Station-To-Station Message Waiting - assign Station Message Waiting (MSG WAIT)
buttons. Two (multiline) terminals must share this feature.
Hardware Requirements
Z34A Message Waiting Indicator for single-line voice terminals not equipped with Message
LEDs.
2-163
MODEM POOLING
Description
Allows switched data connections between digital data endpoints and analog data endpoints.
(Refer also to the description of the system’s data features provided in the “Data Services
Overview’’ subsection.)
Data transmission between digital and analog endpoints requires a conversion resource since
the digital format used by the data module is not compatible with the modulated signals of
an analog modem. The conversion resource translates the digital signals from the digital
endpoint into analog signals and vice versa.
The modem pool is a single group of up to 12 conversion resources (3 Cabinet system) with
the characteristics of a 212A full duplex asynchronous modem that can operate at speeds of
300 and 1200 bps.
The Modem Pooling feature operates transparently to the user whenever possible. The
system adds a conversion resource to a connection when a digital endpoint is connected to an
analog trunk or port without any explicit action by the user.
A voice terminal user who plans to use an analog modem to call a digital endpoint must first
enter the Modem Request Code before dialing the digital endpoint. This is because the
system assumes that a voice call to a digital endpoint will be transferred to data via the
Transfer To Data feature.
A DID call terminating on a digital endpoint will reassigned a modem resource, if available.
Otherwise, the call receives Reorder tone.
For each situation that requires a conversion resource, the system:
1. Determines if a resource is required by examining the types of endpoints that are
to be connected together or by user indication.
2. Once it is determined that a conversion resource is needed, it is reserved. The user
receives Reorder Tone (or the “NO MODEMS - TRY AGAIN” message) if a
resource is not available. The system queries the data port to determine whether
its options are compatible with those supported by the modem pool. If they are not
(e.g., 9600 baud and Permit Mismatch is not enabled), the originating user receives
intercept treatment (i.e., INCOMPATIBLE FAR END) and call setup is abandoned.
3. At data connection time, the conversion resource is seized and placed in the
connection.
4. The call is disconnected within 15 seconds if the conversion resource does not
successfully handshake with both endpoints.
Conversion resources are required for:
●
Data Terminal Dialing: To establish a data connection for calls originated via
Terminal Dialing to intrapremises analog data endpoints.
●
Incoming Trunk Calls: To establish a data connection between an incoming trunk
call and a digital endpoint. Incoming trunk calls that are answered at a voice
terminal can be transferred to a data endpoint using the Transfer To Data feature.
2-164
●
On-Premises Data Calls: To establish a data connection between an on-premises
analog data endpoint and an on-premises digital endpoint.
●
Outgoing Trunk Calls: To establish a data connection between an off-premises
analog endpoint (modem) and an on-premises digital endpoint.
Considerations
Modem Pooling provides a pool of conversion resources that increases data call flexibility.
Conversion resources allow analog data endpoints, using modems, to communicate with
digital data endpoints (using data modules). Also, modem pooling reduces costs by sharing
resources.
Interactions
●
Automatic Route Selection: Data calls may be queued. See Table 2-H, “PLEASE
WAIT” message.
●
Calling Restrictions: If a terminal is toll or access restricted, the modem resource
is released when the user receives intercept treatment.
●
Direct Group Calling: Modem pooling supports calls to data endpoints that are
part of a DGC group. While an incoming data call is in the DGC group queue, the
caller hears Ringback Tone. The conversion resource is inserted if the call is
completed to a digital endpoint.
●
Station Hunting: Modem Pooling supports calls to data endpoints that are part of a
station hunting group.
●
SMDR: SMDR records do not reflect modem resource usage. Interpremises data
calls using a conversion resource are reported as data calls on the SMDR call record.
Administration Requirements
System (Pooled Modems):
●
Modem Request Code (l-9999) - Default = 820. Allows users to indicate a need for a
conversion resource on a data call originated at an analog data endpoint.
●
Receiver Responds To Remote Loop (Yes, No) - Default = Yes. When active, Data
Set Ready is asserted when the modem is in an analog loop test mode.
●
Disconnect On Loss Of Carrier (Yes, No) - Default = Yes. When active, a loss of the
received carrier will cause the modem to terminate the call.
●
CF-CB Common (Yes, No) - Default = Yes. When active, Clear to Send turns off if
Carrier Detect turns off. When a call is being established, Clear to Send and Carrier
Detect are not allowed to turn on until carrier has been received and the Clear to
Send timer has timed out.
●
Disconnect On Received Space (Yes, No) - Default = Yes. When active, the modem
will disconnect after receiving a “Space” signal of approximately two seconds
duration.
●
Send Space On Disconnect (Yes, No) - Default = Yes. When active, the modem, upon
receiving a negation of Data Terminal Ready, sends approximately four seconds of
“Space” signal and then disconnects. Without this option active, the modem, upon
receiving a negation of the Data Terminal Ready signal, disconnects immediately.
2-165
Hardware Requirements
One TN758 Pooled Modem CP provides two conversion resources. Two TN758s are allowed
per system cabinet, for a total of 12 conversion resources in a 3-cabinet system.
2-166
MUSIC-ON-HOLD
Description
Provides rnusic or other audible indication to a held party on an outside line.
On an outside call, if the user places the call on Hold, or after a call into a DGC group
receives the delay announcement, music is provided to the calling party.
Music-On-Hold is not invoked when a conference call is placed on hold or when the attendant
“Start” facility is used to place a call on hold.
When a multiline voice terminal user places a call on hold, the status LED of the held line
winks and music is provided to the held party. The user may return to the held party by
pressing the button associated with the held call. The status LED lights steadily, music is
removed from the line, and a talking connection is again established.
When a single-line voice terminal user places a call on hold by pressing the switchhook
momentarily, the calling party is connected to music or a recording. The station may return
to the held call by pressing the switchhook a second time. The music is removed from the
line and the held party is reconnected to the user.
If a caller receives music because all members of a DGC group are busy, when a group
member becomes available to answer the call, music is removed and the calling party is
connected to the DGC member.
Music-On-Hold is not provided for data calls or inside calls.
Considerations
Music-On-Hold lets the waiting party know that he or she is still connected.
During the process of transferring a call or setting up a conference, the affected parties are
placed in a “special hold” condition. In V1 systems, the parties on special hold will receive
music. In V2, an administration option allows Music-On-Hold to be enabled or disabled for
“Special Hold.”
Interactions
●
●
●
●
Attendant Splitting One-Way Automatic: Music-On-Hold is not provided when
the attendant presses START.
Conference: An outside line placed on hold when CONFERENCE is pressed will
hear music if provided.
Call Park: Parked calls (except parked conferences) receive music.
Direct Group Calling: An incoming call to a busy DGC group that provides a
recorded delay announcement will receive music after the announcement.
Administration Requirements
Assign a port on a TN742 Analog Line or ZTN78 Tip Ring Line CP as required for
the music/message source (special feature port type = 254).
For V2 only: allow Music-On-Hold for special hold (Y/N)
2-167
Hardware Requirements
A music source is needed to support the Music-On-Hold feature. The interface is a port on a
ZTN78 Tip Ring Line CP or TN742 Analog Line CP.
Detailed connection information is provided in the following figures:
●
Figure 2-30—Music-On-Hold Equipment Connections (FCC Registered)
●
Figure 2-31—Music-On-Hold Equipment Connections (Non-Registered).
Descriptions of the SIP (Station Interconnect Panel), TAE (Trunk Access Equipment), and
associated cables and adapters, as shown on the figures, are provided under the heading
“connectivity” in Section 4.
SYSTEM 25
CABINET
PART OF
OCTOPUS
CABLE
— — —
TN742
OR
ZTN78
●
C2
— — —
PART OF
SIP
SIP
ADAPT .
W1
MOH
W1
B1
C5
MUSIC
SOURCE
— — —
LEGEND:
TN742 - ANALOG LINE CP
ZTN78- TIP RING CP
B 1 - TYPICAL-103A CONNECTING BLOCK*
C 2 - OCTOPUS CABLE (WP90780) PEC 2720-05P
C5 - MODULAR CORD (D4BU-87)
W 1 - 4 PAIR INSIDE WIRING CABLE*
MOH - KS-23395 INTERFACE PEC 62517
* - FURNISHED BY INSTALLER
Figure 2-30. Music-On-Hold Equipment Connections (FCC Registered)
2-168
SYSTEM 25
CABINET
TN742
OR
ZTN78
PART OF
OCTOPUS CABLE
PART OF SIP
C2
W1
SIP
ADAPT.
MOH
UNIT
W1
368 VOICE
COUPLER
W1
B1
C6
MUSIC
SOURCE
W1
2012D
TRANSFORMER
LEGEND:
TN742 - ANALOG LINE CP
ZTN78 - TIP RING CP
B 1 - TYPICAL-103A CONNECTING BLOCK*
C2 - OCTOPUS CABLE (WP90780) PEC 2720-05P
C6 - SINGLE-ENDED MODULAR CORD, D4BY
W 1 - 4 PAIR INSIDE WIRING*
PEC 62513 INCLUDES KS-23395 MOH UNIT, 2012D TRANSFORMER
AND 36A VOICE COUPLER
* - FURNISHED BY INSTALLER
Figure 2-31. Music-On-Hold Equipment Connections (Non-Registered)
2-169
NIGHT SERVICE
Description
Allows users to answer incoming calls on specified trunks when the attendant is not on duty.
There are two types of Night Service (NS):
●
Directed NS: Redirects incoming calls on specified trunks to designated voice
terminals.
●
Trunk-Answer-from-Any -Station (TAAS) NS: Allows users to answer
incoming calls on specified trunks by dialing the Night Service access code.
Both types of NS may be provided (specified on a per-trunk basis).
To obtain Night Service, the system must be equipped with an Attendant Console, and the
console administered with a NIGHT button. In a system with two Attendant Consoles, both
consoles may be assigned a NIGHT button. Either attendant can press NIGHT to activate
Night Service. The LEDs of both NIGHT buttons will light to indicate that the system is in
the Night Service mode. Pressing NIGHT a second time (by either attendant) deactivates
Night Service and turns Off both LEDs.
Directed NS:
Allows an incoming trunk call to be directed to up to four designated voice terminals.
Different trunks may be directed to different voice terminals.
When the attendant presses NIGHT, incoming calls on trunks administered to receive
Directed NS treatment will automatically be routed to the designated voice terminals (all
designated NS stations ring simultaneously). Calls not answered within a specified number
of rings will receive a Night Service delay announcement, if available. While at the
announcement, they may be bridged onto by going off-hook at a station with a line
appearance. The announcement is dropped at this point. If all Directed NS stations for a
given trunk are busy (both System Access buttons busy on multiline sets), calls go to the
announcement immediately. Directed NS calls do not hunt or receive call coverage, but they
can be picked up via the Call Pickup feature.
Personal Line calls that are directed to NS will also ring at the Personal Line appearances
and receive normal call coverage.
Incoming calls receiving Directed NS treatment will not activate external alerting devices
associated with TAAS NS and cannot be answered by dialing the NS access code.
Directed NS is activated under the following conditions:
●
An attendant has pressed NIGHT on either console.
●
Directed NS has been administered for the trunk.
●
Stations have been administered to receive NS calls.
Note that at least one station must be designated as a NS receiver for this feature to work
properly. If only an announcement is required, administer the announcement device as a
station and make this station the NS receiver.
Refer to the “Night Service Delay Announcement” feature description for additional
information on the delay announcement.
2-170
Trunk-Answer-from-Any-Station:
Allows any user to answer NS calls. Incoming trunk calls activate an external alerting
device such as a bell (“External Alerts” feature). A user can then dial the NS access code
and answer the call. Night Service is activated under the following conditions:
●
An attendant has pressed NIGHT on either console.
●
TAAS NS has been administered for the trunk.
●
A NS external alert has been installed and administered.
Note that TAAS NS calls will not activate the delay announcement associated with Directed
NS.
Considerations
Directed NS provides a means of ensuring that Night Service calls are answered by
designating individual voice terminals to receive the calls. In noisy environment, for
example, NS via external alerting devices may not be practical. Directed NS provides a
solution to the noise problem. Also, Personal Line calls to executives can receive special
handling by providing Directed NS. Calls continue to ring at the attendant position or
Personal Line appearances when NS is activated. They also ring the external alert (TAAS)
or Directed NS station.
Trunk-Answer-from-Any -Station provides the capability for any user to answer NS calls.
Interactions
●
DID Trunks: DID trunks are not assignable to NS. A DID call will ring at the
appropriate station whether NS is activated or not.
●
Tie Trunks: Dial-in Tie Trunks will not receive night service treatment.
●
Call Following: Directed night service calls will not be given call following
treatment if the PDC is logged in at another station.
Administration Requirements
Trunk Ports:
●
Assign trunk Class of Service with Night Service - (8-15)
●
Assign Directed Night Service trunk (Yes, No) - Default = Yes.
Voice Terminal: (Station Port)
●
Directed
NS
— Add Night Service trunk number to station list.
●
Assign External Alert for TAAS NS.
Attendant Console: (Station Port)
●
Night Service is defaulted to a button on the first (VI: the primary) Direct Trunk
Attendant Console only. On a second (VI: secondary) Direct Trunk Console or on a
Switched Loop Attendant Console (V2 only), assign Night Service to a flexible button.
2-171
●
Assign Night Service Access Code.
Hardware Requirements
TAAS NS requires an associated external alert (such as a bell). Each alert requires a port
on a ZTN78 Tip Ring Line or a TN742 Analog Line CP. Refer to the “External Alerts”
subsection for detailed information and a connection diagram.
2-172
NIGHT SERVICE DELAY ANNOUNCEMENTS
Description
Provides a recorded announcement for incoming trunk calls when the system has Directed
Night Service (NS) activated and a call is not answered.
Directed NS calls not answered within a specified number of rings (1-15) may be directed to
a recorded announcement. Two different recorded announcements may be assigned.
Note that NS calls to a terminal that are not answered do not receive Station Hunting or
Call Coverage treatment (unless the trunk also appears on a station’s Personal Line button).
After the announcement is played, the call is disconnected.
Considerations:
Night Service Delay Announcements provide the calling party with a message that
acknowledges the call and can provide additional information as well. Once an NS call goes
to the delay announcement, the call will be disconnected from the system after the
announcement has been played.
Interactions
●
Night Service: Incoming calls receiving TAAS NS treatment will not activate the
delay announcements. Only trunks that receive Directed NS will activate these
announcements.
Administration Requirements
Special Feature Ports:
●
Assign first Night Service delay announcement (code 251 )
●
Assign second Night Service delay announcement (code 252)
●
Assign number of rings before Night Service delay announcement (0-15)
Station Ports
●
Assign port circuits (ZTN78 or TN742) for each recorded announcement.
Hardware Requirements
An AT&T Answer-Record 2500 or a Code-A-Phone 2540 or equivalent announcement machine
is required for each delay announcement. The equipment requires a port on a ZTN78 Tip
Ring Line (or TN742 Analog Line) CP.
The system supports two Directed Night Service delay announcements.
Detailed connection information is provided in Figure 2-32.
Descriptions of the SIP (Station Interconnect Panel), TAE (Trunk Access Equipment), and
associated cables and adapters, as shown on the figures, are provided under the heading
“connectivity” in Section 4.
2-173
SYSTEM 25
CABINET
TN742
OR
ZTN78
PART OF
OCTOPUS
CABLE
— — —
C2
PART OF
SIP
●
SIP
ADAPT.
W1
— — —
B1
C5
DELAY
ANNOUNCEMENT
EQUIPMENT
— — —
LEGEND:
TN742 - ANALOG LINE CP
ZTN78 - TIP RING CP
B 1 - TYPICAL-103A CONNECTING BLOCK*
C2 - OCTOPUS CABLE (WP90780) PEC 2720-05P
C5 - MODULAR CORD (D4BU-87)
W 1- 4 PAIR INSIDE WIRING CABLE*
* - FURNISHED BY INSTALLER
Figure 2-32. Delay Announcement Equipment Connections (FCC Registered)
2-174
OFF-PREMISES STATIONS (OPS)
Description
An OPS is a single-line voice terminal that is located in another building and connected to
System 25 via arrangements with the local CO. The station has the same features as an onpremises single-line station except that it is counted as an outside party in a conference call.
Also, the Message feature will not operate with these sets.
Considerations
This service is sometimes furnished to executives at their residences. It allows them remote
access to System 25 features and services.
Interactions
●
Conference:
outside lines.
●
Distinctive Ringing: For V2 systems, Distinctive Ringing is not provided; OPS
stations will always receive standard (that is, single) ringing for calls.
For conference purposes, an OPS counts as one of the two allowable
Administration Requirements
Single-Line Voice Terminals (Station Port):
●
Assign port on Analog Line (TN742) CP
●
Make This an Extended Station (Yes/No) - Default = No. (This is how the system
knows the station is an OPS.)
Hardware Requirements
Requires a port interface on a TN742 Analog Line CP.
The OPS must be a FCC registered single-line voice terminal.
Connectivity information is provided in Section 4, “Hardware Description.”
2-175
OUT-OF-BUILDING STATIONS
Description
Single-line voice terminals and multiline 7300H series terminals may be directly connected
to the system even though they are not located in the same building. For 7300H series
terminals special In-Range Out of Building (IROB) units are used to protect the switch and
its users from lightning, power crosses, etc. Out-Of-Building Stations can access all system
features.
Considerations
Single-line voice terminals may be located at distances up to 24000 feet from the system
cabinets. Carbon protection devices are required for lightning and power cross protection.
Multiline voice terminals must be located within 2000 feet of the system cabinets (local
power required beyond l000 feet) and must have IROB protection devices. MET sets may not
be used for Out-Of-Building service.
Interactions
None
Administration Requirements
None
Hardware Requirements:
Out-Of-Building multiline voice terminals require IROB units. Single-1ine voice terminals
require carbon protection devices and must be connected to ports on the TN742 Analog Line
CP. Connectivity information is provided in Section 4, “Hardware Description.”
2-176
PAGING SYSTEM ACCESS
Description
Provides users with dial access or feature button access to paging equipment.
As many as three paging zones can be provided, each with its own access code. (A zone is
the location of paging loudspeakers, for example, conference rooms, warehouses, or
storerooms.) In addition, one access code can be provided to activate all zones.
A single-line or multiline voice terminal user (including the attendant) can access paging
equipment by dialing the zone access code. Multiline users can press a Flex DSS or DSS
button that has been programmed with the paging access code.
A paging zone maybe administered to be dial restricted. This restricts users from accessing
the equipment unless they are assigned an access button.
A PagePac* paging system may be used. Some paging systems require only one port
assignment to support all zones. Other systems may require separate ports for each zone.
Compatible PagePac paging systems include:
●
PagePac 20 (Powermate)
This is the smallest PagePac system. The basic system provides a single paging zone
with an input source for background music. An Auxiliary Trunk port interface is
required for this system.
●
PagePac 20 Control Unit and Zone Mate 9 (includes Common Control Unit)
Allows a user to dial one paging access code and then dial a single zone or all zones
code (l-9) to access a paging zone. This system is equipped with a Control Unit and
connects to a Ground Start or Loop Start trunk port.
PagePac equipment is easy to use. A user simply dials the paging access code and receives
Confirmation Tone. If the equipment provides just one zone, the user then makes the
announcement. If the equipment provides multiple zone access, the user, after hearing
Confirmation Tone, dials a code to access the desired zone(s) before paging.
Considerations
Paging is particularly useful when used in conjunction with the Call Park feature. When a
user is away from his or her location and receives a call, the call can be answered and
parked by another user. The called party can then be paged and told what extension number
to call to retrieve the parked call. The called party can then retrieve the call from any voice
terminal.
If PagePac multi-zone equipment is used, only one port assignment is required.
* Trademark of Harris Corporation Dracon Division
2-177
Interactions
●
Direct Inward Dialing: A DID call may access a paging code. This allows the
user to dial in and utilize the Paging System Access feature. Dial restricting the
paging code will block this interaction.
Administration Requirements
Special Feature Ports (Auxiliary Trunk Interface):
●
Assign Paging access code for each paging zone (maximum = 3) to be provided
●
Assign All Zones access code
●
Dial restrict zone (Yes, No).
Multiline Voice Terminals (Station Port):
●
Assign DSS button with paging zone access code as required.
Trunk Ports:
●
Requires a port on a ZTN76 Ground Start Trunk or ZTN77 Loop Start Trunk CP for
each port interface required. If the paging equipment requires a contact closure, a
port on a TN763 Auxiliary Trunk CP is required instead of the ZTN76 or ZTN77.
●
Assign Trunk Access Code
●
Dial restrict zone (Yes, No)
●
Assign other appropriate CO trunk parameters.
Hardware Requirements
Requires a PagePac or other compatible paging system. Also requires Auxiliary or CO trunk
ports.
Paging may occur in up to three zones or in alI zones at once. The recommended method of
interfacing is via CO trunks (either loop or ground start) using a PAGE-PAC 20 equipped
with Zone Mate 9; this requires only one trunk port interface.
The following PAGE-PAC units may be used:
●
PAGE-PAC-Family equipped with Common Control Unit (one zone only)
●
PAGE-PAC-Family equipped with Zone Mate 9.
Some customer-provided equipment may require separate ports on the Trunk CPs for each
zone. The ZTN76 Ground Start Trunk or ZTN77 Loop Start Trunk CP is the preferred
paging equipment interface. If the paging equipment requires a contact closure, a TN763
Auxiliary Trunk CP Pack must be used.
Detailed connection information is provided in Figures 2-33 and 2-34.
Descriptions of the SIP (Station Interconnect Panel), TAE (Trunk Access Equipment), and
associated cables and adapters, as shown on the figures, are provided under the heading
“Connectivity” in Section 4.
2-178
PART OF TAE
700A NETWORK
INTERFACE BLOCK
(110- OR 66-TYPE
CONNECTING BLOCK)
SYSTEM 25
CABINET
I
ZTN76
OR
ZTN77
W1
B
W1
B1
C5
†
PAGING
SYSTEM
LEGEND :
Z T N 7 6 - CO GROUND START TRUNK CP
Z T N 7 7 - CO LOOP START TRUNK CP
B - 3 TO 1 SPLITTER CABLE-CONNECTORIZED (OR6016) PEC 2720-06X
B1 - TYPICAL-103A CONNECTING BLOCK*
C 5 - MODULAR CORD (D4BU-87)
W1 - 4 PAIR INSIDE WIRING CABLE*
* FURNISHED BY INSTALLER
† PAGING SYSTEM - PAGE PAC 20 E/W ZONE MATE 9 - PROVIDES 9
PAGING ZONES, PLUS ALL-ZONE PAGING
Figure 2-33. Paging Equipment Connections
Registered)
2-179
Using CO Trunk Ports (FCC
SYSTEM 25
CABINET
25 PAIR DE CABLE
PART OF
66-BLOCK
TN763
A
❘❘❘❘❘ ❘❘❘❘❘
W1
278A
ADAPTER
W1
C5
B1
PAGING OR
DICTATION
SYSTEM
-V
D-181321
ZENER KIT
GRO
C6
48V DC
P1
GRD
LEGEND :
-48V DC
I
T N 7 6 3- AUXILIARY TRUNK CP
A - SINGLE-ENDED 25 PAIR CONNECTOR CABLE (A25D) (NOTE 1)
B1 - TYPICAL - 103A CONNECTING BLOCK*
C5 - MODULAR CORD (D4BU-87)*
C6 – SINGLE-ENDED MODULAR CORD (DYB4) (NOTE 2)
P1 - KS-22911, L1, POWER SUPPLY, 48 VOLT DC (NOTE 2)
W1 – INSIDE WIRING CABLE (4-PAIR)*
NOTES :
1. APPARATUS CODE D-181523 (PEC 62511) INCLUDES
66E3-25 BLOCK CONNECTOR AND CABLE B25A 15/DE.
2. APPARATUS CODE D-181524 (PEC 62512) INCLUDES C6, P1, 278A
ADAPTER AND ZENER KIT.
*
FURNISHED BY INSTALLER
Figure 2-34. Paging Equipment Connection to TN763 CP Using 278A Adapter
2-180
PERSONAL DIAL CODE (PDC)
Description
Each station is assigned a PDC. The user may log in the PDC at any other voice terminal,
and calls to the PDC will follow the user.
A PDC can be assigned to a convenience voice terminal (i.e., not associated with a particular
user) and to data terminals with modems. Digital data endpoints are assigned Data Dial
Codes (DDCs).
There are two types of PDCs:
●
PDCs: Assigned to users with their own voice terminal (referred to as the “home
terminal”). To use the PDC at another voice terminal, the PDC may be logged-in
there.
●
Floating PDCs (FPDCs): Assigned to employees who do not have their own voice
terminal and to visitors who will be receiving calls. Calls to an FPDC will ring at
the terminal where it is logged in, or, optionally, at the Attendant Console if the
FPDC is not logged in anywhere.
Up to 200 PDCs and 300 FPDCs can be assigned in a system.
Note: The following call types are station oriented. They do not redirect to an “away
terminal.”
●
Automatic
Intercom
Calls
●
Directed Night Service Calls.
●
DSS Calls (V1 only)
●
DGC Group Calls
●
Manual
Signaling
●
Message
Waiting
●
Personal Line Calls
Calls Placed to A PDC:
If the PDC is either logged in at its home terminal or is not logged in anywhere, a call to the
PDC will simply be directed to the home terminal and will receive that terminal’s normal
hunting or call coverage treatment.
If the PDC is logged in at another terminal, then that terminal is termed the “away
terminal.” There are two cases to consider when a call is placed to this PDC:
●
Call not placed from the away terminal (the general case)
●
Call placed from the away terminal (a special case).
(1) Call Not Placed from The Away Terminal (the general case): The call will first
be directed to the away voice terminal. Ringing will occur at the away terminal if it is an
on-hook single-line voice terminal or if it is a multiline voice terminal with an idle System
Access button.
2-181
A call unanswered at the away terminal will be directed back to the home terminal unless
one of the following busy conditions exists at the home terminal: (1) it is a multiline
terminal with both System Access buttons busy and with no idle call coverage receiver or (2)
it is a single-line voice terminal that is off-hook, has no idle Call Coverage receiver, and has
no idle hunt-to station. While either of these busy conditions exists at the home terminal,
the call will not return to the home terminal. Instead, it will continue to ring at the away
terminal until answered or timed out.
If either of the above two busy conditions ends at the home terminal while the call is
waiting at the away terminal, the call will be directed back to the home terminal. If the call
is sent back to the home terminal, it can be answered or it can receive the terminal’s
hunting or call coverage treatment. The call coverage treatment given to calls that are
returning from an away terminal differs in two respects from the treatment provided to
calls initially directed to the home terminal. First, the home terminal and its coverage
station receive ringing simultaneously, rather than having the coverage ringing delayed.
Second, coverage terminals will ring for calls returning to the home stations even if the call
coverage ring options of the home terminal are “no ring. ” Both of these call coverage
modifications expedite the answering of calls that are returning to the home station from an
away station.
Once the call is directed back to the home terminal, it is removed from the away terminal.
This is true even if the away terminal was busy but subsequently became idle after the call
was sent back to the home terminal.
(2) Call Placed From The Away Terminal: A call to a PDC from the station where it is
logged in will be directed to the PDC’s home terminal.
Calls Placed to a Floating PDC:
When a valid FPDC is dialed, the call will be directed to the terminal where the FPDC is
logged in and will be provided the coverage treatment administered for that terminal. If the
FPDC is not logged in and if the attendant position is that FPDC’s “home,” then the call
will be directed to the attendant position. However, if the FPDC call was placed from the
attendant position, then it will not be redirected to the attendant but will instead be
provided Reorder Tone. Finally, if the FPDC is not logged in, and if the attendant position
is not that FPDC’s “home,” then the calling party will receive Reorder Tone.
For non-DID calls, if an invalid FPDC is dialed, then the calling party will receive Reorder
Tone. If a DID call does not match any assigned number in the dialing plan, it will be
directed to the attendant or to Reorder Tone, as administered.
Considerations
The Personal Dial Code (PDC) feature provides flexibility for users and visitors. Visitors,
once assigned a FPDC, can inform callers and the attendant. Calls can then be directed to
the voice terminal where the FPDC is logged in. Calls to FPDCs not logged in may be
directed to the attendant for further handling.
2-182
Interactions
●
Call Coverage: Calls to a logged-in FPDC receive the call coverage of that
terminal. lJnanswered calls to a PDC at an away terminal return to the home
terminal and receive the home terminal’s call coverage treatment; they do not receive
the away terminal’s call coverage.
●
Direct Inward Dialing: In systems with DID service, PDCs, FPDCs, DGC group
numbers, DDCs codes, and facility access codes may match the last 2, 3 or 4 digits of
DID numbers. For example, the code matching DID number 555-2345 may be 45, 345
or 2345, depending on the system dial plan.
●
Direct Station Selection: If an attempt is made to program an FPDC (rather than
a PDC) on a Flex DSS button, Reorder Tone is received.
Administration Requirements
System:
●
Send DID calls to unassigned DID numbers to the Attendant - Default = Yes
●
Route calls to not-logged-in FPDCs to the attendant - Default = Yes
●
Add/Delete
FPDCs.
Hardware Requirements:
None
2-183
PERSONAL LINES
Description
Provides a dedicated outside line for multiline voice terminal users.
Unlike pooled facilities, which can be accessed via dial codes, Personal Lines can be accessed
only via a dedicated feature button, and provide both incoming and outgoing service. Up to
16 terminals may share a Personal Line (up to 8 in R1V1). Up to four parties may be offhook on the line at the same time (the line itself is the fifth conferee). When the line is
busy, its status LED lights at all terminals on which the line appears. Ringing may be
provided optionally to one or more of the terminals sharing the line.
For each Personal Line, one station is administered as the principal (owner). The call
coverage of that terminal determines the call coverage of the Personal Line.
Considerations
Personal Lines provide facilities to users who desire direct access to the exchange network.
In addition, Personal Line appearances are provided on the Direct Trunk Attendant Console
for general use trunks. Appearances of these lines may also be provided at selected
multiline voice terminals to ensure call coverage when the attendant is not available. DID
trunks cannot be terminated on Personal Line buttons.
Personal Lines provide direct access for callers, bypassing the attendant. In some cases,
they may substitute for DID service.
Interactions
●
Attendant Console, Direct Trunk: On the Direct Trunk Attendant Console
(DTAC), trunks are terminated as Personal Lines. The DTAC can accommodate a
maximum of 26 Personal Lines (24 is the practical limit).
●
Attendant Console, Switched Loop: Personal Lines cannot be terminated on a
Switched Loop Attendant Console.
●
Call Coverage: The call coverage of the principal station (owner) determines call
coverage for the line.
●
Call Park: A parked Personal Line is bridgeable by any user with a button
appearance of that line. Bridging on to the connection does not unpark the call. The
parked call will not return to the parking user.
●
Call Pickup: After a call is picked up from a Personal Line button, the called
terminal can still enter the call.
●
Direct Group Calling: The same trunk may be used as a Personal Line and also be
directed to a DGC group. If an incoming call is not answered by the DGC group
after a predetermined number of rings, ringing and LED flashing will be transferred
to all button appearances of the line (unless a DGC delay announcement is provided).
●
Hold: A Personal Line cannot be placed on hold if any other stations that share the
line are also off-hook on the line.
2-184
●
Line Selection (Prime Line Preference): Prime Line Preference may be
assigned to a Personal Line.
●
Pooled Facility: A Personal Line may also be a member of a pooled facility group.
●
Toll Restriction (see “Calling Restrictions”): A call over a Personal Line is
subject to the toll restrictions of the station on which the call was placed.
Administration Requirements
Voice Terminal (Station Port):
●
Assign Personal Line trunk number
●
Assign Personal Line feature button
●
Make this the Principal Station (Yes, No)
●
Enable Personal Line Ringing (Yes, No).
Hardware Requirements
Requires port assignments for each trunk interface to be provided and a button termination
on multiline voice terminals.
2-185
POOLED FACILITY - DIAL ACCESS
Description
Allows both multiline and single-line voice terminal users to access a common pool of trunks
by dialing a facility access code.
Up to 16 facility access codes can be assigned (one per trunk group). The codes can be one to
four digits in length. A group of similar trunks assigned the same access code is referred to
as a trunk group. Additional information is provided in the “Trunk Groups” feature
description.
After going off-hook on a System Access or Loop button, receiving system dial tone, and
dialing a facility access code, the user will be connected to an idle trunk. However, the
connection will not be made if the terminal is restricted from dialing this trunk group or if
dial access is restricted, in general, to trunks in the group. The LEDs associated with the
System Access button will be lighted, and the user may complete the call. Single-line users
do not receive LED indications of the status of the pool. An attempt to originate a call on a
busy facility will result in Reorder Tone (fast busy).
Considerations
Pooled Facility-Dial Access provides users of single-line terminals, or multiline voice
terminals without Direct Facility Access (FACILITY) buttons, access to the system’s pooled
facilities.
Interactions
●
Direct Inward Dialing (DID): Access to pooled facilities via DID is permitted.
This includes access to WATS, FX, tie trunks, private lines, dictation equipment, and
paging systems. This access is provided by selecting facility access codes so that they
will match DID numbers.
●
Outward and Facility Access Restriction (see “Calling Restrictions”): A
terminal can be denied dial access to some or all pooled facilities, or may be totally
restricted from making any outside calls.
●
Toll Restriction (see “Calling Restrictions”): Denies the use of pooled facilities
for certain toll calls, but does not block access to the pooled facilities.
Administration Requirements
Voice Terminal (Station Port):
●
Restrictions - Refer to “Calling Restrictions” feature description.
Trunk Port:
●
Assign facility access codes
●
A11ow dial access for facility (Yes, No).
Hardware Requirements
None
2-186
POOLED FACILITY - DIRECT ACCESS
Description
Allows multiline voice terminal users to access a common pool of trunks via a Direct Facility
Access (FACILITY) button.
Upon pressing a FACILITY button and going off-hook, a multiline voice terminal user is
connected to a common pool of outside trunks (i.e., CO, FX, WATS, tie). If the Status and IUse LEDs associated with the button light steadily, the user may complete the call. If no
idle trunk is available (facility busy indication), an attempt by the user to originate a call
will be denied and the I-Use indicator will be Off. A user requiring access to several
different trunk pools must have a separate FACILITY button for each pool.
If all trunks in a pool are busy, the Status LED will be lighted. The user may press
FACILITY and remain on-hook to receive the busy-to-idle reminder when a trunk becomes
available. The busy-to-idle reminder is a short burst of tone that will be heard when a trunk
in the pool becomes available. When Prime Line Preference is assigned to a FACILITY
button, the button must be pressed to invoke the busy-to-idle reminder, even though the IUse LED is lighted.
Refer to the “Trunk Groups” feature description for additional information.
Considerations
Pooled Facility-Direct Access provides easy access to the exchange network for users who
make many outside calls. The feature eliminates the need to dial a facility access code. In
addition, the associated status LED provides pool busy/idle status and the busy-to-idle
reminder.
Interactions
●
Automatic Route Selection (ARS): Multiline voice terminal users who have
pressed FACILITY to activate busy-to-idle reminder must wait until all queued ARS
users have been serviced.
●
Facility Access Restriction (see “Calling Restrictions”): A trunk group may
be reserved for a group of users by dial-access restricting the trunks. In this way,
only users who have a FACILITY button, a Personal Line appearance, or who use
ARS can use the trunks.
●
Line Selection (Prime Line Preference): Pressing a FACILITY button to invoke
the busy-to-idle reminder overrides Prime Line Preference.
●
Toll Restriction (see “Calling Restrictions”): Toll-restricted voice terminals
receive standard toll restriction treatment on all FACILITY buttons.
2-187
Administration Requirements
Voice Terminal (Station Port):
●
Assign Direct Facility Access (FACILITY) buttons.
Trunks:
●
Assign Facility Access Codes.
Hardware Requirements
None
2-188
POWER FAILURE TRANSFER (PFT)
Description
Provides service to and from the CO for a limited number of prearranged single-line v o i c e
terminals during a commercial power failure (or when voltage drops below 90 volts for
longer than 250 milliseconds) and during other service interruptions. Any loop start or
ground start trunk may be arranged to terminate at a specific station on a one-to-one basis.
When a failure occurs, these prearranged connections are made, bypassing the system and
connecting terminals directly to the CO trunks. System features and restrictions are not
available during this time.
The system supports up to four Emergency Transfer Units (ETUs). Each ETU can provide
up to five voice terminals with direct connection to CO trunks.
When the system connects to dial pulse trunks, only rotary sets may be used to support
Power Failure Transfer (PFT). When the system interfaces the CO via touch-tone trunks,
touch-tone single-line voice terminals are used as PFT stations.
When power is restored, the following will be restored to their previous state:
1. Night Service mode (on or off).
2. User-programmed Flex DSS numbers.
3. PDCs logged in at a ’’home station” or an ’’away station’’ remain logged in there.
4. If a voice terminal has been removed/not removed from a DGC group, the terminal
will remain in that state.
5. User-programmed Repertory Dialing numbers.
6. All system/station features programmed through system administration.
Considerations
Power Failure Transfer provides emergency incoming and outgoing telecommunications
service to a number of predesignated single-line voice terminals. This is particularly
important for organizations providing public services such as fire, police, medical, etc.
Interactions
None
Administration Requirements
None
Hardware Requirements
The 10B Emergency Transfer Unit (ETU) in Figure 2-35 supports up to five Power Failure
Transfer (PFT) sets and a DID make-busy function. Up to four ETUs can be supported for a
maximum of 20 PFT sets. The sets can be connected to selected Loop Start or Ground Start
2-189
trunks. If Ground Start trunks are used, a 55A1 Ground Start Button must be provided at
each PFT set.
Only FCC registered single-line voice terminals may be used for PFT stations. Rotary sets
must be used for dial pulse PFT trunks; touch-tone sets must be used for touch-tone PFT
trunks.
In the event of a Power Failure Transfer (switch has lost power or a major fault has
occurred) a contact closure is provided to the Central Office (CO) over a dedicated pair of
wires. The CO then makes busy all DID trunks. When power is restored, the closure is
removed and the CO restores DID service. External alarm contacts are provided on the
front of the ETU for use as required.
Note: It is recommended that customers with Disservice make provisions with their CO to
provide this arrangement.
The ETUs are mounted on the cross-connect backboard. Connections are via 25-pair
receptacle-ended (CO and SIP) and plug-ended (switch line and trunk) connectors. Modular
jacks are provided for the -48V control signal from the CPU (Call Processor Unit) and for
additional ETUs. Screw terminals are provided for the connection of external alarms.
When calculating Unit Loading (see Section 5, “Unit Loads”), all ETU loading counts against
Cabinet 1.
The 10B ETU is mounted on the cross-connect field as shown in Section 6, “Environmental
Requirements.”
ETU Power Failure Transfer connections are shown in Figure 2-36. Part (a) on the figure
shows a single-line voice terminal that has been connected as a Power Failure Transfer
station. In normal operation, the Call Processor CP supplies -48V dc to the ETU. The voice
terminal is connected through the ETU to the station port CP and can support all calling
activities. The trunk connection through the ETU to the trunk port supports normal trunk
calls.
Part (b) on Figure 2-36 shows the ETU connections when a Power Failure Transfer has
occurred. The transfer is initiated by the removal of the -48V dc to the ETU. All
connections through the system are dropped, and direct connections between PFT stations
and CO trunks are established. A contact closure toward the CO makes all DID trunks busy.
When the system is again able to process calls, normal service is automatically restored.
A multiple ETU arrangement is shown in Figure 2-37. As discussed earlier, separate -48V dc
control signals from the Call Processor are provided via legs 7 and 8 on Octopus cable C2.
The 25-pair cable from the Analog “Line” CP provides connectivity for eight voice terminals
at the Line input to the ETU. Since the ETU supports only five PFT stations, three of the
voice terminals are wired straight through the ETU and are not switched during service
interruptions. A similar condition exists for the 25-pair cable (D) from the CO Trunk CP to
the Trunk input of the ETU. Three of the eight trunk port appearances are wired straight
through the ETU to the CO and are not switched. Trunk ports connected by legs 2 and 3 of
the splitter cable are wired directly to the TAE Block.
2-190
AT&T
10B EMERGENCY
TRANSFER UNIT
(TRUNK)
SWITCH
(CPU)
ADDITIONAL
ETU
FOR PROPER OPERATION
THIS UNIT MUST BE
GROUNDED
Figure 2-35. 10B Emergency Transfer Unit (ETU)
2-191
SYSTEM 25
CABINET
TRUNK
PORT
ETU
25-PAIR
ETU TO SWITCH
(TRUNK)
— — — — —
-48V
CALL
—
ETU TO
CO
— — —
TO
ADDITIONAL
ETUs
ADDITIONAL
❑ ETU
ETU TO
ETU TO SWITCH
STATION
PORT
25-PAIR
OFFICE
25-PAIR
❑ CPU
PROCESSOR
CENTRAL
TAE
OCTOPUS
CABLE
(LINE)
—
—
—
—
—
—
—
—
—
PART OF SIP
858A
ADAPT.
UP TO FIVE POWER FAILURE
TRANSFER (PFT) STATIONS
(REGISTERED SINGLE-LINE)
(a) ETU CONNECTIONS (DASHED LINES) NORMAL OPERATION - (NO PFT)
(TRUNK SUPPORTS STANDARD CO CALLS)
ETU
TAE
-48V REMOVED
❑
25-PAIR
❑
TO PFT
STATION
(b) ETU CONNECTIONS (DASHED LINES) ON PFT
Figure 2-36. Emergency Transfer Unit Connections
2-192
CENTRAL
OFFICE
700A NETWORK
INTERFACE BLOCK
(66-, OR 11O-TYPE)
DIGITAL SWITCH
CO
TRUNK CP
NETWORK
ACCESS
CO
TRUNK CP
RJ2x1
CO
TRUNK CP
CO TRUNKS
CO
TRUNK CP
CENTRAL
OFFICE
CO
TRUNK CP
C2
ETU
(2)
CPU
GROUND
CO
TRUNK CP
ANALOG
STATION CP
CALL PROCESSOR
OCTOPUS CABLE
— — —
— — —
— — —
LEG 7
C1
C1
PART OF
CPU
ETU
(3)
ETU
(4)
CPU
GROUND
GROUND
●
LEG 8
●
●
PART OF SIP
-48V DC
TO SINGLE
-48V DC
POINT GROUND
PFT STATION
"
"
FROM
VOICE TERMINALS
(4 PAIR
BUILDING WIRE)
"
●
LEGEND :
A - SINGLE ENDED 25
PAIR CABLE (A25D)
B - 3 TO 1 SPLITTER
CABLE (OR 6016)
C1 - MODULAR CORD (D8W-87)
C2 - OCTOPUS CABLE (WP9078O)
D - 25 PAIR CONNECTORIZED
CABLE (B25A)
"
NON-PFT STATION
"
"
NOTES :
1. TRUNK AND STATION CONNECTIONS TO ETU 2-4 ARE SIMILAR TO ETU (l).
2. THREE OF EIGHT STATION LINES (FROM ANALOG STATION CP) AND THREE OF EIGHT TRUNKS
(FROM CO TRUNK CP) WHEN USED ARE "FED-THROUGH" ETU, THEY ARE NOT SWITCHED IN THE EVENT
OF A POWER FAILURE.
3. MAXIMUM ETU(S) = 4, MAXIMUM PFT STATIONS PER ETU = 5.
Figure 2-37. Multiple ETU Arrangements
2-193
PROGRAM
Description
Enables system users to store numbers for access by feature buttons or code dialing.
Multiline voice terminal users can program numbers on REP DIAL and FLEX DSS buttons.
Both multiline and single-line terminal users can store Personal Speed Dialing numbers.
FLEX DSS buttons provide access to inside extension numbers only; REP DIAL buttons and
Personal Speed Dialing are used for account codes and outside numbers (maximum of 28
digits and 25 digits, respectively).
Special Characters (VI)
The following special characters may be used in Repertory Dialing and Personal Speed
Dialing numbers on R1V1 systems only:
CHAR.
USED IN REPERTORY DIALING NUMBERS
*
(1) In the first character position, transmits an actual “*” (as the
first character of certain feature access codes).
(2) Otherwise, produces a l.5 second pause. (Since System 25 does not
have a Dial Tone detector, judicious use of the pause character will
help to ensure that intermediate Dial Tones are obtained before more
digits are sent.)
#1xx
Represents a System Speed Dialing code (where xx = 00-89); # is
allowed only in the first character position of the stored number.
#2x
Represents a Personal Speed Dialing code (where x = 0-6); # is
allowed only in the first character position of the stored number.
CHAR.
USED IN PERSONAL SPEED DIALING NUMBERS
*
Produces a 1.5 second pause. (Since System 25 does not have a Dial
Tone detector, judicious use of the pause character will help to ensure
that intermediate Dial Tones are obtained before more digits are
sent.)
#1xx
Represents a System Speed Dialing code (where xx = 00-89); # is
allowed only in the first character position of the stored number.
2-194
Special Characters (V2)
The following special characters may be used in Repertory Dialing and Personal Speed
Dialing numbers on R1V2 systems only:
CHAR.
USED IN REPERTORY DIALING NUMBERS
*
Produces a 1.5 second pause. (Since System 25 does not have a Dial
Tone detector, judicious use of the pause character will help to ensure
that intermediate Dial Tones are obtained before more digits are
sent. )
#*
Transmits an actual “*”.
##
Transmits an actual “#”.
#1xx
Represents a System Speed Dialing code (where xx = 00-89) or a
Virtual Facility code (where xx = 90-99). If using a Virtual Facility
code, it may appear only at the beginning of the stored number.
#2x
Represents a Personal Speed Dialing code (where x = 0-6).
#3
Marks the beginning of End-to-End Signaling. (System begins
transmitting touch-tone signals to the far end switch.)
CHAR.
USED IN PERSONAL SPEED DIALING NUMBERS
*
Produces a 1.5 second pause. (Since System 25 does not have a Dial
Tone detector, judicious use of the pause character will help to ensure
that intermediate Dial Tones are obtained before more digits are
sent.)
#*
Transmits an actual “*”.
##
Transmits an actual “#”.
#1xx
Represents a System Speed Dialing code (where xx = 00-89) or a
Virtual Facility code (where xx = 90-99). If using a Virtual Facility
code, it may appear only at the beginning of the stored number.
#3
Marks the beginning of End-to-End Signaling. (System begins
transmitting touch-tone signals to the far end switch.)
2-195
Programming Procedures
Program mode may be entered either by dialing a code (#4) or by moving the Test/Program
switch to P (on voice terminals so equipped). If the code is used to enter program mode, the
terminal remains in program mode until the user goes on-hook or a timeout occurs. If the
switch is used to enter program mode, the terminal remains in program mode until the
switch is returned to the midpoint between P and T; the system will send a single-ring
reminder every 60 seconds until the switch is repositioned.
Programming a number always removes the number that was previously stored in the same
location. If a user wants to remove an old number and not replace it with a new one, “ 0“
should be entered in place of the number to be programmed.
To Program a Number by Using a Code:
1. Lift the handset and listen for dial tone.
2. Dial #4.
3. Dial the number you wish to program.
4. Either:
— press the FLEX DSS or REP DIAL button,
or
— dial the Personal Speed Dialing code (#20 -#26)
to indicate where this number should be stored.
5. Listen for confirmation tone and dial tone.
6. Hang up.
To Program a Number by Using the Switch:
1. Slide the switch on the left side of the voice terminal to P.
2. Lift the handset and dial the number you wish to program.
3. Either:
— press the FLEX DSS or REP DIAL button,
or
— dial the Personal Speed Dialing code (#20-#26)
to indicate where this number should be stored.
4. Listen for confirmation tone and dial tone.
5. Hang up.
6. Slide the switch back to the midpoint between P and T.
2-196
Considerations
T h e P r o g r a m feature allows users to assign their own Personal Speed Dial/Flex
DSS/Repertory Dial numbers. This is particularly useful where working arrangements or
personnel responsibilities change.
Interactions
●
Users cannot place or answer calls while in the program mode. Incoming calls will
receive Busy Tone.
●
Should a number require more digits than are free in the common Personal Speed
Dialing/Repertory Dialing memory (approximately 5000 digits), Reorder Tone will be
returned after the indication of where this number is to be stored (see Procedures,
above).
Administration Requirements
Voice Terminal (Station Port):
●
Assign FLEX DSS and REP DIAL buttons
●
Allow Personal Speed Dialing on a per-station basis.
Hardware Requirements
None
2-197
RECALL
Description
Single-line voice terminal users can obtain System 25 Recall Dial Tone by pressing the
RECALL button (not all single-line sets have a RECALL button). Pressing RECALL is
equivalent to briefly pressing and releasing the switchhook (switchhook flash).
The RECALL button on a multiline voice terminal can be used, under specialized conditions,
to send a switchhook flash to the Central Office (for example, to access Centrex services).
However, it can never be used to send a switchhook flash to the System 25.
Considerations
None
Interactions
None
Administration Requirements
None
Hardware Requirements
None
2-198
REMOTE ADMINISTRATION INTERFACE
Description
Provides dial-up access to the system’s administration port, either for a standard system
administration terminal or for a PC running Advanced Administration software.
Both read and write capability is provided with access to all system translation and fault
tables. A remote administration terminal can perform the same functions as the onpremises SAT.
Remote Administration allows remote access to the system by maintenance personnel, the
System Administrator, and others.
Interactions
Only one System Administration Terminal can be connected at one time.
Administration Requirements
Depends on the connecting arrangements selected (see below).
Hardware Requirements
Requires a remote SAT.
Requires that port #1 of the Call Processor CP be connected to: (1) a dedicated modem and
dedicated facility (private line or CO trunk), or (2) a dedicated modem connected to a tip
ring station port, or (3) an ADU connected to a data line port. See the subsection titled
“System Administration” for additional information. Connectivity information is also
provided.
2-199
REPERTORY DIALING
Description
Allows multiline voice terminal users to store a telephone number, account code, or feature
access code in the system’s memory and associate that number with a REP DIAL button.
Pressing REP DIAL is equivalent to dialing the stored number. Individual numbers can be
up to 28 digits in length.
Programming the number is accomplished from the user’s voice terminal. Programming
procedures and other information can be found in the “Program” feature description.
Should the user attempt to enter more than 28 digits, Reorder Tone will be given.
The user can press REP DIAL under any of the following conditions:
1. When off-hook receiving Dial Tone
2. When off-hook on a call on which more dialed digits are expected
3. When off-hook on a call and connected to an outgoing trunk (End-to-End Signaling
might apply in this case)
4. After pressing ACCT ENTRY.
When REP DIAL is pressed, the button’s status LED lights briefly and then goes dark.
Considerations
Repertory Dialing simplifies dialing long or frequently called numbers, and allows onebutton access to many features.
Interactions
●
Account Code Entry: An Account Code may be stored on a REP DIAL button. The
REP DIAL button should be pressed at the point where the account code would
normally be dialed.
●
Calling Restrictions: A user can not use Repertory Dialing to access a number
that he or she is restricted from dialing.
●
Last Number Dialed (V2): A number called by pressing a REP DIAL button is
saved by the Last Number Dialed feature.
●
Speed Dialing: Numbers already stored as System Speed Dial numbers can also be
stored as Repertory Dial numbers. Storing a System Speed Dial code (#100 -#189) on
a REP DIAL button saves memory space (compared to storing the whole number
again on a REP DIAL button).
●
Virtual Facilities (V2): Virtual Facility codes can be stored on REP DIAL buttons.
2-200
Administration Requirements
Voice Terminal (Station Port):
●
Assign Repertory Dial (REP DIAL) buttons.
Hardware Requirements
None
2-201
SEND ALL CALLS (V1)
Description
Allows multiline voice terminal users whose calls are covered to temporarily direct some
incoming calls to coverage and turn off their ringers to these calls.
This feature is activated by pressing the SEND ALL CALLS button. It is deactivated by
pressing the button a second time.
Considerations
Send All Calls is a relevant feature only for stations that have some form of coverage
treatment. Activating the feature has no effect at stations with no coverage; it cannot be
used simply to create a “do-not-disturb” condition.
When Send All Calls is activated, full (repeated) ringing for incoming calls must o c c u r
somewhere:
●
If there is a coverage station that has an idle COVER button and is on-hook, then
ringing is sent to the coverage station; the call flashes at the sending station.
●
If there is no coverage station that has an idle COVER button and is on-hook, the
incoming call will remain at the sending station and ring, even though Send All Calls
is in effect.
When Send All Calls is in effect at a station, ringing at that station is not necessarily
canceled completely. A single-ring reminder for incoming calls is optional, assigned by the
System Administrator for each Send All Calls button. Single-ring reminder sounds under
the following conditions:
●
Send All Calls is activated, and a call goes to coverage.
●
A covering receiver is not available, and a call comes in while this station is off-hook;
this occurs even if single-ring reminder was not assigned.
The following types of calls always ring at a station, regardless of the
Calls:
●
Automatic Intercom calls.
●
Directed Night Service calls.
●
Calls to a PDC that is logged in at the station.
●
Calls returning to a DTAC on RTN-BUSY or RTN-DA buttons.
S tatUS
of Send All
Send All Calls must be assigned to a button that has a status light. The light turns on when
the feature is activated.
Interactions
●
Call Coverage: If no coverage station is available (that is, on-hook with an idle
COVER button) to accept a redirected call from a station with Send All Calls
activated, the call remains at the sending station and rings.
2-202
A covered station with Send All Calls activated will ring when called by its
individual coverage station if there is only one coverage receiver for this station.
●
Distinctive Ringing: Normal audible ringing is turned off for most incoming calls
when Send All Calls is activated, unless single-ring reminder is administered.
●
Line Status Indications: The line status LEDs still flash for incoming calls when
Send All Calls is in effect, even though normal ringing is cut off. The LED at the
sending station stops flashing when a call is answered by a covering station (LED
lights steadily) or dropped by the caller (LED goes dark).
●
Personal Line: Ringing on Personal Lines is turned off by activation of Send All
Calls only at the principal station. Calls continue to ring on non-principal
appearances and cannot be turned off by Send All Calls.
Administration Requirements
Voice Terminal:
●
Assign Send All Calls button
●
Assign single-ring reminder if required.
Hardware Requirements
None
2-203
SEND ALL CALLS (V2)
Description
Allows multiline voice terminal users to turn off their ringers and invoke a “do not disturb”
condition toward incoming calls. In addition, users who have coverage will have those calls
directed immediately to coverage, without the normal system ringing delay. Send All Calls
also allow-s covering users to temporarily remove their voice terminals from the coverage
path.
This feature is activated by pressing the SEND ALL CALLS button. It i S deactivated b y
pressing the button a second time.
Considerations
Send All Calls gives the user the option of having incoming calls sent directly to coverage or
making the terminal busy to incoming calls without sending them to coverage. The feature
is intended for occasional or temporary use.
Send All Calls must be assigned to a button that has a status light. The light turns on when
the feature is in effect.
The following types of calls always ring at a station, regardless of the status of Send All
Calls:
●
Automatic
●
Directed Night Service calls.
●
Intercom
calls.
Calls to a PDC that is logged in at the station.
Note: Calls to an FPDC do not ring when Send All Calls is in effect.
●
Calls returning to a DTAC on RTN-BUSY or RTN-DA buttons.
Send All Calls cannot be assigned to a SLAC. The Attendant Position Busy feature provides
a similar capability.
When Send All Calls is in effect at a station, and incoming calls are directed to coverage,
ringing at the sending station is not necessarily canceled completely. A single-ring reminder
for incoming calls is optional, assigned by the System Administrator for each Send All Calls
button.
On calls to non-busy stations where Send All Calls has been activated, the callers hear
ringing until a covering station answers, or, if the station is not covered, until the call is
dropped.
Interactions
●
Call Coverage (General): Send All Calls works in conjunction with the Call
Coverage features at covered and covering stations. At stations not associated with
Call Coverage, Send All Calls simply serves to silence the ringer on incoming calls; no
redirection occurs.
2-204
If a station is translated to not send ringing to coverage when calls to this station
are unanswered, the Send All Calls feature overrides this instruction (ringing will b e
sent).
If a covering station activates Send All Calls, the station is removed from the
coverage path completely. Coverage calls will not be directed to the station.
●
●
Call Coverage — G r o u p : If a station with Send All Calls activated has group
coverage and all the coverage receivers are busy, a call waits at the station while the
system checks for an idle receiver every five seconds. When one becomes available,
the call is directed to the covering station.
Call Coverage —Individual: If a station with Send All Calls activated has only
individual call coverage and all coverage receivers are busy, a call stays at the
station; it does not go to coverage.
●
Distinctive Ringing: Normal audible ringing is turned off for incoming calls when
Send All Calls is activated, unless single-ring reminder is administered.
●
Hands-Free Answer on Intercom: Activating Send All Calls will cause an active
AUTO ANS button to turn off. As long as the Send All Calls feature is in use, AUTO
ANS cannot be turned on.
●
Line Status Indications: The line status lights still flash for incoming calls when
Send All Calls is in effect even though normal ringing is cut off. The lights stop
flashing when the calls are answered by a covering station or dropped by the caller.
●
Personal Line: Ringing on Personal Lines is turned off by activation of Send All
Calls whether the station is the principal (owner) of the line or not. Personal Line
calls follow the coverage arrangements of the principal station. If the principal
station is not covered, the call will simply stay at the principal station until dropped
(even if other stations with that Personal Line have coverage).
Administration Requirements
Voice Terminal:
●
●
Assign Send All Calls button
Assign single-ring reminder if required.
Hardware Requirements
None
2-205
SPEAKER
Description
Allows 7300 H-series voice terminal users to turn On a built-in speaker. The speaker allows
on-hook dialing, group listening, and monitoring of call progress signals. The speaker is
turned on by pressing SPEAKER. Pressing the button at an idle terminal has the same
effect as lifting the handset: the user is connected to the selected line and hears Dial Tone.
An associated LED is lighted when the Speaker is On. Speaker volume may be adjusted by
the terminal’s volume control located on the left side of the set.
The speaker and associated LED are turned Off by pressing SPEAKER again or by lifting
the handset. The latter operation connects the handset to the associated voice channel.
When using the handset, pressing SPEAKER will turn On the speaker to support the Group
Listen feature; pressing SPEAKER again will turn Off the speaker and associated LED.
Note that once the user has lifted the handset, it is not possible to return to “hands-free”
operation. Hanging up the handset will terminate the call whether the speaker is On or
off.
Note: The built-in speaker provides one-way communication (listen only). The user must
pickup the handset to converse.
Considerations
The built-in speaker supports group listening, monitoring of calls (e.g., while waiting on
hold), and on-hook dialing.
Interactions
None
Administration Requirements
None
Hardware Requirements
Only 7300H-series (MERLIN) voice terminals with a SPEAKER button support this feature.
2-206
SPEAKERPHONE ADJUNCT
Description
The speakerphone adjunct permits users of voice terminals not equipped with built-in
speakerphones to place and receive calls without lifting their handsets. The adjunct has an
On/Off switch, a switch to temporarily mute the microphone, status lamps, and a volume
control (for incoming voice only).
All voice terminal features operate normally with the speakerphone adjunct.
Lifting the handset during speakerphone operation automatically turns off the speakerphone.
The speakerphone may be turned on during a call by pressing the On/Off switch and
hanging up the handset.
Considerations
Speakerphone operation allows users to perform other activities while carrying on a
conversation. Speakerphones also facilitate conference calls.
Interactions
●
Headset Adapter Adjunct: A voice terminal cannot have both a speakerphone
adjunct and a headset adapter adjunct.
Administration Requirements
None
Hardware Requirements
4A Speakerphone System
The 2500SM single-line voice terminal and 2991-type 10-Button MET set require a 4A
Speakerphone System. The 4A (Figure 2-38) provides a speaker and associated microphone,
indicator lamp and operating controls. The controls include a two position ON OR
QUIET/OFF rocker switch and a volume control.
The 4A Speakerphone requires an 85B1 power unit.
S101A/S102A Speakerphone (PEC 3163-HFU)
The S101A Speakerphone (Figure 2-38) can be used with the 12-Button MET Set (7203 M).
The S102A Speakerphone can be used with 7300 H-series voice terminals except the 5-Button
and HFAI sets.
The S101A/S102A speakerphones are equipped with a 4-foot connecting cord that plugs into
the voice terminal. Connecting cords are available in optional lengths of 18 inches and 14
feet.
The unit has a SPEAKERPHONE ON/OFF pushbutton switch and a MICROPHONE
ON/OFF pushbutton switch. The former controls the entire unit; the latter turns the
microphone on and off for privacy. Each button has an associated green status LED.
2-207
The S101A Speakerphone must be powered locally with a 2012D Transformer that plugs into
a 115V ac receptacle. Adjunct power supplies are described in Section 4, “Hardware
Description.” The S102A Speakerphone does not require supplemental power, except when
used with a 34-Button Deluxe voice terminal.
Detailed speakerphone adjunct connection information is provided in the following figures:
●
Figure 2-39–Speakerphone Connections for 7300H Series Multiline Voice Terminals
(Except 34-Button Deluxe)
●
Figure 2-40 —Speakerphone Connections for 34-Button Deluxe Voice Terminals
●
Figure 2-41 –Speakerphone Connections for 12-Button MET Sets.
/
—
ON LAMP
TRANSMITTER
SPEAKERPHONE
4A SPEAKERPHONE SYSTEM
SIOIA/S102A SPEAKERPHONE
Figure 2-38. Speakerphone Adjuncts
2-208
SYSTEM 25
CABINET
PART OF
OCTOPUS CABLE
— — —
ZTN79
HYBRID
LINE CP
●
—
PART OF SIP
SIP
ADAPT
C2
—
W1
B1
C1
VOICE
TERMINAL
T1
—
C8
—
—
—
—
—
—
LEGEND
:
B1
C1
C2
C8
T1
W1
-
*
- FURNISHED BY INSTALLER
S102A
SPEAKERPHONE
PEC 3163-HFU
TYPICAL-103A CONNECTING BLOCK*
MODULAR CORD (D8W-87) - FURNISHED WITH SET
OCTOPUS CABLE CABLE (WP90780) - PEC 2720-05P
SPECIAL CORD- F U R N I S H E D W I T H A D J U N C T
7300H SERIES VOICE TERMINALS EXCEPT 34-BUTTON DELUXE
4 PAIR INSIDE WIRING CABLE*
Figure 2-39. Speakerphone Connections For 7300H Series Multiline Voice
Terminals (Except 34-Button Deluxe)
2-209
SYSTEM 25
CABINET
PART OF
OCTOPUS CABLE
— — — —
ZTN79
HYBRID
LINE CP
●
C2
PART OF SIP
SIP
ADAPT
B1
Z400F
ADAPT
C1
VOICE
TERMINAL
T1
— — — —
C7
C8
— — — —
PWR
SUPPLY
P1
— — — —
S102A
SPEAKERPHONE
PEC 3163-HFU
LEGEND :
B1 - TYPICAL-103A CONNECTING BLOCK*
C1 - MODULAR CORD (D8W-87) - FURNISHED WITH SET
C2 - OCTOPUS CABLE CABLE (WP90780) - PEC 2720-05P
C8 - SPECIAL CORD - FURNISHED WITH ADJUNCT
T1 - 7305H02B VOICE TERMINAL (34-BUTTON DELUXE)
W1 - 4 PAIR INSIDE WIRING CABLE*
C7 -MODULAR CORD (D6AP-87)
PEC 62510
P1 -KS 22911 POWER SUPPLY
Z400F - ADAPTER
* - FURNISHED BY INSTALLER
Figure 2-40. Speakerphone Connections For 34-Button Deluxe Multiline Voice
Terminals
2-210
SYSTEM 25
CABINET
PART OF
OCTOPUS
CABLE
— — —
TN735
— — —
●
MET
●
PART OF SIP
C2
LINE CP
SIP
ADAPT
W1
B1
400B2
ADAPT
MET SET
T1
C1
— — —
C7
C8
248B
ADAPT
S101A
SPEAKERPHONE
PEC 31711
— — —
— — —
— — —
2012D
TRANS
LEGEND:
B1 - TYPICAL-103A CONNECTING BLOCK*
C1 - MODULAR CORD (D8W-87) - FURNISHED WITH SET
C2 - OCTOPUS CABLE (WP90780) PEC 2720-05P
C8 - SPECIAL CORD - FURNISHED WITH ADJUNCT
T1 - 7203M SET - 12-BUTTON MET SET
W1 - 4 PAIR INSIDE WIRING CABLE*
\
248B ADAPTER - MODULARIZES 2012D TRANSFORMER
400B2 ADAPTER - POWER ADAPTER
PEC 21691
2012D TRANSFORMER - 15-18V AC TRANSFORMER
C7 - MODULAR CORD (D6AP-87)
*
/
- FURNISHED BY INSTALLER
Figure 2-41. Speakerphone Connections For 12-Button MET Sets
2-211
SPEED DIALING
Description
There are two types of Speed Dialing: (1) System Speed Dialing, and (2) Personal Speed
Dialing.
System Speed Dialing:
Allows the System Administrator to store up to 90 numbers (maximum of 28 characters in
length) that are accessible by dialing 3-digit codes from any voice or data terminal.
Examples of typical System Speed Dial ring numbers include frequently-dialed DDD
numbers (together with leading facility access codes for WATS, FX etc. ) and account codes.
V1 Systems: The following special characters may be used in System Speed Dialing
numbers.
CHAR.
*
FUNCTION
Produces a 1.5 second pause. (Since System 25 does not have a Dial
Tone detector, judicious use of the pause character will help to ensure
that intermediate Dial Tones are obtained before more digits are
sent. )
V2 Systems: The following special characters may be used in System Speed Dialing
numbers.
CHAR.
FUNCTION
*
Produces a 1.5 second pause. (Since System 25 does not have a Dial
Tone detector, judicious use of the pause character will help to ensure
that intermediate Dial Tones are obtained before more digits are
sent. )
#*
Transmits an actual’’*”.
##
Transmits an actual “ #“ .
#lxx
Represents a Virtual Facility code (where xx = 90-99). This may
appear only at the beginning of the stored number.
#3
Marks the beginning of End-to-End Signaling. (System begins
transmitting touch-tone signals to the far end switch.)
A user cannot use a Speed Dialing number for which he/she is toll restricted, outward
restricted, or facility access restricted.
System Speed Dialing Operation: To place a call using a System Speed Dialing number,
the user goes off-hook and presses the # button on the dial pad followed by the 3 digit code
assigned to the desired number. The system interprets the associated stored number as if it
2-212
were dialed directly.
restriction.
This includes analysis of the number for the various types of
This feature can also be used when entering account codes. After pressing ACCT ENTRY or
flashing the switchhook and dialing “*O”, the user can enter a System Speed Dialing code.
The stored number associated with the code (the account code) will be listed in the SMDR
report.
Multilane voice terminals users may assign System Speed Dialing codes to Repertory Dialing
(REP DIAL) buttons.
Personal Speed Dialing:
Allows users to program up to seven Personal Speed Dialing numbers (maximum of 25
characters in length) that are accessible only from their terminals. The numbers are
accessed by dialing associated access codes (#20-#26).
Personal Speed Dialing is authorized on a per-station basis through System Administration.
The System Administrator will inform users if they can use this feature.
Refer to the “Program” feature description for more information about programming
Personal Speed Dialing numbers.
If enough storage space is available in memory to allow assignment of a Station Speed
Dialing number, Confirmation Tone will be returned after each number is programmed. If
not, Reorder Tone will be returned.
Note: Personal Speed Dialing is voice terminal oriented, not PDC oriented. A user
who logs in at another terminal cannot use his/her Personal Speed Dialing numbers.
Considerations
System Speed Dialing allows users to dial a number by simply dialing #100-#189. The stored
number associated with each code is (typically) a common-use phone number and is
programmed via System Administration.
Personal Speed Dialing allows users to program up to seven (private) numbers for their
personal use; these numbers can only be accessed from the terminal where originally
programmed.
The system will compare the restrictions applicable for the voice terminal against the
number associated with the Speed Dialing code, then allow or deny the call just as if the
number had been dialed directly from the terminal.
Interactions
●
Account Code Entry: System Speed Dialing codes may be used to store account
codes.
●
Calling Restrictions: A terminal that is restricted from placing a particular call
cannot avoid restriction by using the Speed Dialing feature.
●
Data Terminal Dialing: System Speed Dialing codes can be entered during Data
Terminal Dialing.
●
Last Number Dialed (V2): A number called with a Speed Dialing code is saved by
the Last Number Dialed feature.
2-213
●
Personal Lines: The Speed Dialing feature is not accessible from Personal Lines.
●
Repertory Dialing: Storing a System Speed Dialing code (#100 -#189) on a REP
DIAL button saves memory space, compared to storing the whole number again on
the REP DIAL button.
●
Speed Dialing: A Personal Speed Dialing number can include a System Speed
Dialing code only as the first four characters (but nowhere else). Personal Speed
Dial numbers cannot include Personal Speed Dialing codes. System Speed Dialing
numbers cannot include any Speed Dialing codes.
●
Virtual Facilities (V2): A Virtual Facility code may be used within Personal or
System Speed Dialing numbers. When used, it must appear at the beginning of the
stored number (first four characters).
Administration Requirements
System:
●
Assign System Speed Dial Numbers.
Voice Terminal (Station Port):
●
Allow/Deny Personal Speed Dialing on a per station basis.
Hardware Requirements
None
2-214
STARLAN NETWORK ACCESS (V2)
Description
The AT&T STARLAN NETWORK (STARLAN NETWORK) Access feature provides
connectivity between System 25 and a colocated STARLAN NETWORK. This connectivity is
provided by a combination of hardware and software elements.
The STARLAN INTERFACE circuit pack (ZTN84) is the principal hardware element
connecting System 25 and the STARLAN NETWORK. One or more of these circuit packs
m a y b e m o u n t e d i n t h e S y s t e m 2 5 c a b i n e t ( s ) . The STARLAN circuit pack (CP)
communicates with System 25 call processing over System 25’s Time Division Multiplex
(TDM) bus. To System 25, this circuit pack functions like a 4-port Data Line circuit pack
(DLC). To the STARLAN NETWORK, the STARLAN CP appears as a STARLAN
NETWORK workstation.
Communication between STARLAN NETWORK equipment (workstations, servers, hosts)
and data terminals, PC6300s, and host computers connected to System 25 is provided by
firmware on the STARLAN CP and communications program(s) on the PCs and hosts.
Two communications programs are available to users:
●
System 25 STARLAN NETWORK ACCESS (ACCESS)
ACCESS allows MS-DOS personal computers (PCs) connected (via the PC’s serial
port) to System 25 to communicate with DOS Servers on the STARLAN NETWORK
and to function as client workstations. The interface from the System 25 to the
STARLAN NETWORK is the STARLAN CP operating in bridge mode. Bridge mode
provides a transparent connection between the PC and the STARLAN NETWORK.
Personal computer users may access the STARLAN NETWORK just as though they
were connected to the STARLAN NETWORK with a Network Access Unit (NAU),
although at lower speed. (The NAU is a CP mounted in STARLAN NETWORK
workstations that permits access to other workstations and/or servers in the
network. ) Data transmission through the STARLAN CP is limited to a maximum of
9,600 bps. This is much less than the 1 million bps transmission rate between
workstations/servers on a STARLAN NETWORK.
Applications that require frequent and lengthy transfers of data over the Local Area
Network (LAN) will appear slow. Applications should be designed/configured to run
the executable program locally (on the PC) and to access data from the file server on
the LAN. ACCESS is recommended primarily for shared file and printer access.
Applications should be copied to the user’s (local) disk before they are run.
This program also permits STARLAN NETWORK access for remote PCs if the
System 25 is equipped with a Pooled Modem CP (TN758) or external modem pool.
Remote PCs can dial the STARLAN CP through a modem using either the Direct
Group Calling (DGC) feature or Direct Inward Dialing (DID) trunks to obtain a
connection through System 25 to the STARLAN NETWORK.
ACCESS must be used in conjunction with the AT&T STARLAN NETWORK Server
software (Version 2.0 or later). Installation software furnished with ACCESS
requires the STARLAN NETWORK client installation diskette in order to install
ACCESS.
2-215
●
Communications Access Manager (CAM)
CAM is an MS-DOS applications program that provides an enhanced calling interface
and terminal emulation for PCs connected to System 25 or a STARLAN NETWORK.
This connection must be through a DLC or a STARLAN NETWORK that is, in turn,
connected to System 25 by a STARLAN CP. Refer to the Communications Access
Manager (CAM) Program feature description for a more detailed description of the
program.
STARLAN INTERFACE Circuit Pack
The STARLAN INTERFACE CP (ZTN84) requires a single modular connection to the
STARLAN NETWORK (see Figure 2-42). It provides an interface between System 25’s Time
Division Multiplex (TDM) bus and STARLAN NETWORK’s packet switched network. The
STARLAN CP provides four full-duplex data connections at speeds up to 9,600 bits per
second.
The STARLAN CP operates in two modes: Gateway Mode and Bridge Mode. Gateway mode
supports connections from System 25 data terminals to STARLAN NETWORK UNIX®
system hosts, or from STARLAN NETWORK UNIX system hosts or client workstations to
System 25 hosts or modem pools. In Bridge Mode, the STARLAN CP passes the STARLAN
NETWORK’s Universal Receiver Protocol (URP) through System 25 to a local or remote PC.
This is referred to as Bridge Mode and provides a through connection between PCs running
ACCESS and a STARLAN NETWORK. The proper mode (Bridge or Gateway) is autoselected by the system.
HOST
COMPUTER
PERSONAL
COMPUTER
DATA
TERMINAL
DATA
TERMINAL
OR
PC WITH
MODEM
NETWORK
EXTENSION
UNIT
DATA
LINE
CP
STARLAN
CP
POOLED
MODEM
CP
●
●
●
STARLAN NETWORK
(WORKSTATIONS/SERVERS/HOSTS)
ANALOG
LINE
CP
TRUNK
CP
DATA
TERMINAL
OR
PC WITH
MODEM
SYSTEM 25
Figure 2-42. STARLAN NETWORK and System 25 Configuration
2-216
The STARLAN NETWORK View of System 25
From the STARLAN NETWORK, the STARLAN CP functions like a STARLAN NETWORK
workstation equipped with a Network Access Unit (NAU). The NAU enables STARLAN
NETWORK workstations and servers to access and exchange data over the network. Plug
number 1 of the STARLAN CP octopus cable should be connected to an “IN” jack on the
Network Extension Unit (NEU) (see Figures 2-43 and 2-44).
The System 25 View of a STARLAN NETWORK
From System 25, the STARLAN CP looks and functions like a TN726 DLC (with only four
ports). The STARLAN CP differs from the DLC in that, when it is dialed, the STARLAN
CP auto-answers the call and provides a second dialing prompt for completing the call to a
STARLAN NETWORK address. Depending on user’s data terminal type, the STARLAN CP
automatically selects the operating mode and enables the user to access and exchange data
over the network as described below.
Procedures for setting up connections (calls) between STARLAN NETWORK devices and
devices connected to System 25 vary, depending on both calling and called device. Generally,
a two-stage dialing procedure is used. The scenarios described below cover most situations.
Calls from System 25 to the STARLAN NETWORK
A. An MS-DOS PC connected to System 25 uses the STARLAN NETWORK ACCESS
software to run STARLAN NETWORK applications.
The PC may be connected either to a System 25 DLC port or to an analog station or
trunk port. (The latter arrangement uses System 25’s Modem Pooling feature. )
A typical call is as follows:
1. The PC user loads ACCESS and is automatically connected to the STARLAN
NETWORK. (The STARLAN CP phone number may be entered when
ACCESS is installed.)
2. The PC user may now access the STARLAN NETWORK just as if he/she were
a client connected to the STARLAN NETWORK through an NAU.
N o t e : Applications that are to be run frequently or are large (>10K
bytes) should be copied to the user’s disk before they are run.
B. A Data terminal user accesses a UNIX system host on the STARLAN NETWORK.
When the user dials the STARLAN CP, the CP answers in Gateway Mode and presents
the user with a “STARLAN Address” prompt.
A typical call is as follows:
1. The user dials the STARLAN CP.
2. The STARLAN CP provides the address prompt. The user enters the logical
name of the STARLAN NETWORK host (for example, 3B2).
3. The user is connected to the UNIX system host and receives the host login
prompt.
2-217
Calls from STARLAN NETWORK to System 25
A STARLAN NETWORK workstation accesses a host computer connected to System 25
(either a local host or a remote host that can be reached using the Modem Pooling feature).
1. The STARLAN NETWORK workstation loads CLIENT and NAUCOM and then
CAM software (discussed in the Communications Access Manager Program feature
description) and selects a directory entry for the host.
2. CAM communicates with the STARLAN CP to place the call.
3. After a connection message is received, CAM automatically switches to terminal
emulation (data) mode.
4. The user may now log into and converse with the remote host.
5. To disconnect, the user selects the CAM disconnect command.
Flow Control
Software flow control (XON/XOFF) may be enabled or disabled by System 25 data
endpoints. After the “STARLAN Address” prompt is returned to the user, a CONTROL-X
may be entered instead of a logical name. The user will be prompted further to enable or
disable flow control. After that, the user is again prompted for a STARLAN address. This
option also works for calls from the STARLAN NETWORK to System 25.
Data Call Disconnect
Data calls may be disconnected at either endpoint. Connections are dropped through the
normal disconnect procedures of each network. If a failure in the established connection
occurs, call disconnections are initiated from both sides.
Third-Party Call Setup
A data terminal (on System 25) or workstation (on the STARLAN NETWORK) can set up a
call between two other stations (voice or data) using the Third-Party Call Setup feature.
Since voice port/data port associations are not meaningful for STARLAN CP ports,
STARLAN NETWORK workstations must always specify the Personal Dial Code of the
source voice terminal or the Data Dial Code of the source data terminal. Note that this
feature can only be administered for the STARLAN CP ports as a group, and not for
individual STARLAN NETWORK workstations.
When placing voice calls using CAM, Third-Party Call Setup is used automatically.
Wiring
The STARLAN NETWORK wiring plan is based on standard 4-pair building wiring. The
STARLAN NETWORK uses two pairs of the 4-pair cable, allowing the remaining two pairs
to be used for voice service. STARLAN NETWORK data is transmitted over pairs two and
three. Figures 2-43 and 2-44 provide typical connection information.
A Y-adapter may be used to combine/split the pairs at the System 25 cross-connect field.
STARLAN NETWORK NAUs provide an RJ11 phone jack that terminates pair 1. Single
line sets may be plugged directly into this jack (Figure 2-43). MERLIN Communications
System sets require an ATL adapter and local power (Figure 2-44).
2-218
PART OF
SIP
C2
I
ZTN78
I
I
TN742
I
I
I
I
❘
I
NEU
(STARLAN HUB)
B1
A1
A2
❘
ZTN84
SYSTEM 25
W1
2500
SINGLELINE PHONE
C2
(LEG 1
ONLY )
IN
C1
STARLAN WORKSTATION
PHONE
IN
OUT
IN
C1
PC
NAU NEU ZTN84 ZTN78 TN742 A1 A2 B1 C1 C2 C5 W1 -
C5
NAU
NETWORK ACCESS UNIT - PEC 2614-100
NETWORK EXTENSION UNIT - PEC 2610-001
STARLAN CP- PEC 62518
TIP/RING LINE CP - PEC 62504
ANALOG LINE CP - PEC 63511
WP90851-L1 (Y ADAPTER) - PEC 2750-T05 (NOTE 1)
858A
ADAPTER
103A CONNECTING BLOCK*
MODULAR CORD D8W-87 (FURNISHED WITH NAU)
OCTOPUS CABLE WP90780- PEC 2720-05P (NOTE 1)
MODULAR CORD D4BU-87 (FURNISHED WITH PHONE)
FOUR PAIR BUILDING WIRING*
NOTE 1: C2 AND A1 ARE NOT REQUIRED IF NO PHONE IS
PLUGGED INTO THE NAU.
* FURNISHED BY INSTALLER
Figure 2-43. STARLAN NETWORK Connection to System 25 (With 2500
Single-Line Telephone)
2-219
PART OF
SIP
ZTN79
❘
C2
W1
B1
A1
A2
ZTN84
SYSTEM 25
ATL
PHONE
C2
(LEG 1
ONLY)
C1
C3
C4
T1
A3
IN
C1
NEU
PHONE
IN
OUT
IN
C1
PC
NAU
STARLAN WORKSTATION
NAU NEU ZTN79 ZTN84 A1 A2 A3 B1 C1 C2 C3 C4 T1 W1 NOTE 1:
NETWORK ACCESS UNIT - PEC 2614-100
NETWORK EXTENSION UNIT - PEC 2610-001
ATL LINE CP - PEC 62505
STARLAN CP - PEC 62518
“Y” ADAPTER WP90851-L1- PEC 2750-T05 (NOTE 1)
858A ADAPTER
ATL ADAPTER (NOTE 1)
CONNECTING BLOCK 103A*
MODULAR CORD D8W-87 (FURNISHED WITH NAU AND PHONE)
OCTOPUS CABLE WP90780 - PEC 2720-05P
6 INCH MODULAR CORD (PART OF A3)
7 FOOT MODULAR CORD (PART OF A3)
48 VOLT DC POWER SUPPLY KS22911 (NOTE 1)
FOUR PAIR BUILDING WIRING*
PEC 62520 INCLUDES A1, A3, AND T1
* FURNISHED BY INSTALLER
Figure 2-44. STARLAN NETWORK Connection to System 25 (With ATL-Type
Telephone)
2-220
ATL Adapters
The ATL adapter (KS23475) is a connection block that provides: 48V dc power from T1 (via
modular cords C4 and Cl) to the ATL phone, data connections from the STARLAN
workstation, and phone connections from the ATL phone (via modular cord Cl) over a
shared common cable (C3) to System 25. The phone and data connections are on separate
wire pairs that are split-out at the SIP “Y” adapter (Al). Phone wiring is cabled to the ATL
Line CP (ZTN79) by octopus cable C2, and data wiring is cabled to the NEU by modular cord
Cl.
Administrable Parameters
The STARLAN CP is administered as a type of data port. Some items administered on one
port are automatically administered for all four ports on the CP, others are individually
administrable.
Individually Administrable (default)
●
DDC of port
●
DDC to hunt to next (none)
●
Display ID
Common Administration (default)
●
CO trunk pool dial restriction (no)
●
Other trunk pool dial restriction (no)
●
Outward restriction (no)
●
Toll Restriction Class (none)
●
ARS FRL (3)
●
Restrict Third-Party Call Setup feature (yes)
Hardware Requirements
Requires a STARLAN INTERFACE CP. Each CP provides four interface ports between the
System 25 and the STARLAN NETWORK.
2-221
STATION HUNTING
Description
Provides linear, circular, or combinational hunting sequences for calls to busy single-line
voice terminals and data terminals.
Calls to a busy terminal may hunt to (only) one other terminal; however, up to five
terminals may hunt to the same terminal.
Although hunting is not available to or from multiline terminals, single-line terminals may
have their calls covered by multiline terminals.
Station Hunting takes precedence over Call Coverage. Calls to a single-line voice terminal
that is assigned both Station Hunting and Call Coverage will first hunt. If no hunted-to
station is available, the call goes to coverage.
The following are examples of the three types of hunting allowed:
●
Linear Hunting Example:
Terminals x, y, and z are arranged for linear hunting as follows: (1) Terminal x
hunts to Terminal y, (2) Terminal y hunts to Terminal z, and (3) Terminal z does not
hunt.
An incoming call to a busy terminal in the chain will hunt in one direction only.
Hunting will be toward the terminal that does not hunt.
●
Circular Hunting Example:
Terminals x, y, and z are arranged for circular hunting as follows: (1) Terminal x
hunts to Terminal y, (2) Terminal y hunts to Terminal z, and (3) Terminal z hunts to
Terminal x.
An incoming call to a busy terminal in the chain hunts in one direction until it finds
an idle terminal and then rings at that terminal. Any call coverage options assigned
to that terminal will then be invoked. If the hunt finds all terminals busy, it will
stop at the called terminal. Any call coverage options assigned to the called terminal
will then be invoked.
●
Combinational Hunting Example:
Terminals w, x, and y all hunt to Terminal z.
An incoming call to a busy w, x, or y Terminal will ring at Terminal z, and any call
coverage options assigned Terminal z will be invoked. If Terminal z is busy, the call
remains at the called terminal. Any call coverage options assigned the called
terminal will then be invoked.
Considerations
Station Hunting provides several flexible alternatives to ensure that calls do not go
unanswered. Note that only calls to busy terminals will hunt; once a call begins ringing at a
terminal it will remain there unless picked up or covered.
2-222
Interactions
●
Attendant Camp-On: When the attendant extends a call to a busy terminal in a
hunt group, the call hunts for an idle terminal. If none is found, the call Camps-On
to the called terminal.
●
Call Coverage: Station Hunting initially overrides all call coverage options. When
a call to a voice terminal that has Call Coverage exhausts the terminal hunting
possibilities, call coverage is invoked.
Administration Requirements
Voice or Data Terminal (Station Port):
●
Assign PDC/DDC of terminal to hunt to next.
Hardware Requirements
None
2-223
STATION-TO-STATION MESSAGE WAITING
Description
Allows a multiline voice terminal to turn On a Message Waiting LED located on another
multiline voice terminal by assigning a MSG WAIT button on each terminal.
Pressing MSG WAIT causes the LED on the signaling and signaled terminals to light. This
feature allows a user to inform another user that they have a message for them. Of course,
other arrangements can be made as to the meaning of the signal.
A subsequent MSG WAIT press (at either terminal) turns Off both LEDs.
No talking path is associated with this feature.
Considerations
This feature is functionally similar but separate from the Call Coverage Message Waiting
and Attendant Message Waiting features. Note that this feature does not light the Message
LED; the status LED next to the assigned MSG WAIT button is lighted.
Interactions
None
Administration Requirements
Voice Terminal (Station Port):
●
Assign MSG WAIT on both multiline stations.
Note: The MSG WAIT button must always be assigned to pairs of stations, and
works just between the two stations. If station “A” wants to signal stations “B” and
“C”, station “A” needs a separate MSG WAIT button for each.
Hardware Requirements
None
2-224
SYSTEM ADMINISTRATION
Description
The software that controls System 25 operation consists of tables located in system memory.
These tables contain data associated with:
●
Trunk, Station, and Auxiliary Equipment Ports
●
System Parameters
●
Direct Group Calling Groups
●
Toll Calls Allowed Lists
●
Peripheral Equipment Data Communications Parameters
●
Automatic Route Selection.
Collectively, these software tables are referred to as translations. The system comes
equipped with default translations data; when full-default cold started, the default
translations are copied into translation memory.
System Administration is the process of managing the translations by making changes to
modify system operation to meet customer requirements.
The System 25 Implementation Manual describes how a system can
specific customer needs. Information about a desired configuration is
forms that are used when entering the initial system translations
system). These forms are filed in the Administration Records Binder
for on-going record keeping.
be configured to meet
recorded on a set of
(i.e., initializing the
and provide the basis
Modification of initial assignments can be made to meet changing customer needs.
The system provides an EIA RS-232 interface to a System Administration Terminal (SAT),
the primary means of entering and modifying translations.
System 25 administration consists of:
●
Centralized
Administration:
Configuration of the system and assignment of
feature-related parameters, i n c l u d i n g a s s i g n m e n t o f f e a t u r e b u t t o n s o n v o i c e
terminals. Centralized Administration is performed via the SAT.
●
Advanced Administration: The Advanced Administration Software (AAS)
package is a major improvement in system management. It provides the System
Administrator a user-friendly, powerful tool for accurately and quickly making
changes in voice/data terminal assignments, call coverage, access codes, and other
system functions such as ARS.
The software operates on a PC6300 (with 512K ram) and provides an alternative to
use of the SAT input terminal.
A main menu gives the user ready access for these tasks:
— Adding/changing/removing voice station assignments
— Adding/removing users to and from coverage groups
— Saving translations.
2-225
A significant advantage of the AAS package is that it can be used either at the same
location or can be used remotely via a dial-up connection.
Procedures for performing Centralized and Advanced Administration are provided in System
25 Adrninistration Manual and System 25 Advanced Administration Manual, respectively.
Considerations
For RlV2, the default system administration password (systemx5) can be changed through
a new administration item. Note, however, that a system warm start or
the password to the default.
Hardware Requirements
The System Administration Terminal (SAT) is a Model 703 Data Terminal (see Figure 2-45).
It is a general purpose asynchronous full duplex printing data terminal with an RS-232
interface for data entry and retrieval. It provides a paper record of all transactions. When
located within 50 feet of the system cabinets, it can be directly connected to channel 1 on the
ZTN82 or ZTN128 Call Processor CP. Either on-premises or off-premises access to the
administration port is supported. The terminal operates at a speed of 1200 bps (1200 baud).
The Model 703 requires 115V ac 60-hertz commercial power from a 3-wire grounded outlet.
The terminal should be located on a flat surface such as a desk or table top. It is
approximately 12 inches wide, 9 inches long, and 3 inches high.
The Model 703 keyboard generates ASCII codes. The terminal produces two audible tones to
indicate the completion of activities.
●
●
Short Tone–A tone of less than one half-second indicates the normal termination of
an operation
Long Tone–A one-second tone indicates that an error or an abnormal operating
condition has been detected.
The Model 703 SAT Supplement contains a complete set of operating instructions for the
Model 703 Data Terminal. This document may be of use to customers who want to use the
terminal for other purposes in addition to system administration. All the information
needed to use the terminal as a SAT is included in the System 25 Administration Manual.
The SAT can be connected to the system cabinets in several different ways:
●
●
A direct connection within 50 feet when sharing the same AC outlet as the system
cabinets
A direct on-premises connection at a distance greater than 50 feet from the system
cabinets
●
A direct off-premises connection via the Central Office (OPS or CO trunk)
●
An on-premises switched connection
●
An off-premises switched connection.
Maximum cabling distances from the system cabinets are provided in Section 5, “Technical
Specifications.”
Installation details are provided in the System 25 Installation and Test Manual.
The SAT may also be provided by the customer. It must be a RS-232 compatible terminal
that has a 25-pin connector providing signal on DTR (pin 20). In addition, it should have the
2-226
following
characteristics:
Display: The minimum display size is 16 lines by 80 columns. The port provides both
carriage return and line feed characters to position the cursor at the start of the next line.
Destructive scrolling is also expected (new lines added at the bottom of the screen and topmost lines disappear). Full duplex operation is required. Alphabetic ASCII characters in
both upper-case and lower-case will be sent to the SAT, along with ASCII numerals and
some basic ASCII symbols. The device used must be capable of displaying ASCII alphabetic
characters when either upper-case or lower-case characters are received. However, upperto-lower case mapping (or vice-versa) for display is acceptable since no meaning is associated
with case.
Keyboard: The administration port requires ASCII alphanumeric characters as well as
some symbol characters. If the keyboard generates only upper-case or only lower-case
alphabetic characters the administration port will respond appropriately, since upper and
lower case input is considered identical. The SAT should be capable of sending the following
ASCII Characters:
A-Z or a-z
0-9
* , #
.
?
BACKSPACE
RETURN
“ (V2)
The data transfer rate is set when a carriage return character is received by the
administration port. There are two supported transfer rates: 1200 bps and 300 bps.
Refer to the System 25 Administration Manual for administration procedures and additional
information.
SAT Connection Information
Detailed connection information is provided in the following figures:
●
Figure 2-46–SAT On-Premises Direct Connections (Sharing Same AC Outlet)
●
Figure 2-47—SAT On-Premises Direct Connections (Greater Than 50 Feet from
System Cabinet)
●
Figure 2-48—SAT On-Premises Switched Connections
●
Figure 2-49—SAT Off-Premises Direct Connections
●
Figure 2-50—SAT Off-Premises Switched Connections
Descriptions of the SIP (Station Interconnect Panel), TAE (Trunk Access Equipment), and
associated cables and adapters, as shown on the figures, are provided under the heading
“Connectivity” in Section 4.
Maximum cabling distances from the system cabinets to the SAT are provided in Section 5,
“Technical Specifications.”
2-227
Figure 2-45. Model 703 System Administration Terminal
2-228
SYSTEM 25
CABINET
ZTN82
OR ZTN128
CALL
PROCESSOR
PART OF
OCTOPUS
CABLE
— — —
— — —
C2
●
PART OF
SIP
Z210A
ADAPT.
— — —
C1
355A/AF
SYSTEM
ADMINISTRATION
TERMINAL
— — —
LEGEND :
C1 - MODULAR CORD (D8W-87) - PEC 2725-07G
C2 - OCTOPUS CABLE (WP90780) - PEC 2720-05P
355A ADAPTER RS 232 PLUG TO MODULAR JACK - PEC 2750-A24
355AF ADAPTER RS 232 RECEPTACLE TO MODULAR JACK - PEC 2750-A25
Figure 2-46. SAT On-Premises Direct Connections (Sharing Same AC Outlet)
2-229
SYSTEM 25
CABINET
ZTN82
OR ZTN128
CALL
PROCESSOR
PART OF
OCTOPUS
CABLE
———
C2
PART OF
SIP
Z210A
ADAPT.
●
C1
355AF
ADAPT.
C3
C1
Z3A4
ADU
— — —
PART OF
SIP
— — — — — — —
SYSTEM
ADMINISTRATION
TERMINAL
❘
Z3A1
ADU
C1
B1
W1
C4
❘
SIP
400B2
❘
ADAPT.
ADAPT. ❘
❘— — — — — — —❘
C7
2012D
TRANS.
LEGEND :
248B
ADAPT.
B 1 - TYPICAL-103A CONNECTION BLOCK*
C 1- MODULAR CORD (D8W-87) - PEC 2725-07G
C 2 - OCTOPUS CABLE (WP90780) - PEC 2720-05P
C 3 -EIA CROSS OVER CABLE (M7U-87) - PEC 2724-30C
C 4 - ADU CROSS-OVER CABLE (D8AM-87) - PEC 2724-38X
W1 - 4 PAIR INSIDE WIRING CABLE*
355AF ADAPTER RS-232 RECEPTACLE TO MODULAR JACK - PEC 2750-A25
Z3A1 ADU - EQUIPPED WITH A THREE FOOT PLUG-ENDED EIA CORD - PEC 2169-001
Z3A4 ADU - EQUIPPEO WITH A 3 FOOT RECEPTACLE ENDED EIA CORD
C 7 - MODULAR POWER CORD (D6AP-87)
248B ADAPTER - MODULARIZES 2012D TRANSFORMER
PEC 21691
400B2 ADAPTER - P O W E R A D A P T E R
2021D TRANSFORMER - 15-18V AC TRANSFORMER
* - FURNISHED BY INSTALLER
Figure 2-47. SAT On-Premises Direct Connections (Greater Than 50 Feet From
System Cabinet)
2-230
SYSTEM 25
CABINET
PART OF
OCTOPUS
CABLE
— — —
ZTN82
OR ZTN128
CALL
PROCESSOR
C2
— — —
Z210A
ADAPT.
C1
355AF
ADAPT.
C3
PART OF
SIP
—
—
—
— — — —❘
— — — ❘
C2
400B2
Z210A
●
❘ ADAPT.
❘
ADAPT.
— — — ❘— — — — — — —❘
(PORT X)
TN726
DATA
LINE
TDM
BUS
●
PART OF
SIP
Z3A4
ADU
C1
C7
2012D
TRANS.
— — —
TN726
C2
DATA
LINE
●
248B
ADAPT.
PART OF
SIP
SIP
ADAPT.
W1
C1
B1
SYSTEM
ADMINISTRATION
TERMINAL
(NOTE)
Z3A1/A4
ADU
— — —
LEGEND:
B1
C1
C2
C3
W1
355AF
Z3A1 ADU
Z3A4 ADU
248B ADAPTER
400B2 ADAPTER
2021D TRANSFORMER
C7
-
TYPICAL-103A CONNECTING BLOCK*
MODULAR CORD (D8W-87) - PEC 2725-07G
OCTOPUS CABLE (WP90780) - PEC 2720-05P
EIA CROSS OVER CABLE (M7U-87) - PEC 2724-30C
4 PAIR INSIDE WIRING CABLE*
ADAPTER (RS-232 RECEPTACLE TO MODULAR JACK) - PEC 2750-A25
EQUIPPED WITH A 3 FOOT PLUG-ENDED EIA CORD - PEC 2169-001
EQUIPPED WITH 3 FOOT RECEPTACLE ENDED EIA CORD - PEC 2169-004
MODULARIZES 2012D TRANSFORMER
POWER ADAPTER
15-18V AC TRANSFORMER
MODULAR CORD (D6AP-87)
PEC
21691
* - FURNISHED BY INSTALLER
NOTE:
SYSTEM
ADMINISTRATION
TERMINAL
OR
MULTILINE
VOICE
TERMINAL
WITH
DATA
BUTTON
DIALS
PORT
X
ESTABLISH
DATA
CONNECTION
TO
ZTN-82.
DDC
Figure 2-48. SAT On-Premises Switched Connections
2-231
TO
PART OF
OCTOPUS
CABLE
— — —
SYSTEM 25
CABINET
ZTN82
OR ZTN128
CALL
PROCESSOR
— — —
CO OR PRIVATE
LINE CIRCUIT
CONNECT VIA TAE
PART OF
SIP
Z210A
–
ADAPT.
l
— — —
–
TYPICAL
MODEM
C1
— — —
–
SYSTEM
ADMINISTRATION
TERMINAL
355AF
–
TYPICAL
MODEM
LEGEND:
C1
C2
C3
355AF ADAPTER
- MODULAR CORD (D8W-87) - PEC 2725-07G
- OCTOPUS CABLE (WP90780) - PEC 2720-05P
- EIA CROSS OVER CORD (M7U-87) - PEC 2724-30C
- (RS-232 RECEPTACLE TO MODULAR JACK) - PEC 2750-A25
Figure 2-49. SAT Off-Premises Direct Connections
2-232
TELCO
CENTRAL
OFFICE
SYSTEM 25
CABINET
ZTN82
OR ZTN128
CALL
PROCESSOR
TN758
POOLED
MODEM
TDM
BUS
TN726
DATA
LINE
TN742
ANALOG
LINE
CO
TRUNK
FACILITY
PART OF
OCTOPUS
CABLE
— — —
C2
PART OF
SIP
Z210A
ADAPT.
●
— — —
C1
355AF
ADAPT.
C3
PART OF
SIP
— — — ❘— — — — — — —❘
C2
Z210A
400B2
●
ADAPT.
ADAPT.
— — — ❘— — — — — — —❘
———————
Z3A4
ADU
—
C1
❘
2012D
TRANS.
CONNECTED
AS OPS
OR CO
FACILITY
CO CABLE
C7
248B
—
ADAPT.
CENTRAL
OFFICE
❘ — — — — — — — — — — —❘
SYSTEM
TYPICAL
❘
ADMINISTRATION
MODEM
TERMINAL
❘
(212 TYPE) ❘
OR CO TRUNK
❘—OPS
— — — — — — — — — —❘
❘
LEGEND:
C1 C2 C3 OPS 355AF ADAPTER Z3A4 ADU 248B ADAPTER 400B2 ADAPTER 2021D TRANSFORMER C7 -
MODULAR CORD (D8W-87) - PEC 2725-07G
OCTOPUS CABLE (WP90780) - PEC 2720-05P
EIA CROSS OVER CABLE (M7U-87) - PEC 2724-30G
OFF PREMISES STATION
RS-23Z RECEPTACLE TO MODULAR JACK PEC 2750-A25
EQUIPPED WITH 3 FOOT RECEPTACLE - PEC 2169-004
MODULARIZES 2012D TRANSFORMER
POWER ADAPTER
PEC 21691
15-18V AC TRANSFORMER
MODULAR POWER CORD (D6AP-87)
Figure 2-50. SAT Off-Premises Switched Connections
2-233
SYSTEM MAINTENANCE
Description
The primary objective of System 25 maintenance is to detect, report, and clear troubles as
quickly as possible and with minimum disruption to normal service. This goal is supported
by periodic automatic diagnostic tests and fault detection hardware. System design allows
most troubles to be resolved to the circuit pack level.
System 25 hardware and software are organized as independent units or maintenance
objects. Each maintenance object is normally a separately replaceable unit. These units
include circuit packs, power units, fans, voice and data terminals, cross-connect hardware,
auxiliary, and peripheral equipment.
There are two general categories of system errors: system-detected errors and user-reported
problems. The system can automatically detect and log errors without human intervention.
For system-detected errors, an Alarm LED on the Attendant Console is lighted if the error
qualifies as a Permanent System Alarm (a serious error). Most alarms can be verified by
checking the LEDs located on the front edge of the system circuit packs. (At least one Red
LED will be on. ) User-reported problems are usually detected at individual voice and data
terminals and are often related to alarmed conditions.
Alarms may be retired automatically and can also be cleared manually. After a trouble has
been cleared, the system retests the previously faulty area. If the fault is no longer present,
the error message (and alarm, if applicable) is cleared. It is not necessary for maintenance
personnel to retire alarms after a problem has been fixed. However, they may clear error
messages and alarms by entering the proper commands at the System Administration
Terminal.
System Errors And Alarms
If a maintenance object fails periodic tests, the system automatically generates an error
record that is placed in one of three software tables (error logs). The failure may be
classified as a Permanent System Alarm or as an unverified failure that never becomes a
Permanent System Alarm. A Permanent System Alarm causes the Alarm LED on the
Attendant Console to light. This alarm indication is a signal to the attendant to contact
maintenance personnel.
System alarms are classified as:
●
Permanent
System
Alarms: Failures that cause degradation of service and
require immediate attention. These alarms cause the Alarm LED on the Attendant
Console to light and an alarm record to be stored in the Permanent System Alarm
error log.
●
T r a n s i e n t S y s t e m E r r o r s : Potential failures that may cause degradation of
service. These do not light the Alarm LED on the Attendant Console. These are
errors that have not been verified by system self-tests, and are not yet serious
enough to be classified as Permanent System Alarms.
If an error that begins as a Transient System Error is verified or reaches a threshold
level of severity, it is reclassified as a Permanent System Alarm.
Transient system errors are stored in the Transient System Error log. The system
can store a combined total of 40 Permanent System Alarms and Transient System
Errors in the error tables.
2-234
●
M o s t R e c e n t S y s t e m E r r o r s : The ten Most Recent System Errors are recorded
by the system, regardless of their level of severity. These are stored in the Most
Recent System Errors log.
Error Logs
The three error logs can redisplayed via the System Administration Terminal. The data in
the log is useful in diagnosing and analyzing troubles, particularly when the problem has not
yet caused an alarm or when alarms cannot be retired by replacement of maintenance
objects.
The error log is historical in nature. It lists faults that have not been resolved as well as
past alarms, and provides a profile of system maintenance.
Automatic Maintenance Tests
There are two kinds of maintenance testing initiated (only) by the system:
●
Periodic
●
Demand
Periodic tests are run by the system at fixed intervals. The tests do not affect service.
Demand tests are run by the system when it detects a condition requiring a need for testing.
Demand tests are only performed when errors are detected. Maintenance personnel cannot
initiate these tests.
For additional information, see AT&T System 25 Maintenance Manual.
2-235
TANDEM TRUNKING (V2)
Description
Tandem trunking provides an enhanced networking capability for System 25. With this
feature, tie trunks can be used to call through System 25 to reach another switching system
(CO or PBX ). Calls may be completed over on-network or off-network facilities.
To be treated as tandem trunks, tie trunks must be assigned trunk numbers beginning with
9. Incoming calls on these trunks may route out of System 25 over ground start, loop start,
or tie trunks. Tandem trunks can gain access to outgoing facilities either indirectly (by the
ARS feature) or directly (by dial access).
System 25’s Tandem Trunking feature does not support traveling class marks or centralized
attendant service.
Considerations
The use of tandem trunking with tie trunks provides a cost-effective alternative to toll
calling between branches.
Interactions
●
A u t o m a t i c R o u t e S e l e c t i o n : Tandem trunk calls that route outbound via ARS
receive the same treatment as calls originated by a System 25 station, with one
exception. If all facilities in a routing pattern are busy, call queuing is not provided.
In this case, busy tone is returned to the calling party.
The second digit of the trunk number is used to specify the trunk’s “ station” Facility
Restriction Level (FRL) for use with ARS. FRLs may be specified as follows:
Trunk Number
Range
9000-9199
9200-9399
9400-9599
9600-9999
I
Second
Digit
0orl
2or3
4or5
6, 7, 8,
or 9
I
Station
FRL
0
1
2
3
To gain access to an ARS routing facility, the tandem trunk’s “station” FRL must be
equal to or greater than the route’s FRL. Thus, a tandem trunk with an FRL of 0
has the least ARS privileges, while an FRL of 3 provides the most privileges. If the
restriction level of the tandem trunk is less than all route FRLs, reorder tone is
returned to the calling party.
●
Dial Access: No toll restriction is provided for tandem trunk calls. However, access
to outgoing facilities can be controlled via the “allow dial access” option in the
outgoing trunk’s administration. Tandem Trunk calls receive the following treatment
when attempting dial access of System 25 facilities:
2-236
— If the requested trunk pool is dial accessible, an outgoing trunk is selected
and the call proceeds normally.
— If the requested trunk pool is not dial accessible, reorder tone is returned to
the calling party.
Any attempt to dial an outgoing trunk pool by non-tandem tie trunks (that is, tie
trunks whose trunk number does not begin with 9) results in reorder tone being
returned to the calling party.
●
S t a t i o n M e s s a g e D e t a i l R e c o r d i n g ( S M D R ) : After accessing an outgoing
facility, the tandem trunk’s Facility Access Code (FAC) will be recorded in the STN
field on the call’s SMDR record. If the tandem trunk has no FAC, then the 4-digit
trunk number (9 X X X ) will be recorded in the STN field. All other SMDR fields are
unaffected.
Administration Requirements
Except for the need to specify the trunk number as described above, administration of
tandem trunks is the same as for any other tie trunk.
Hardware Requirements
Only dial-in tie trunks (types 1003-1008) may be used for tandem trucking.
2-237
TEST
Description
Provides users of the 7300H series voice terminals the ability to test their terminals.
Placing the Test/Program (T/P) switch in the “T’’position causes all red and green LEDs to
light alternately. The terminal also rings during the test.
Considerations
Test assures users that all LEDs are working and that the built-in speaker is functional.
The Test switch on some voice terminals is spring loaded; upon release, the switch returns to
a normal on-line position. On other terminals, the switch must be manually returned to the
center (normal on-line) position.
Interactions
None
Administration Requirements
None
Hardware Requirements
None
2-238
THIRD-PARTY CALL SETUP (V2)
Description
The Third-Party Call Setup feature allows a data terminal (the third party) in Command
Mode to set up a call between an on-premises voice or data terminal (the source) and
another voice or data terminal (the destination; can be on- or off-premises). Once the call has
been set up, the third-party drops off and is not included in the call.
Each third-party data terminal may be administered to have a particular source terminal
“associated” with it. This association allows an abbreviated form of dialing when activating
the Third-Party Call Setup feature. Through further administration, the third-party data
terminal may be given permission to set up calls for any source terminal, for only the
associated source terminal, or for no source terminal (feature disabled).
When the user successfully activates Third-Party Call Setup and has dialed all digits
correctly, the following occurs, depending on the source terminal type:
●
Voice
Terminal
(source)
The source voice terminal (if not busy) receives priority ringing. A priority ringing
cycle consists of two short bursts followed by one long burst. The source terminal’s
handset must be picked up within three ringing cycles; the destination terminal will
then be called. Regardless of the call outcome, the third-party data terminal displays
the message C O N F I R M E D and D I S C O N N E C T E D immediately after calling the
destination terminal.
If the source terminal’s handset is not picked up within three priority ringing cycles,
the third-party data terminal displays the messages N O A N S W E R a n d
D I S C O N N E C T E D . If the source phone is busy, the third-party data terminal
displays BUSY and D I S C O N N E C T E D .
A call to a source terminal that has the Hands-Free Answer on Intercom (HFAI)
feature activated results in the automatic answering of the source end, and the
destination terminal will be called. The third-party data terminal displays
CONFIRMED and D I S C O N N E C T E D .
●
Data
Terminal
(source)
If the source terminal is a data terminal whose speed is set to the highest optioned
speed of the data port, the message REMOTE SETUP is displayed at the s o u r c e
terminal and the C O N F I R M E D / D I S C O N N E C T E D message is displayed at the
third-party data terminal. The destination terminal will then be called automatically
from the source terminal. If the source terminal and destination terminal are
compatible, a data connection is established.
Since the System 25 does not provide call progress tone detection for an off-premises call
(can’t detect second dial tone, for example), pause characters should be inserted at
appropriate places in the dialed digit string. In addition, Third-Party Call Setup calls are
subject to the administered restrictions assigned to the source voice or data port. For
example, if the source terminal is restricted to on-premises calls only, a call to an offpremises destination terminal will be blocked.
2-239
Setting Up A Third-Party Call
To set up a call from the third-party data terrninal, the user selects <Voice call> from the
entry-level Command Mode menu (see Figure 2-51). The user then enters the characters as
required to call the destination terminal. Calls may be completed as follows.
Note: If a character is entered incorrectly, the ASCII character backspace (BS or
CTRL-H keys) or underscore (_) may be used to cancel a previously entered character.
Calling a Destination Terminal (source terminal is NOT associated with the thirdparty data terminal)
After the user has selected <Voice call> from the Command Mode menu, a DIAL: prompt
is displayed on the third-party data terminal. The user has 15 seconds to begin entering the
digits to be dialed before being disconnected. The format of the digits following the DIAL:
prompt is shown below:
DIAL: {Destination}F{Source}
The Destination number must include all digits required to call the destination terminal and
may contain facility access codes, speed dialing codes, and pauses (*). An “F” may be
entered immediately following the Destination digits; this character is used to separate the
Destination number from the Source number. The Source number must be a Personal Dial
Code when the source is a voice terminal, or a Data Dial Code when the source is a data
terminal. Floating PDCs are not allowed.
If the user enters the Destination number but not the Source number, the system prompts as
indicated below:
FROM: { Source}
The user must then enter the Source number.
Calling a Destination Terminal (source terminal is associated with the third-party
data terminal)
Following the DIAL: prompt the user enters the Destination number only. (System 25 will
automatically select the associated PDC or DDC as the Source number. )
If the third-party data terminal is permitted to establish calls for any source terminal, the
format {Destination}F{Source} must be used to set up calls for any terminal except the
associated source terminal.
Considerations
With this feature, computer-based telemarketing or other calling applications can set up
calls for the user.
2-240
Interactions
●
Call Coverage
Third-Party Call Setup calls to the source terminal will not be directed to a coverage
station. If the source terminal is not answered before coverage is invoked, the call is
dropped and the N O A N S W E R / D I S C O N N E C T E D messages are displayed at the
third-party data terminal.
●
Call Following
If calls to a source terminal have been forwarded to another terminal (via Call
Following), Third-Party Call Setup calls will be forwarded. Note, however, that the
maximum of three ringing cycles (combined cycles at the home and away terminals)
still applies before the system drops the call and displays the NO
ANSWER\DISCONNECTED
messages at the third-party data terminal.
●
Expert
Mode
Refer to the Expert Mode feature description for additional methods in dialing when
using the Third-Party Call Setup feature.
●
Station Message Detailed Recording (SMDR)
SMDR records will be generated for Third-Party Call Setup calls just as if they were
placed by the source terminal.
Administration Requirements
●
Data
Port
The user’s data terminal may be administered to have a particular source terminal
associated with it. This allows the abbreviated form of dialing when activating the
Third-Party Call Setup feature.
A source terminal may be associated via administration with several third-party data
terminals. A third-party data terminal, however, may be associated with only one
source terminal (voice or data).
In addition, the third-party data terminal can be administered so that calls may be
established for:
— Any voice or data source terminal
— One associated source terminal only
— No source terminals (feature disabled; default).
●
AT&T
STARLAN
NETWORK
Access
Ports administered on the STARLAN INTERFACE circuit pack may not have a
particular source terminal associated with them. Depending on the administration
parameters enabled, third-party data terminals on the STARLAN INTERFACE
circuit pack may establish calls for:
— Any voice or data source terminal
— No source terminals (feature disabled; default).
Hardware Requirements
None
2-241
TIE TRUNKS
Description
Provides a private communications link between System 25 and another PBX.
Incoming tie trunk calls may be directed to the attendant, to a voice terminal, or to a data
endpoint. Service may be either automatic, immediate dial, delay dial, or wink start. Dial
pulse or touch-tone signalings supported on both incoming and outgoing calls (and maybe
different for incoming and outgoing calls).
Considerations
Tie Trunks provide for efficient communications between company employees at different
locations. This provides a private network whose control and utilization can be managed.
Tie trunks can be administered for tandem trunking in R1V2. This arrangement enables
users to call through an intermediate System 25 to a remote System 25 or other PBX. Refer
to the “Tandem Trunking” subsection for more information.
Interactions
●
Conference: A tie trunk that is part of a conference counts as one of two allowable
outside parties.
●
Direct Group Calling: Only automatic incoming tie trunks can be directed to a
DGC group; however, they are not eligible for DGC delay announcement. Dial-in tie
trunks can access DGC groups, but are not eligible for queuing (that is, if the DGC
group is busy, Busy Tone is returned to the caller).
●
Night Service: Dial-in tie trunks cannot serve as Night Service trunks.
●
Personal Lines: When a dial-in tie trunk is assigned as a Personal Line and the
line is used for outgoing service at the same time that a call is coming in on the line,
the terminal may be connected to the incoming call even though the call is intended
for another terminal that shares the line. For this reason, it is recommended that
tie trunks not be assigned as Personal Lines.
Administration Requirements
Trunk Port:
●
Assign Trunk Type And Number
●
Assign Class Of Service Code - (0-15)
●
Assign Facility Access Code - Default = 102
●
Allow Dial Access (Yes, No) - Default = Yes
●
For Auto-in Type Only–Assign To DGC Group (Group Number l-32, or 0 for none) Default = 0
●
Make This a Directed Night Service Trunk (Yes, No) - Default = Yes
2-242
●
Assign Night Service Delay Announcement (1, 2, or 0 for none) - Default = 0
●
Dial-Inward Capability (Tone or Pulse) - Default = Pulse.
Hardware Requirements
Requires port interfaces on a TN760B Tie Trunk CP.
2-243
TOUCH-TONE AND DIAL PULSE SERVICES
Description
All single-line voice terminals and MET sets are equipped with touch-tone dial pads that
generate Dual Tone Multifrequency (DTMF) signals when a dial button is pressed. The
7300H series (MERLIN) voice terminals are equipped with touch dial pads that generate
digitally coded signals when a dial button is pressed.
Each pool of outside lines and each Personal Line maybe independently arranged for either
touch-tone or dial pulse service.
Touch-Tone Dial Pads
On outgoing calls on trunks requiring touch-tone signals, cut-through-dialing is provided.
Where the trunk requires dial pulse signals, conversion of the touch-tone signals to dial
pulses is provided until an end of dialing signal is detected. Cut-through is then provided,
and all subsequent digits are sent as touch-tone signals. See the “End-to-End Signaling”
subsection for more information.
Touch Dial Pads
On outgoing calls on trunks requiring touch-tone signals, all dialed digits are converted to
touch-tone signals. Where the trunk requires dial pulse signals, the dialed digits are
converted to dial pulses until an end of dialing signal is detected. Cut-through is then
provided, and all subsequent digits are converted to touch-tone signals. See the ’’End-to-End
Signaling’’ subsection for more information.
Considerations
None
Interactions
None
Administration Requirements
Trunk Port:
●
Assign trunk
specification.)
Class
Of
Service
(COS).
Hardware Requirements
None
2-244
(COS includes touch-tone/dial pulse
Description
Allows a user to transfer any call to another voice terminal.
A user can transfer calls either with or without announcement. A multiline terminal user
presses TRANSFER; the party is automatically placed on Special Hold (indicated by a
broken-flutter on the status LED of the call appearance button) and may receive Music-OnHold, if available. The system will automatically select an idle System Access button. The
user may dial the desired number or select another facility button and dial the call. The
user then can do one of two things: (1) hang up or (2) wait until the called party answers,
announce the call, and then hang up. The held call receives music if provided (and so
administered) until the transferring station hangs up, after which it receives ringback until
the transferred-to station answers. Unanswered transfers will receive the coverage
treatment of the transferred-to station.
A Personal Line transferred by a multiline voice terminal user will indicate the Special Hold
status at the transferring voice terminal until answered, and may be reentered if the call is
not answered. Reentering the call will automatically terminate the transfer attempt.
Single-line voice terminal users may transfer calls by flashing the switchhook, which puts
the caller on hold, listening for Recall Dial tone, dialing the second party, and going on-hook
either immediately or after announcing the call to the second party. A call may also be
transferred by setting up a conference and then hanging up.
Considerations
Transfer provides a convenient way to redirect a call to another voice terminal. Attendant
assistance is not required and the caller does not have to redial. While it is possible to
transfer a call without announcing it, it is recommended that call transfers always be
announced.
Interactions
●
●
Account Code Entry: A user may transfer a call to another user, then, instead of
hanging up, enter an account code. Subsequent account code entries will be ignored.
Attendant Console: In most cases, the attendant should not use this feature to
extend incoming calls, but should use the Start button or Selector Console instead.
The exception to this rule occurs if a trunk-to-trunk transfer is desired (see below).
●
Call Coverage: Coverage treatment of the transferred-to station is provided to
transferred calls. When a covering station transfers a covered call to another
station, the call will no longer appear at the covering station but will still appear at
the covered station. Note that if you attempt to transfer a call to a station that you
provide call coverage for, and that station does not answer, call coverage might n o t
be invoked. (This is one of the reasons why announced transfer is recommended. )
●
Call Pickup: A transferred call may be answered via Call Pickup.
●
Calling Restrictions: A non-restricted user (typically the attendant) can transfer a
CO trunk to an outward restricted or toll restricted station, giving the station
outward service. The toll restriction class of the transferring station will apply for
2-245
calls over a transferred trunk.
●
Hold: An outside call placed on hold during call transfer receives music-on-hold, if
available. A user attempting to return to a held internal call that has been
abandoned will hear nothing. A user attempting to return to a held CO trunk call
that has been abandoned hears CO dial tone or receives CO intercept treatment until
the CO disconnects.
●
Music-On-Hold:
For V2, a new administration item allows Music-On-Hold to be
enabled or disabled for “Special Hold.”
●
Trunk-To-Trunk Transfer: A trunk call may be transferred to another trunk.
Refer to the subsection on “Trunk-To-Trunk Transfer” for additional information.
Administration
Requirements
None
Hardware Requirements
None
2-246
TRANSFER TO DATA
Description
Allows multiline voice terminal users to originate or answer a call from their voice terminals
and then establish a data connection by transferring the call to a data terminal. This
feature was formerly called One-Button Transfer to Data (V1).
(Refer also to the overview of the system’s data features provided in the “Data Services
Overview” description.)
Data terminal calls can be set up from a multiline voice terminal with a DATA button. The
DATA button is associated by Data Dial Code (DDC) with a digital data endpoint. A
separate DATA button must be provided for each data terminal that the voice terminal can
transfer calls to.
The DATA button status LED provides status indications for the data endpoint:
●
Dark–Data
endpoint
is
idle
●
Winking–Data endpoint is reserved
●
Flashing–Data endpoint is being alerted to an incoming call
●
Steadily
Lighted–Data
endpoint
off-hook
(busy).
The DATA button status LED will wink only when a voice terminal reserves a data endpoint
by Data Call Preindication.
Data Call Origination Using Transfer to Data
A voice terminal user, after calling a DDC or a PDC (to reach an analog data endpoint)
receives either answer tone or called party answer, respectively. The user then transfers the
call to the associated data terminal by pressing DATA and hanging up. The called party
may also use Transfer To Data to transfer the call to a data terminal.
An inside call cannot be transferred via Transfer To Data until the far end answers.
If a handshake failure occurs after Transfer To Data, the data call will be disconnected and
the data terminal left in the idle (on-hook) state.
Note: Even if the associated data port is optioned for autobaud, the call will be set up
at the highest common speed that the calling and called data terminals are
administered for, independent of the current data terminal settings.
Data Call Preindication
A multiline voice terminal user, by going off-hook and pressing an idle DATA button, may
indicate that a data call will be attempted. This reserves the associated data port and a
modem pool conversion resource. This procedure is recommended when the data call is a
trunk call. The data port reservation is acknowledged by a winking status LED at the
DATA button. Subsequently, invoking Transfer To Data transfers the call to the associated
data terminal.
2-247
Preindication is canceled:
●
If the user goes on-hook before transferring the call to data
●
If the user preindicates on a second DATA button
●
If, after dialing is complete, a second DATA button is pressed. Preindication is
canceled for the first data terminal and the data call is transferred to the second
data terminal.
When Preindication is canceled,
reservation is canceled.
the associated pooled modem conversion resource
Interactions
●
Modem Pooling: If a conversion resource is required on an external call, invoking
Data Call Preindication will cause a pooled modem conversion resource to be
reserved. If none is available (e.g., the system has no Pooled Modem CP), Reorder
Tone is provided. (This will occur whether a conversion resource is actually required
or not.)
Administration Requirements
●
Assign DATA buttons on rnultiline voice terminals.
Hardware Requirements
None
2-248
TRUNK GROUPS
Description
Allows each trunk in the system to reassigned to one of up to 16 trunk groups.
Trunks link two switching systems, such as System 25 and the local CO or System 25 and
another PBX. Although not required, trunks can be grouped together in trunk groups
(sometimes referred to as pooled facilities) when all the trunks in the group perform the
same function. This grouping provides resource pooling that results in better service with a
given number of trunks. It also simplifies administration and calling. Calls are routed to
the appropriate trunk group; an idle trunk, if available, is selected from the group. Up to 16
trunk groups (pooled facilities) may be assigned in the system.
Several different kinds of trunk groups can be assigned in System 25:
●
Central Office (CO)—Provides a link with the local CO for calls except Direct Inward
Dial (DID) calls. Trunks classed as “CO” have a number of special characteristics.
●
Foreign Exchange (FX)—CO trunks that connect to a CO other than the local CO.
●
Wide Area Telecommunications Service (WATS)—CO trunks that connect to an
Outward WATS office or a dial 800 (in-WATS) Service office.
●
Direct Inward Dial (DID)—Provides incoming (only) service from the local CO.
These calls go directly to voice terminals instead of through the attendant.
●
Tie–Provides a link with another private switching system or network.
Trunk groups can be one-way (incoming) or two-way. Selection of the trunk group to be
used for a given call is determined by the initial digits of a dialed number (or by the ARS
feature ). These digits are referred to as the facility access code. Each trunk group is
assigned a unique code. Assuming an idle trunk in the selected group is found, a seizure
signal (service request) is sent to the distant switch. If the distant switch requires dialed
digits (as all but some tie trunks do), a signal (Dial Tone) is returned to System 25,
indicating readiness to accept dialed digits.
Trunk type refers to the physical design of a trunk circuit. The trunk types supported and a
brief description of each are given below. Refer to Section 3, “Functional Description” and
Section 9, “Glossary” for additional information.
●
Loop Start–A closure signal is sent through the loop formed by the trunk leads.
●
Ground Start–Similar to loop start but enhanced with ground signals.
●
Immediate Start—No start dial signals are used. On outgoing calls, the system waits
at least 80 ms after sending the seizure signal before sending the digits required by
the distant switch. This allows the distant switch enough time to attach a digit
receiver to the trunk. (Tie and DID trunks. )
●
Wink Start–A momentary signal (wink) is sent to the distant switch. (Tie and DID
trunks.)
●
Delay Dial–A steady signal is sent to the distant switch and is removed when ready
to receive digits. (Tie trunks only. )
●
Automatic–No start dial signals are used. The seizure signal sent or received is
sufficient to route the call. (Tie trunks only. )
2-249
Trunk groups connecting with a local CO, WATS office, or FX office can be ground or loop
start. DID trunk groups can be immediate or wink start. Tie trunks groups can be
automatic, wink start, immediate start, or delay dial.
Dual Tone Multifrequency (DTMF) signaling (touch-tone) or dial pulse signaling can be used
between the System 25 and the far end switch. System 25 can send or receive either type of
signaling required by the distant switch (DID trunks can only receive dial pulse signals).
The type to be used is specified when the associated trunk is administered.
An incoming call can be connected to another trunk, a voice terminal, a data endpoint, an
attendant console, or an announcement. When the call is answered, an off-hook indication is
sent to the serving office. This signal may be used to initiate the recording of call details
normally used for billing.
Trunks in a two-way trunk group should be translated (at the SAT) in the same order that
the serving office hunts when searching for an idle trunk. System 25 will then hunt in
reverse order. This reduces the probability that both switches will attempt to seize the same
trunk at the same time.
Considerations
Trunks of the same type and Class Of Service may be assigned a (Pooled) Facility Access
Code. This provides users with dial or button access to the trunk pool. Trunks may be dial
access restricted to reserve them for AILS and button access only.
Refer to “Recommended Central Office Trunk Facilities” (Section 5) for an estimate of CO
trunk requirements based on traffic considerations. See the “Pooled Facility-Button Access”
and “Pooled Facility-Dial Access feature descriptions for additional information.
Trunks may be reserved for incoming calls (e.g., sales or service department calls) by
specifying this in the (administered) Class Of Service code.
Interactions
●
●
●
●
Direct Group Calling: Most trunks may be administered so that incoming calls are
directed to a specified DGC group. For tie trunks, only the automatic-in type may be
so administered.
F a c i l i t y A c c e s s R e s t r i c t i o n ( s e e “Calling Restrictions”): Stations may be
restricted from dialing the CO trunk pool and/or all (fifteen) other trunk groups (as
a whole). Stations so restricted may still dial out if they are transferred to a trunk
by another station not so restricted.
Tie Trunks: Refer to the Tie Trunk description.
Toll
Restriction
(see “ C a l l i n g R e s t r i c t i o n s ” ) : When toll restricted stations
access FX, WATS, or Tie trunks, they are not toll restricted (i.e., toll restriction
applies to CO trunks only).
2-250
Administration Requirements
Trunk Port:
●
Assign Trunk Type And Number
●
Assign Class Of Service Code [DID - (l-4); Other - (0-15)]
●
Assign Facility Access Code - Default (See Note)
●
Allow Dial Access (Yes, No) - Default = Yes
●
Assign To DGC Group (Group Number 1-32, or O for none) - Default =
●
Make This a Directed Night Service Trunk (Yes, No) - Default = Yes
●
Assign Night Service Delay Announcement (Announcement 1 or 2 or 0 for none) Default = 0
●
Dial-Inward Capability (Tone or Pulse) - Default = Pulse (Tie trunks only).
✐
N o t e : Default Facility Access Codes are based on the CPs in a system.
Defaults are assigned as follows:
●
Loop Start Trunks - 100
●
Ground Start Trunks - 101
●
Tie Trunks - 102.
Hardware Requirements
Associated trunk port interfaces.
2-251
TRUNK-TO-TRUNK
TRANSFER
Description
Allows users to connect incoming trunk calls to other outside trunks and then hang up
(under certain conditions).
Incoming trunk calls may be transferred to another trunk, or conference with another
trunk. In all cases and at all times, either a System 25 station must remain in the
conference or one of the calls must be an incoming call on a ground start, DID or tie trunk.
The other call may be on any type of trunk and may be incoming or outgoing.
Considerations
Trunk-to-trunk transfer is particularly useful when an outside caller requests a transfer to
another outside number. For example, an employee can call in and have their call
transferred elsewhere. Note that as long as an inside station stays on the call (even if a
multiline station puts the call on hold and hangs up) any two trunks may be conference. If
the station drops out of the call, the trunk conference will be torn down unless the above
conditions are met.
If a System 25 station enters a trunk-to-trunk transfer call via a line appearance button for
one of the conference trunks, the call will still be broken down when one of the outside
parties hangs up.
A Direct Group Calling call that comes in on a ground start trunk and is answered at a
single-line set is not eligible for trunk-to-trunk transfer.
Interactions
●
Conference: Trunk-To-Trunk transfers may be set up using the Conference feature.
The conference must include an incoming trunk call on either a ground start, DID, or
tie trunk if it is to continue after all inside stations have dropped off.
●
Direct Group Calling: A DGC call that comes in on a ground start trunk and is
answered at a single-line set is not eligible for trunk-to-trunk transfer.
Administration Requirements
None
Hardware Requirements
None
2-252
USER CHANGEABLE OPTIONS (V2)
Description
User Changeable Options allows a data terminal user who is in the Command Mode to view
and change the settings of certain data port options. This feature is available to users of
Data Line CP ports; users of STARLAN Interface CP ports cannot change their port
options. Table 2-J contains brief descriptions of the user changeable options.
TABLE 2-J. User Changeable Options
Definition
Option
Speed
low, 300, 1200, 2400, 4800, 9600, 19200, autobaud
Parity
odd, even
Permit Mismatch
Allows two data endpoints to communicate at
different rates.
Local Echo
Determines whether characters from the data
equipment will be echoed by System 25 during
Command Mode.
Answer Text
Enables call progress messages to be displayed at
the called data endpoint.
Connection Indication
Determines whether users who have Command
Mode enabled will receive the “CONNECTED”
message when a connection has been made.
Recall Sequence
(disconnect)
Two short breaks or one long break; the sequence
used to disconnect a data call.
N o t e : The System Administrator may, under data port administration,
permission for users of specific data ports to self-administer these options.
deny
The user selects the Options menu from the Command Mode entry level menu. (Figure 2-51
illustrates all available Command Mode menus. ) The user now has the choice of viewing
options, changing options, or exiting the Options menu.
2-253
<eXit>
{Change
<eXit>
(Options
Display
table)
<300><1200><2400><4800><9600><19200>
<auto>
<eXit> <Yes> <No>
(Cancel
changes)
(Enable
changes)
DISCONNECTED
Figure 2-51. Command Mode Menu Tree
(Options
Display
table)
<Yes>
<No>
<eXit> <Break break> <Long break>
<eXit>
<eXit> <Yes> <No>
<eXit> <Answer text> <Conn ind> <Recall seq.> <Others>
<Undo> <Change options> <View options> <Enable options>
<low>
<eXit> <Yes> <No>
<eXit> <Odd> <Even>
DISCONNECTED
<View options>
<All>
<Hangup>
<eXit>
options>
<Options>
DIAL:
<Voice call>
<eXit> <Speed> <Parity> <permit Mismatch> <local Echo> <Others>
DIAL:
<Data call>
(ENTRY LEVEL)
Viewing Options
W h e n <View options> is selected, current and default values for the various data port
options are displayed, as shown below:
OPTIONS
Speed (highest)
Parity
Mismatch
Local Echo
Answer Text
Connect Indicat.
Recall Sequence
CURRENT
9600
Even
Yes
Yes
Yes
Yes
Br - Br
DEFAULT
19200 (Auto)
Even
No
Yes
Yes
Yes
Br-Br
<eXit>
<View options>
<Change options>
At this point the user can exit from the View Options menu, Change options, or View options
again (redisplays the Options table). If the user elects to exit, the terminal returns to the
Command Mode entry level menu.
Note: Typing the capital letter found within a menu will select that item and move the
user up or down the menu tree. For example, the user simply enters X or x (lowercase) to <eXit> the Options menu shown above and return to the entry level menu.
Changing Options—General
When on the Options Menu, the user selects < C h a n g e o p t i o n s > either by moving the
cursor (with the space bar) beneath < C h a n g e o p t i o n s > and pressing RETURN, or b y
typing the single letter code (c) associated with that item.
If the user selects <Change options> from the Options menu, the first half of the Change
Options menu is displayed, as shown below and on Figure 2-51.
<eXit>
<Speed>
<Parity>
<permit Mismatch>
<local Echo>
<Others>
If the user selects <Others>, the second half of the Change Options menu is displayed:
<eXit>
<Answer text>
<Corm ind>
<Recall seq. >
<Others>
If the user selects <Others> from the second half of the menu, the first half of the Change
Options menu is redisplayed. In this way, users can “toggle” back and forth between the
first and second halves of this menu.
Since these two lines are actually two halves of a single menu, users may select a particular
menu item while active on either half of the menu. For example, users who are active on the
first half of the menu may select <Answer text> by typing “a”.
Once a user has selected an option to be changed, a menu of valid settings for this option is
displayed (<Yes>, <No>, etc.). An “X” is displayed beneath the current setting of the
options, or beneath an option that may have been changed but not yet enabled. For all
options except <Speed> (see below), settings may be selected either by moving the cursor
(using the space bar) beneath the item desired and then pressing RETURN, or by typing the
single-letter code associated with that setting. The user is then returned to the Change
Options menu to make additional changes if required.
2-255
Changing Data Port Speed
The procedure for changing Speed settings is different from the procedure for changing the
settings of other options. Within the Speed menu, the user may find that several values are
marked with Xs. To change a speed, move the cursor beneath each value to be changed and
t y pe “+” to add the value or “-” to delete it. Once the new settings have been marked, press
RETURN to translate the plus and minus signs to their proper “X” values and then type “x”
to <eXit> from the Speed menu and return to the Change Options menu. For example:
The user enters the Speed menu and finds the following settings active:
<eXit>
<low>
<300>
X
<1200>
X
<2400>
X
<4800>
X
<9600>
X
<19200>
<auto>
To remove 9600 baud and activate autobaud, enter - under <9600> and + under <auto>, a s
shown below:
<eXit>
<low>
<300>
X
<1200>
X
<2400>
X
<4800>
X
<9600>
X-
<19200>
<auto>
+
With the cursor under any item except <eXit>, pressing RETURN provides the following:
<eXit>
< low>
<300>
X
<1200>
X
<2400>
X
<4800>
X
<9600>
<19200>
<auto>
X
Entering “x” now allows the user to <eXit> the Speed menu and return to the Change
Options menu. The user can make additional changes, as required.
When all of the changes have been made, the user should <eXit> the Change Options menu.
The following menu is then displayed:
<Undo>
<Change options>
<View options>
<Enable options>
From the above menu:
●
If the user selects <Undo>, the user is returned to the Command Mode entry level
menu, deleting any option-change requests.
●
If the user selects <Change options>, the Change Options menu is displayed and
the user can make additional changes as required.
●
If the user selects <View options>, the following menu is displayed:
CURRENT
9600
Even
Yes
Yes
Yes
Yes
Br-Br
OPTIONS
Speed (highest)
Parity
Mismatch
Local Echo
Answer Text
Connect Indicat.
Recal1 Sequence
REQUESTED
4800 (AUTO)
Even
Yes
Yes
Yes
Yes
Br - Br
The value in the CURRENT column indicates the current (active) status of the
option. The value in the REQUESTED column indicates the most recently entered
value (not yet enabled).
2-256
●
If the user selects < E n a b l e o p t i o n s > , the system incorporates the changes
requested and displays the message D I S C O N N E C T E D . If Autobaud is off, the
user must now press BREAK to return to Command Mode. If Autobaud is on, the
user must press BREAK and RETURN to return to Command Mode.
Note: If a user attempts to enable options during a “SAVE” operation by the system
administrator, the message “options changed FAILED” will be displayed. The user will
be returned to the top level of the Command Mode menu, and all change requests
discarded. After waiting a few minutes, the user may try again to change his/her data
port parameters.
Considerations
For those cases where a data terminal user accesses various data endpoints, each requiring
option changes, the User Changeable Options feature simplifies the process of administering
the data port to allow data call setup. A system administrator is not required to enter each
change.
Interactions
●
Expert Mode: See the Expert Mode feature description for an abbreviated method
of accessing Command Mode menus.
Administration Requirements
●
Data
Port
A data port must be administered to allow the terminal user to change options from
the Command Mode menu. If permission is denied, the user may view the current
option settings but not change them.
●
AT&T STARLAN NETWORK Access
The User Changeable Options feature is not available for ports administered as
STARLAN INTERFACE ports. Refer to the “STARLAN NETWORK Access” feature
description for additional information.
Hardware Requirements
None.
2-257
VIRTUAL FACILITIES (V2)
Description
A virtual facility (VF) is a call routing facility that is not defined by the physical facility
(trunk) over which calls are routed, but is instead defined by a combination of access codes,
authorization codes, and coded characters that allow special handling of the destination
telephone numbers. VFs can be used to automatically route calls via other carrier networks,
private networks, or tie trunks.
Up to ten virtual facility numbers (VFNs) may be administered. Each stored number may
be up to 28 characters in length and is associated with a code in the range of #190 to #199.
The first digits in a stored VFN must be the facility access code (FAC) for a physical trunk
group over which the call is to be routed. A series of digits and special characters are stored
following the FAC to define additional routes, Inter-Exchange Carrier (IXC) codes,
identification codes, or instructions concerning special handling of the destination telephone
number. When a VF has been defined using a particular trunk group, it has full access to all
trunks in that group. It is considered “busy” only when the physical trunk group is busy.
When a VF is dial accessed by a system user, calling restriction is based on the station’s
administered calling restrictions.
A system user may gain access to a VF by:
●
Dialing the VF code (#190-#199).
●
Storing the VF code as the first digits on a REP DIAL button. Other digits (for
example, the destination telephone number) may be stored following the VF code.
●
Dialing a System Speed Dialing code (#100-#189) that contains a VF as part of the
stored number. The VF code must be the first digits stored in the speed dialing
number.
●
Dialing a Personal Speed Dialing code (#20 -#26) that contains a VF as part of the
stored number. The VF code must be the first digits stored in the speed dialing
number.
●
Using Automatic Route Selection (ARS) and having a VF as the route selected by
ARS.
Note: The system can be administered to allow or restrict dial access for each VF
code. If dial access is restricted (system default), a VF may be accessed only when used
in an ARS routing pattern.
When virtual facilities are used in ARS patterns, they assume the same capabilities and
restrictions as physical facilities. For example:
●
Each VF may have a digit deletion and insertion scheme associated with it.
●
Selective restriction of a VF may be accomplished by assignment of Facility
Restriction Levels (FRLs).
If the VF is used in the first position of a routing pattern, calls may queue on it if all of the
routes are busy.
2-258
Whenever a VF is used to complete a call (either by dial access or through ARS), call
processing treats the number as a physical facility for Station Message Detail Recording
(SMDR) purposes. Thus, if VF code #190 is used to complete a call, the SMDR call record will
show “#190” as the facility used.
A VFN may contain up to 28 characters. The pound sign (#) is used as an escape character
within the digit string and indicates that the character following the pound sign requires
special interpretation. The following table defines the special characters that may be
included in a VFN.
CHARACTER
*
FUNCTION
1.5 second pause
#*
Transmit *
##
Transmit #
#3
End-to-End
Begin
transmission of
Signaling (system begins transmitting
touch-tone signals to the far end switch).
#5
Insert dialed digits (destination telephone
number) here. The destination telephone
number may be up to 16 digits in length
(21 if ARS digit translations have
occurred).
If #5 is used, it must be placed within the
last nine digits of the VFN. If #5 is n o t
used within a VFN, the dialed digits are
appended to the end of the VFN.
Examples of Virtual Facility Numbers
The use of VFS can be demonstrated with the following examples:
●
Example 1: VF Code = #191, VFN = 100 10288
— The first three digits (100) represent the FAC for a CO trunk group.
— 10288 represents an access code for a non-primary IXC.
— The destination telephone number (dialed by the user) will be transmitted
after this IXC access code, since “#5” was not used within the VF number.
●
Example
2:
VF Code = #193, VFN = 2219* 5554343 *#5#*12345
— The first three digits (221) represent the FAC for a tie trunk to a remote
PBX.
— The “9” is used to access the remote PBX’s ARS.
.
The “*” represents a pause of 1.5 seconds (allows time for dial tone to occur).
— The “555 4343” defines the local address of a private network and its internal
routing table.
— The “*” represents a 1.5 second pause.
2-259
The “#5” indicates that the destination number should be inserted here,
rather than at the end of the VF translation.
— The “#*” indicates that the system should transmit a “*” symbol as the first
character of an identification code.
— The “12345” represents the remaining characters in the identification code.
●
Example
3:
VF Code = #195,
VFN = 104 5554567 ****1234*9
This example demonstrates how a VF might be used in place of a tie trunk group
connecting two local PBXs, when you have permission to access the other PBX’s
facilities to complete calls.
— The first three digits (104) represent the FAC for a CO trunk group.
– The “5554567” represents the number for the other PBX.
— The “****” represents a 6 second pause (allows time for the other PBX to
answer and return new dial tone).
— The “1234” represents a “barrier” (security) code required to access the other
PBX’s facilities.
— The * represents a 1.5 second pause.
— The “9” represents an ARS access code for the other PBX.
— Since “#5” was not used within the VFN, the destination telephone number
(dialed digits) will be transmitted after this ARS access code.
Accessing a Virtual Facility
Dial access is provided by dialing the VF code (#190-#199, including the “#”), followed by the
destination telephone number.
●
Example:
— The user dials the following VF code and associated destination telephone
number.
#192 12125551643.
— The stored VFN associated with VF code #192 is defined as 2222*333*444.
— The first four digits (2222) represent the FAC for a tie trunk group to a
remote PBX.
— The * represents a 1.5 second pause.
— The next three digits (333) represent the security code required by the remote
PBX, indicating that you have permission to access their facilities.
— The * represents a 1.5 second pause, as the remote PBX checks the validity of
your security code.
— The final three digits (444) represent the FAC required by the remote PBX to
access their Band 5 WATS trunks.
2-260
— Since “#5” was not embedded within this VFN, the destination telephone
number (12125551643) will be transmitted after the WATS access code.
VF codes may be included in numbers stored in REP DIAL buttons, System Speed Dialing
codes, and Personal Speed Dialing codes if the VF code is used at the beginning of these
numbers. VF codes may not be assigned to FACILITY buttons and may not be embedded in
other virtual facility numbers.
VFs may be used in ARS routing patterns just as if they were physical facilities.
Considerations
VFs enhance the Automatic Route Selection feature by increasing the number of facility
types available for use in routing patterns. Using ARS ensures that the least-cost facility is
used to complete each call. User intervention is minimized and associated user dialing errors
are essentially eliminated.
In addition, in those systems where users are permitted dial access to VFs, user dialing of
long digit strings is minimized, as are the associated dialing errors.
Interactions
When using a VF through dial access, calling restrictions will be based on the station’s class
of service. Dial access VF calls will be completed only if:
●
The VF code is valid and not dial restricted.
●
The station is not outward restricted.
●
The station has dial access permission for the physical facility embedded within the
VFN.
●
The destination telephone number is valid and allowed for the station’s toll
restriction class.
VFs cannot be assigned to Facility buttons. Button access is provided by programming REP
DIAL buttons only.
Administration Requirements
A VF must be programmed via System 25 administration. The following items are
administrable:
●
Specify a virtual facility code (#190-# 199).
●
Assign a virtual facility number to this code.
●
Allow dial access to this virtual facility (l = Y/0 = N).
Hardware Requirements
None.
2-261
LIST OF FIGURES
Figure 3-1.
System
.
3-1
Figure 3-2.
Call Processor (ZTN82 or ZTN128) Circuitry . . . . . . . . . .
3-3
Figure 3-3.
Memory
.
3-5
Figure 3-4.
TDM Bus Time Slot Generation (Not A Timing Diagram) . . . . .
3-6
Figure 3-5.
TDM Bus Diagram - Three Cabinet System . . . . . . . . . .
3-9
Figure 3-6.
Equipment Connected to System 25 Via the Call Processor and Port
Circuit Packs (Sheet l of 3) . . . . . . . . . . . . . . . .
3-11
Figure 3-7.
Port
Figure 3-8.
Unique
Figure 3-9.
25
Digital
(ZTN81
Switch.
or
.
.
ZTN127)
.
.
.
Circuitry
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
3-15
Circuitry
.
.
.
.
.
.
.
.
.
3-17
Unique Loop Start Trunk (ZTN77) Circuitry . . . . . . . . . .
3-19
Figure 3-10. Unique Tip Ring Line (ZTN78) Circuitry . . . . . . . . . . .
3-20
Figure 3-11. Unique ATL Line (ZTN79) Circuitry . . . . . . . . . . . . .
3-23
Figure 3-12. Unique Data Line (TN726) Circuitry . . . . . . . . . . . . .
3-25
Figure 3-13. Unique MET Line (TN735) Circuitry . . . . . . . . . . . . .
3-27
Figure 3-14. Unique Analog Line (TN742) Circuitry . . . . . . . . . . . .
3-28
Figure
.
3-31
Figure 3-16. Unique Tie Trunk (TN760B) Circuitry . . . . . . . . . . . .
3-33
Figure 3-17. Tie Trunk (TN760B) Circuit Pack Option Switches . . . . . . . .
3-33
Figure 3-18. Unique Auxiliary Trunk (TN763) Circuitry . . . . . . . . . . .
3-36
Figure
.
3-40
Figure 3-20. Tone Detector (TN748) Circuit . . . . . . . . . . . . . . .
3-42
Figure 3-21. Pooled Modem (TN758) Circuit . . . . . . . . . . . . . . .
3-43
Figure
3-48
3-15.
3-19.
3-22.
Circuit
Unique
Service
System
Pack
Ground
DID
Common
Start
Trunk
Circuit
Software
Circuitry
Trunk
(ZTN76)
(TN753)
(ZTN85)
.
Partitioning
.
.
Circuitry
.
.
.
-ii-
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
LIST OF TABLES
TABLE 3-A. TDM BUS Time Slots .
TABLE 3-B. Signaling Type Summary
.
.
.
.
.
.
.
.
.
.
.
. .
.
.
.
TABLE 3-C. TN7670B Tie Trunk Preferred Signaling Formats
-iii -
.
.
.
.
.
.
.
.
3-8
.
.
.
.
.
.
.
.
3-34
.
.
.
.
.
.
.
.
3-34
SECTION 3—FUNCTIONAL DESCRIPTION
This section describes how the digital switch and the software of System 25 provide control
and switching.
Digital Switch
Figure 3-1 shows a sehematic diagram of the System 25 digital switch.
The basic switch hardware consists of the following:
●
Common Control
— Memory Bus
Call Processor Circuit Pack (CP)
— Memory CP.
●
Switching
Network
Time Division Multiplex (TDM) Bus
— Port Circuits
— System Resource Circuits: Service Circuit, Tone Detector, and Pooled Modem.
MEMORY BUS
COMMON
CONTROL
CALL
MEMORY
PROCESSOR
TDM BUS
SWITCHING
NETWORK
PORT
CIRCUITS
SERVICE
CIRCUIT
TONE
DETECTOR
SYSTEM
RESOURCES
TRUNKS,
VOICE TERMINALS
Figure 3-1. System 25 Digital Switch
3-1
POOLED
MODEM
Common Control
The Common Control circuitry consists of the Call Processor (ZTN82 in R1V1 or ZTN128 in
R1V2) and Memory (ZTN81 in R1V1 or ZTN127 in R1V2) Circuit Packs and associated
memory bus which is a 60-wire (including grounds), 39-bit (16-data, 23-address,), 6 MHz
frontplane flat ribbon cable.
Call Processor Circuit Pack (ZTN82 or ZTN128)
The Call Processor runs the system feature code. It is powered from the backplane by +5
and -5 volts. It also draws -48 volts from the backplane to drive the Emergency Transfer
Unit. Each system must include one Call Processor Circuit Pack. The Call Processor
circuitry, as shown in Figure 3-2 includes:
●
Microprocessor
●
Memory management
●
On-board
●
EIA
●
Network
●
Clock
●
Front plane interface
●
Reset
●
Bus error circuitry
●
Interrupt
●
Emergency Transfer Unit Control.
memory
channels
controller
circuitry
circuitry
Microprocessor: A 68010 16-bit microprocessor that executes call processing and data
processing features. This includes all maintenance, administration, testing, and reporting
software.
Memory Management:
Memory management separates the on-board Random Access
Memory (RAM) into 1024 memory pages of 256 bytes each. Each page is read and write
protected, generates bus errors when violated, and each is recappable allowing data areas to
remain contiguous.
On-Board Memory: On-board memory includes 64k bytes of Read Only Memory (ROM)
containing the power-up tests and the switch operating system. In addition, there is 80k
bytes of protected RAM containing writable data storage for call processing. The RAM is
backed up by an on-board trickle-charge battery that maintains memory contents for up to
two months. Of the 80k RAM, 24k is dedicated to translation data. The remainder is
dedicated to call status data and the operating system message queues.
EIA Channels:
communication
(SMDR) device,
port can support
Four asynchronous RS-232 EIA ports (l-4) are included to permit
with an administration terminal, a Station Message Detail Recording
and a digital tape unit. (The fourth port is reserved for future use.) Each
300, 1200,4800, or 9600 baud rates.
3-2
TO EMERGENCY
TRANSFER UNIT
(ETU)
EIA CHANNELS
(RS-232C)
INTERRUPT —
CIRCUITRY
\
/
SERIAL SERIAL SERIAL SERIAL
CHANNEL CHANNEL CHANNEL CHANNEL
1
2
3
4
-48V DC
ETU
CONTROL
—
BUS
ERROR
CIRCUITRY
MICROPROCESSOR
(68010)
EIA
—
CONTROL
EIA
CONTROL
LED
RESET
CIRCUITRY
PROCESSOR
BUS
●
●
●
●
●
●
MEMORY
MANAGEMENT
FRONTPLANE
INTERFACE
(BUS
BUFFERS)
—
/ TO MEMORY\
NETWORK
CONTROLLER
TDM
BUS
LEADS
/
SAKI
●
●
●
READ
ONLY
MEMORY
(64K)
TIME OF
DAY
CLOCK
CIRCUIT
PACK VIA
FRONT PLANE
BUS
RANDOM
ACCESS
MEMORY
(80K)
BUFFERS
\
+5V DC
BATTERY
(POWER
FAIL
(DETECT)
Figure 3-2. Call Processor (ZTN82 or ZTN128) Circuitry
3-3
Network Controller: The network controller transmits control channel messages between
the Call Processor and the port circuits over the TDM bus. The controller also monitors
system clocks.
The controller includes an 8-bit microprocessor that acts as a throttle passing messages
between the Call Processor and the port board microprocessors.
All uplink messages from the port circuits are checked for consistency and passed to the
Common Control. The controller is the distribution control point for all downlink control
messages. It continuously scans, over the TDM bus, the port circuit microprocessors for
sanity and activity. External RAM associated with this microprocessor- stores control
channel information and port related information.
The controller consists of bus buffers and a Sanity and Control Interface (SAKI). The
buffers provide the interface between the TDM bus and the on-board data buses to
SAKI. The SAKI receives and transmits control messages on the first five time slots on
TDM bus. The microprocessor communicates with the SAKI and external RAM over
address and data bus.
bus
the
the
the
Clock: A clock provides both time-of-day information (in seconds, minutes, and hours), and
the date to the 68010. The clock automatically adjusts for leap years. An on-board battery
backs up the clock, so that accurate time is maintained even when the systern power is off.
Front Plane Interface: Dedicated buffers provide an interface to the front plane, which is
the communication path to the Memory Circuit Pack.
Reset Circuitry: The processor is automatically reset when power is turned on, when the
+5 volt power supply drops below 4.5 volts (after it returns to +5 volts), or when the
network controller determines that the processor is not functioning correctly. The processor
can also reset the network controller when it determines that the network controller is not
functioning correctly.
Bus Error Circuitry: Bus errors suspend the processor from executing code. Bus errors
are generated when memory management detects illegal reads or writes to RAM, when the
processor attempts to access circuit packs or chips not physically present, or when the
network controller determines that the processor is not functioning correctly.
Interrupt Circuitry: Interrupts are prioritized into seven levels, of which the highest
(leve1 7) is nonmaskable. The interrupts are:
Interrupt
Level
AC Fail
Work cycle
Off board
EIA ports 3 and 4
EIA ports l and 2
Off board
Off board
7
6
5
4
3
2
1
Emergency Transfer Unit (ETU) Control: Removes -48V de power from the ETUs of
the system when the system loses power or a major system malfunction occurs.
3-4
Memory Circuit Pack (ZTN81 or ZTN127)
The Memory Circuit Pack provides for the storage of software associated with system
operation. This software includes call and administration processing, and other related
programs. The circuit pack is powered from the backplane by +5 volts. Each system must
include one Memory Circuit Pack. The Memory Circuit Pack circuitry (Figure 3-3) includes:
●
Address and data buffers
●
ROM array
●
ROM select
●
Timing and control logic
●
Built-in TDM bus termination resistors.
Address and Data Buffers: The address and data buffers interface the Memory Circuit
Pack to the address and data lines on the front plane.
ROM Array: The memory array consists of 16 ROM devices of 32K, 8 bit bytes each, for a
total capacity of 512K ROM. The ROMs are organized into pairs allowing the Call Processor
to access 16 bit words.
ROM Select:
information.
The memory selects the proper pair of ROMs according to address
Timing and Control Logic Circuit: C o n t r o l s t h e a c c e s s s p e e d o f t h e R O M ( n o w a i t
states) by returning a Data Transfer Acknowledge signal at the proper time.
Bus Terminators: These resistors are required for proper operation of the TDM bus. The
Memory Circuit Pack provides the proper termination for one end of the bus, and a plug-in
TDM bus termination circuit card (plugs into cabinet backplane) is used to terminate the
other end. For this reason, the Memory Circuit Pack must always be located in slot #l of
Cabinet 1.
\
ROM
SELECT
TERMINATOR
RESISTORS
/
FRONT PANEL
MEMORY BUS
(TO CALL
PROCESSOR
CIRCUIT
PACK )
ADDRESS
AND
DATA BUFFERS
\
TIMING
AND
CONTROL
ROM
ARRAY
Figure 3-3. Memory (ZTN81 or ZTN127) Circuitry
3 - 5
/
TO
TDM
BUS
Switching Network
System 25 uses distributed processing techniques to provide switched voice and data services.
The switch operates at 64 Kbps. The switching network consists of the following:
●
Time Division Multiplex (TDM) bus
●
Port
●
System Resource Circuits.
Circuits
The TDM bus connects the intelligent ports to the Common Control circuit packs and other
ports through the network control circuit. The system resource circuits provide tone sources,
The intelligent ports connect external
receivers, detectors, a n d p o o l e d m o d e m s .
communications facilities to the TDM bus.
TDM BUS
The TDM bus consists of two groups of eight signal leads and five control leads, each with
matching grounds. The port circuit packs place digitized voice [pulse code modulated (PCM)]
signals on the bus.
The bus operates at 2.048 MHz. The framing pulse rate is 8 kHz. This provides 256 time
slots (0-255) on the bus. The time slots are 488 ns wide. Time slots are generated as shown
in Figure 3-4. The first five time slots are used for communications between the Common
Control, the intelligent port, and resource circuit packs.
SYSTEM FRAME
8 KHZ
( 125 MICROSECONDS)
488 NANOSECONDS
— — — — — — — — —
TIME SLOTS O
1
2
4
5
255
256 TIME SLOTS
Figure 3-4. TDM Bus Time Slot Generation (Not A Timing Diagram)
3-6
0
Two time slots are required for each 2-party conversation. Each party transmits (talks) on
one time slot and receives (listens) on another. Only five parties are allowed in a conference.
During a conference connection, each member of the conference transmits on an individual
time slot while receiving on as many as four other time slots. The actual switch capacity is
115 simultaneous 2-party conversations).
Table 3-A shows the allocation of the 256 time slots. Five are used for system control, 15 for
tones, 235 for call processing, and one is not used.
Physical Characteristics
The TDM bus is an 8-bit bus. The bus snakes continuously between cabinets in a
multicabinet system as shown in Figure 3-5. The total length is about 9 feet for a three
cabinet system. The bus is driven from any of the circuit packs in the cabinets.
Similarly, a signal on the bus can be received by any circuit pack.
Within a cabinet, the bus is printed on one side of the circuit pack carrier backplane
while the other side is solid ground. Ribbon cables are used to cable the TDM bus
between cabinets in a multi-cabinet system.
Electrical Characteristics
The TDM bus is an unbalanced, low characteristic impedance transmission line. Paths
printed over a ground plane on the carriers and the flat ribbon cables between carriers
maintain this impedance level over the full length of the bus.
One end of the bus is terminated to ground with a bus termination circuit card and the
other end is terminated by a network on the ZTN81B or ZTN127 Memory CP. Each
circuit pack connects to the bus through a custom bus driver device. The bus driver is a
switchable constant current source so that even in the “high” output state there is no
bus loading to cause reflections. The current output of the drivers is adjusted so that
logic “high” is 1.5 volts compared to a “low” of 0 volts.
3-7
TABLE 3-A. TDM BUS Time Slots
Time Slot No.
Function
00
thru
04
Control
(5)
-TonesDial Tone
Busy Tone
Re-order Tone
Ringback Tone
Data-Null
Voice-Null
Music
697 Hz*
770 Hz*
852 Hz*
941 Hz*
1209 Hz*
1336 Hz*
1447 Hz*
1637 Hz*
(15)
05
06
07
08
09
10
11
12
13
14
15
16
17
18
19
20
thru
254
Call
Processing
(235)
255
Not Used
(1)
* These tones are used to generate touch-tone signals.
3-8
TDM
BUS
TERMINATOR
CARD
CABINET 3
ON/OFF
SWITCH
AC POWER
# 6 AWG
BUILDING
GROUND
WIRE
TDM BUS
EXTENDER
CABLE
CABINET 2
AC POWER
#6 AWG
GROUND
WIRE
CABINET 1
COUPLED
BONDING
CONDUCTOR
(CBC)
AC POWER
TO SINGLE
POINT GROUND
Figure 3-5. TDM Bus Diagram -Three Cabinet System
3-9
Port Circuits
The following port circuit packs provide the link between trunks and external equipment and
the TDM bus:
●
Analog Line (TN742)
●
ATL Line (ZTN79)
●
Auxiliary Trunk (TN763)
●
Data Line (TN726)
●
DID Trunk (TN753)
●
Ground Start Trunk (ZTN76)
●
Loop Start Trunk (ZTN77)
●
MET Line (TN735)
●
STARLAN
●
Tie Trunk (TN760B)
●
Tip Ring Line (ZTN78).
Interface
(ZTN84)
Figure 3-6 shows the equipment types that can be connected to the digital switch by the Call
Processor and port circuit packs.
3-10
SINGLE-LINE VOICE
TERMINALS
(420, 500, 2500, 2514,
2554, 7101A)
— — — — — — — —
RECORDED ANNOUNCEMENTS
— — — — — — — —
DICTATION
EQUIPMENT
— — — — — — — —
EXTERNAL ALERTING
DEVICES
— — — — — — — —
MUSIC-ON-HOLD
PART OF DIGITAL SWITCH
ZTN78
TIP RING
LINE CP
OR
TN742
ANALOG
LINE CP
MULTILINE VOICE
TERMINALS
(7300H-TYPE)
— — — — — — — —
DIRECT TRUNK ATTENDANT
CONSOLE (7305H02B) OR
SWITCHED LOOP ATTENDANT
CONSOLE (7305H04C)
—
—
—
—
—
—
—
ATTENDANT DIRECT
EXTENSION SELECTOR
CONSOLE
(MODEL 23A1)
ZTN79
ATL LINE CP
TN735
MET LINE CP
MET SETS
DICTATION
EQUIPMENT
— — — — — —
PAGING EQUIPMENT
(PagePac)
—
TIME
DIVISION
MULTIPLEX
BUS
TN763
AUXILIARY
TRUNK CP
—
ZTN84
STARLAN
INTERFACE
CP
NETWORK
EXTENSION
UNIT
Figure 3-6. Equipment Connected to System 25 Via the Call Processor and Port
Circuit Packs (Sheet l of 3)
3-11
PART OF DIGITAL SWITCH
SAT, DTU, SMDR,
CAS (ON-PREMISES
DIRECT CONNECTION)
Z3A1
ADU
*
Z3A4
ADU
*
*REQUIRED FOR CONNECTIONS >50 FEET
OR NOT SHARING SAME AC OUTLET
—
—
—
—
—
—
—
—
—
SAT, SMD,
CAS (ON-PREMISES
SWITCHED CONNECTION)
—
—
—
—
—
—
Z3A1/4
ADU
TN726
DATA LINE CP
Z3A4
ADU
—
—
—
— — — — — —
SAT, SMDR,
CAS (OFF-PREMISES
DIRECT CONNECTION)
—
MODEM
CO
(212-TYPE)
— — — — — — — —
SAT, SMDR,
CAS (OFF-PREMISES
SWITCHED CONNECTION) †
MODEM
(212-TYPE)
CO
—
—
—
—
TN726
DATA LINE CP
—
—
MODEM
(212-TYPE)
—
—
ZTN82 OR
ZTN128
PROCESSOR CP
(1) SAT
(2) SMDR, CAS
(3) DTU
(4) RESERVED
—
—
—
OPS
} OR
CO
† OFF-PREMISES STATION
OR CO TRUNK
Z3A4
ADU
—
ZTN82 OR
ZTN128 CALL
PROCESSOR CP
TIME
DIVISION
MULTIPLEX
BUS
ZTN82 OR
ZTN128 CALL
PROCESSOR CP
TN742
ANALOG LINE CP
ZTN76 GROUND
START TRUNK
CP OR ZTN77
LOOP START
TRUNK CP
TN726
DATE LINE CP
ZTN82 OR
ZTN128 CALL
PROCESSOR CP
Figure 3-6. Equipment Connected to System 25 Via the Call Processor and Port
Circuit Packs (Sheet 2 of 3)
3-12
PART OF DIGITAL SWITCH
DATA TERMINAL
EQUIPMENT,
HOST COMPUTER
RS-232C
Z3A1/2/4
ADU
SINGLE-LINE
VOICE TERMINAL
(2500-TYPE OR
7101A)
DATA TERMINAL
EQUIPMENT,
HOST COMPUTER
TN726
DATA LINE CP
TN742
ANALOG LINE
CP OR
ZTN78 TIP
RING LINE CP
RS-232C
TN726
DATA LINE CP
Z3A5
ADU
MULTILINE
VOICE TERMINAL
(7300H-TYPE)
—
ZTN79
ATL LINE CP
ZTN76
GROUND START
TRUNK CP
OR
ZTN77
LOOP START
TRUNK CP
CO, FX, WATS
—
—
—
—
PAGING
EQUIPMENT
DID TRUNKS
TN753
DID
TRUNK CP
TIE TRUNKS
TN760B
TIE
TRUNK CP
TIME
DIVISION
MULTIPLEX
BUS
LEGEND :
A D U – ASYNCHRONOUS DATA UNIT
CAS – CALL ACCOUNTING SYSTEM
CO – CENTRAL OFFICE
CP – CIRCUIT PACK
DID - DIRECT INWARD DIALING
D T U– DIGITAL TAPE UNIT
FX – FOREIGN EXCHANGE
MET – MULTIBUTTON ELECTRONIC TELEPHONE
O P S- OFF-PREMISES STATION
SAT – SYSTEM ADMINISTRATION TERMINAL
S M D R– STATION MESSAGE DETAIL RECORDING
WATS - WIDE AREA TELECOMMUNICATIONS SERVICE
Figure 3-6. Equipment Connected to System 25 Via the Call Processor And Port
Circuit Packs (Sheet 3 of 3)
3-13
Eight port circuits are provided on most port circuit packs. The Multibutton Electronic
Telephone (MET) Line, Tie Trunk, and Auxiliary Trunk Circuit Packs each contain four port
circuits. The port circuits provide an interface between terminals/trunks and the TDM bus.
The number of port circuit packs required varies according to customer requirements and
equipment configuration.
Each of the System 25 port circuit packs contain a number of common elements (see Figure
3-7) as well as the unique port circuits. The common elements are as follows:
●
Bus buffers
●
Sanity And Control Interface (SAKI)
●
On-board microprocessor with external Random Access Memory (RAM)
●
One or more Network Processing Elements (NPEs)
●
Circuit Pack Address Leads.
Bus Buffers: The bus buffers are the digital interface between the backplane TDM bus
wires (system bus) and the on-board circuitry (data bus). They also receive and distribute
clock and frame signals.
SAKI (Sanity And Control Interface): The SAKI is the control interface between the
Common Control that sends information via the network control circuit down the TDM buses
and the on-board circuitry controlled by the on-board microprocessor. The SAKI receives
control information (down-link messages) on the first five time slots and, as requested by
the on-board microprocessor, transmits control information (up-link messages) on these
same time slots.
The SAKI also does the following functions:
●
Identifies the circuit pack to the Common Control (location and vintage)
●
Controls status indicator Light-Emitting Diodes (LEDs) - red (failure), green
(translated), and yellow (circuit busy)
●
Initiates power-on startup procedures
●
Checks the on-board microprocessor for sanity and causes reinitialization in case of
problems
●
Takes NPEs out of service under control of the on-board microprocessor
●
Resets the protocol handler on the ATL Line Circuit Pack
●
Takes the whole circuit pack out of service on command from the Common Control
or when it determines that on-board interference is present in the control time
slots.
On-Board Microprocessor With External RAM: The on-board processor does all low
level functions such as scanning for changes and relay operations. In general, it carries out
commands received from the Common Control and reports status changes to it. The external
RAM stores control channel information and port-related information.
NPEs (Network Processing Element): Each port circuit pack contains one or two NPEs.
The Analog Line, ATL Line, Tip Ring, Data Line, Ground Start, Loop Start, and DID Trunk
circuit packs contain two NPEs. The MET Line, Auxiliary Trunk, and Tie Trunk Circuit
Packs contain one NPE.
3-14
The NPEs do switching network functions for the port circuits. Under control of the onboard microprocessor, an NPE can connect a port circuit to any one of the TDM bus time
slots. More specifically, it allows a port circuit to talk on one time slot and listen to the
same time slot (NPE sidetone) and on up to four other time slots at the same time. In 2wire circuits that provide their own sidetone, the NPE sidetone is not used.
Circuit Pack Address Leads: Seven leads (BA0-BA6) are tied to Corresponding logic
levels to uniquely identify each CP slot in the system, including multiple cabinet systems.
The logic values on leads BA4 and BA5 are used to identify the cabinet (Cabinet 1, 2, or 3)
and are tied via the cabinet address plugs to either +5V de or ground, as appropriate. Lead
BA6 is tied to ground.
TDM
BUS
LEADS
❘
❘
❘
●
NPE(S)
BUS
BUFFERS
●
CIRCUIT
PACK
ADDRESS
LEADS
❘
❘
❘
●
SAKI
RAM
ON-BOARD
MICROPROCESSOR
RED
LEDS
GREEN
YELLOW
Figure 3-7. Port Circuit Pack Common Circuitry
3-15
PORT
SPECIFIC
CIRCUITRY
Ground Start Trunk (ZTN76)
The Ground Start Trunk Circuit Pack interfaces eight central office trunks and the TDM
bus. Figure 3-8 shows the following Ground Start Trunk unique circuitry:
●
Ground detector circuit
●
Port Input/Output (I/O) circuit
●
Eight port circuits.
Ground Detector Circuit: The ground detector circuit determines if ground has been
applied to the tip lead for incoming seizure. It also senses tip ground on outgoing seizure
indicating dial tone is present. One ground sensor is used for each port circuit. Input for
the ground sensor comes from the port circuit as an analog current to the -48 volt dc supply.
The output of the ground sensor is a port control point to the port 1/0 circuit.
Port I/O Circuit: This circuit consists of bus expanders for communication between the
on-board microprocessor and the port circuits. It receives commands from the on-board
microprocessor and distributes them to the individual port circuits. It also accesses the port
circuit scan points and passes the information to the on-board microprocessor.
Port Circuits: The eight port circuits are identical. Each port circuit consists of a
coder/decoder (codec), hybrid circuit, line transformer, relay driver, and surge protection
circuit.
The codec is a 4-wire circuit that converts the NPEs digital output to an analog signal.
Likewise, it converts the analog signal from a central office trunk to a Pulse Code Modulated
(PCM) data signal to the NPE. The hybrid circuit converts the codec 4-wire analog signal to
a 2-wire analog signal that is connected to the central office trunk by the line transformer.
The relay driver buffers and inverts the relay drive signals from the port 1/0 circuit so that
a logic high input operates the appropriate relay. The relays control circuitry provides the
proper signaling for ground start trunks. The trunks support touch-tone dialing. The surge
protection circuit provides overvoltage lightning surge protection for the circuit pack.
3-16
PORT
CIRCUIT
0
T.O
CODEC
NPE O
HYBRID
●
NPE 1
❘
❘
❘
●
ON-BOARD
MICROPROCESSOR
PORT
1/0
CIRCUIT
●
●
●
PORT
CIRCUIT
4
●
GROUND
DETECTOR
PORT
CIRCUIT
3
●
❘
❘
❘
PORT
CIRCUIT
7
Figure 3-8. Unique Ground Start Trunk (ZTN76) Circuitry
3-17
\
R.O
T.3
R.3
TO
CENTRAL
OFFICE
T.4
R.4
T.7
R.7
/
Loop Start Trunk (ZTN77)
The Loop Start Trunk Circuit Pack interfaces eight central office loop start trunks and the
TDM bus.
Figure 3-9 shows the following Loop Start Trunk unique circuitry:
●
Port Input/Output (I/O) circuit
●
Eight port circuits.
Port I/O Circuit: This, circuit consists of bus expanders for communication between the
on-board microprocessor and the port circuits. It receives commands from the on-board
microprocessor and distributes them to the individual port circuits. It also accesses the port
circuit scan points and passes the information to the on-board microprocessor.
Port Circuits: The eight port circuits are identical. Each port circuit consists of a codec,
hybrid circuit, line transformer, relay driver, and surge protection circuit.
The codec is a 4-wire circuit that converts the NPEs output to an analog signal. Likewise, it
converts the analog signal from a central office trunk to a PCM data signal to the NPE. The
hybrid circuit converts the codec 4-wire analog signal to a 2-wire analog signal that is
connected to the central office trunk by the line transformer.
The relay driver buffers and inverts the relay drive signals from the port 1/0 circuit so that
a logic high input operates the appropriate relay. The relays control circuitry provides the
proper signaling for loop start trunks. The trunks support touch-tone dialing and dial pulse
signaling. The surge protection circuit provides overvoltage lightning surge protection for
the circuit pack.
3-18
PORT
CIRCUIT
o
T.O
CODEC
HYBRID
\
R.O
●
NPE O
❘
NPE 1
❘
❘
ON-BOARD
MICROPROCESSOR
PORT
I/O
CIRCUIT
●
●
PORT
CIRCUIT
3
T.3
R.3
TO
CENTRAL
OFFICE
T.4
●
●
●
●
PORT
CIRCUIT
4
❘
❘
❘
PORT
CIRCUIT
7
R.4
T.7
R.7
/
Figure 3-9. Unique Loop Start Trunk (ZTN77) Circuitry
3-19
Tip Ring Line (ZTN78)
The Tip Ring Line Circuit Pack interfaces eight analog tip and ring voice terminal lines
(single-line voice terminals) and the TDM bus. Figure 3-10 shows the following Tip and Ring
Line unique circuitry:
●
Ringing application circuit
●
Port Input/Output (I/O) circuit
●
Eight port circuits.
PORT
CIRCUIT
0
CODEC
❘
❘
❘
❘
NPE 1
PORT
I/O
CIRCUIT
●
●
●
R.O
T.3
R.3
TO
ANALOG
TIP/RING
TERMINALS
T.4
●
PORT
CIRCUIT
4
●
●
●
❘
❘
❘
●
–48V TO –24V
POWER
CONDITIONER
T.O \
PORT
CIRCUIT
3
●
RINGING
APPLICATION
CIRCUIT
POWER
SUPPLY
ELECTRONIC
BATTERY
FEED
●
NPE O
ON-BOARD
MICROPROCESSOR
HYBRID
PORT
CIRCUIT
7
●
R.4
T.7
R.7
Figure 3-10. Unique Tip Ring Line (ZTN78) Circuitry
3-20
/
Ringing Application Circuit: This circuit receives ringing voltage from the power supply.
It monitors ringing voltage and current and generates signals to the on-board
microprocessor indicating zero ringing voltage and current. It also detects when a terminal
user has lifted the receiver during ringing preventing the application of ringing to the
terminal’s handset receiver.
This circuit includes bus expanders connecting the on-board
Port I/0 Circuit:
It receives c o m m a n d s f r o m the on-board
microprocessor a n d t h e p o r t c i r c u i t s .
microprocessor and distributes them to the individual port circuits. It also accesses the port
circuit scan points and passes the information to the on-board microprocessor.
-48 V To -24 V Power Conditioner: This circuit converts -48 V power from the power
supply into a conditioned source of -24 V power for the electronic battery feed circuits.
Port Circuits: Each port circuit is identical. A port circuit consists of a coder/decoder
(codec), h:ybrid circuit, battery feed circuit, and ring relay.
The codec is a 4-wire circuit that converts the NPEs output to an analog signal. Likewise, it
converts the analog signal from a central office trunk to a PCM data signal to the NPE. The
hybrid circuit converts the codec 4-wire analog signal to a 2-wire analog signal that is
connected to the central office trunk by the line transformer.
The battery feed circuit provides talking battery to the voice terminal. It also detects when
a receiver is lifted, and provides the message waiting signal by periodically reducing the feed
voltage to zero.
The ring relay provides the interface between the ringing application circuit and the port
circuit. It causes ringing to turn on and off.
Note: The TN742 can be used instead of the ZTN78 Tip Ring CP. The TN742 supports
up to five bridged single-line voice terminals; however, only two can be off-hook at one
time. The ZTN78 does not support bridged terminals. In addition, the TN742 supports
out-of-building, extended, and off-premises stations, while the ZTN78 does not. The
ZTN78 supports only a 1.2 Ringer Equivalency Number (REN).
3-21
ATL Line (ZTN79)
The ATL I,ine Circuit Pack interfaces eight hybrid voice terminal (7300H series) lines and
the TDM bus. It terminates three pairs of wires from each terminal: analog voice pair,
digital control pair, and power pair. Figure 3-11 shows the following ATL Line unique
circuitry:
●
Protocol
handler
●
Eight port circuit.
Protocol Handler: The 8-bit on-board microprocessor translates the control information
in Control Channel Message Set (CCMS) message format to the control information message
format used by the 7300H series voice terminals. The protocol handler sends the messages
to the terminals via transceivers located in the port circuits.
Port Circuits: Each port circuit is identical. A port circuit consists of an analog port,
one-half of a transceiver, and an electronic power feed device.
The analog port circuit consists of a codec, a hybrid circuit, an isolation transformer, and
associated power filtering circuitry. The codec and hybrid circuit perform the same function
as the codec and hybrid circuit in the Analog Line Circuit Pack (TN742). The output of the
hybrid circuit is connected to the primary of the isolation transformer. The secondary of the
transformer is connected to the analog voice pair.
The transceiver interfaces the voice terminal pair to the protocol handler. The electronic
power feed device provides -24 volts de on the power pair to the voice terminal. The device is
polled by the on-board microprocessor, periodically and on demand, to test for an
overcurrent or no-current condition.
Each Electronic Power Feed (EPF) circuit supports two ports. If one of the associated lines
becomes overloaded, the associated pair of lines will also be out of service. One EPF
supports Ports O and 1, one Ports 2 and 3, one Ports 4 and 5, and one Ports 6 and 7. The
on/off state of the device is controlled by the on-board microprocessor.
3-22
PORT
CIRCUIT
o
ANALOG
PORT
T.O
CODEC
HYBRID
ISOLATION
TRANSFORMER
R.O
\
/
NPE 0
●
\
DATA
TRANSCEIVER
PROTOCOL
HANDLER
ON-BOARD
MICROPROCESSOR
ON-BOARD
MICROPROCESSOR
●
ELECTRONIC
POWER FEED
/
●
\
PORT
CIRCUIT
3
●
PORT
CIRCUIT
4
●
NPE 1
/
●
PXT.0
PXR.0
PXR.1
/
●
TXR. O
PXT.1
❘
❘
❘
\
TXT. O
●
T.3
R.3
TXT.3
TXR .3
PXT .3
PXR.3
TO
MULTILINE
VOICE
TERMINALS
T.4
R.4
TXT .4
TXR.4
PXT.4
PXR.4
❘
\
❘
❘
PORT
CIRCUIT
7
T.7
R.7
TXT.7
TXR .7
PXT.7
PXR.7
/
Figure 3-11. Unique ATL Line (ZTN79) Circuitry
3-23
Data Line (TN726)
The Data Line Circuit Pack interfaces eight Asynchronous Data Units (ADUs) data devices
and the TDM bus. The ADUs are typically, in turn, connected to RS-232 type devices.
Figure 3-12 shows the Data Line unique circuitry that includes:
●
A bit clock
●
Bus isolation
●
Eight port circuit.
Bit Clock: The bit clock circuitry is used to provide the Octal Asynchronous Terminal
Mode Two EIA Asynchronous LSIs (OATMEALs) with a clock frequency that is a multiple
of each baud rate. In addition, the clock rate is divided down to 160 kHz. The 160 kHz is
then compared to the 160 kHz data clock of the system, and is phase-locked to the system
clock. The phase-locked circuit is required for low speed operation.
Bus Isolation: This portion of the circuit pack is used to isolate the microprocessor bus.
Isolation is required because the realized bus load exceeds the maximum limit specified for
this device, due to the large number of devices controlled by the NPE. The OATMEALs are
isolated from the common bus structure.
Port Circuits. Each of the eight identical port circuits allows the connection of interface
equipment having an RS-232 compatible serial interface to the switch. The circuit provides
asynchronous full duplex data transport at standard speeds from 300 to 19,200 bps and a low
data rate (<300 bps). Each port includes an Asynchronous Data Unit (ADU) to extend the
serial communications link length and provide safe isolation. The ADU terminates to
another ADU at the Customer Provided Equipment (CPE). The distance between the digital
switch and CPE is inversely proportional to the speed at which the link is run.
Throughout the circuit, various gates are used to provide a means of isolating devices for
automated circuit pack testing. Typically, these devices are crystal oscillators or memory
components attached to the microprocessor bus.
3-24
A
/
PORT
CIRCUIT
0
ASYNCHRONOUS
DATA
UNIT
(ADU)
PROTOCOL
HANDLER
(OATMEAL)
●
PXT.O
PXR.0
TXT.0
TXR.0
\
NPE
0
●
\
●
A
PORT
CIRCUIT
1
A
PORT
CIRCUIT
2
A
PORT
CIRCUIT
3
A
PORT
CIRCUIT
4
A
PORT
CIRCUIT
5
●
ON-BOARD
MICROPROCESSOR
●
●
●
/
●
NPE
1
BUS
ISOLATION
●
\
●
TO
ADUs
●
●
●
●
●
●
●
BIT
CLOCK
●
A
●
PORT
CIRCUIT
6
●
A
PORT
CIRCUIT
7
PXT.7
PXR.7
TXT.7
TXR.7
/
Figure 3-12. Unique Data Line (TN726) Circuitry
3-25
MET Line (TN735)
The MET Line Circuit Pack interfaces four Multibutton Electronic Telephone (MET) lines
and the TDM bus. The MET Line unique circuitry consists of four port circuits as shown in
Figure 3-13.
Port Circuits: The four port circuits are identical. Each port circuit consists of an analog
port, a digital port, and an electronic power feed device.
The analog port circuit consists of a codec, a hybrid circuit, an electronic battery feed, and a
power filter. The codec, hybrid circuit, and power filter perform the same function as in the
Analog Line Circuit Pack (TN742). The electronic battery feed provides talking battery to
the MET set. The electronic battery feed produces a controlled de battery feed current for
short and long loops and detects when a MET set user lifts a receiver.
The digital port circuit provides a full duplex channel over two 2-wire pairs. All outgoing
lamp (LT, I,R) and incoming button depression (BT, BR) information is carried on these
channels. Ringing and switchhook information is also sent over these channels.
The electronic power feed device provides phantomed -48 volt dc power for the MET
terminals over the data channels. The electronic power feed device is a “smart” circuit
breaker. When it senses an overcurrent condition, it indicates the condition on an output
lead and goes into thermal shutdown if not turned off by the on-board microprocessor.
When the overcurrent condition disappears, the circuit breaker can be turned on by the onboard microprocessor.
3-26
PORT CIRCUIT
0
\
NPE
ANALOG
PORT
●
T.O
R.O
BT.O
DIGITAL
PORT
LT.O
LR.O
TO MET
TERMINALS
ELECTRONIC
POWER FEED
●
ON-BOARD
MICROPROCESSOR
BR.O
❘
❘
❘
●
PORT CIRCUIT
3
●
T.3
R.3
BT.3
BR.3
LT.3
LR.3
/
Figure 3-13. Unique MET Line (TN735) Circuitry
3-27
Analog Line (TN742)
The Analog Line Circuit Pack interfaces eight analog voice terminal lines and the TDM bus.
Figure 3-14 shows the following Analog Line unique circuitry:
●
Ringing
application
circuit
●
Port Input/Output (I/O) circuit
●
Eight port circuits.
PORT
CIRCUIT
0
CODEC
HYBRID
ELECTRONIC
BATTERY
FEED
OVERVOLTAGE
PROTECTION
●
NPE 0
❘
NPE 1
❘
/
ON-BOARD
MICROPROCESSOR
PORT
I/O
CIRCUIT
❘
❘
●
PORT
CIRCUIT
3
●
●
\
RINGING
APPLICATION
CIRCUIT
●
●
PORT
CIRCUIT
4
●
●
❘
❘
❘
POWER
SUPPLY
PORT
CIRCUIT
7
T.O
\
R.0
T.3
R.3
TO
ANALOG
TIP/RING
VOICE
TERMINALS
T.4
R.4
T.7
R.7
/
Figure 3-14. Unique Analog Line (TN742) Circuitry
3-28
Ringing Application Circuit: This circuit receives ringing voltage from the power supply.
It monitors ringing voltage and current, generates signals to the on-board microprocessor
indicating zero ringing voltage and current, and detects a terminal user lifting the receiver
during ringing. This prevents the application of ringing to the port circuit when a terminal
user lifts the receiver during the ringing phase. Maintenance circuitry is also included. The
maintenance circuitry detects when a terminal is connected to the port circuitry and checks
for faults in the ringing application circuitry.
Port I/O Circuit: This circuit consists of bus expanders connecting the on-board
It receives c o m m a n d s f r o m the on-board
microprocessor a n d t h e p o r t c i r c u i t s .
microprocessor and distributes them to the individual port circuits. It also accesses the port
circuit scan points and passes the information to the on-board microprocessor.
Port Circuits: The eight port circuits are identical. Each port circuit consists of a
coder/decoder (codec), hybrid circuit, electronic battery feed circuit, ring relay, and
overvoltage surge protection circuit.
The codec is a 4-wire circuit that converts the analog signal from a voice terminal to a PCM
data signal. It converts an incoming PCM data signal from the NPEs to an analog signal.
The hybrid circuit converts the 4-wire analog signal from the codec to a 2-wire analog signal
that is connected to the analog line. Filtered power is provided for the codec and hybrid
circuits.
The electronic battery feed circuit provides talking battery to the voice terminal. It also
produces a controlled de battery feed for short and long loops, detects tvhen a receiver is
lifted, and provides the message waiting signal by periodically turning off the feed voltage.
The ring relay provides the interface between the ringing application circuit and the port
circuit. It causes ringing turn on and turn off.
The overvoltage surge protection circuit provides lightning surge and power line cross
protection for the circuit pack.
Note: The TN742 can be used instead of the ZTN78 Tip Ring CP. The TN742 supports
up to five bridged single-line voice terminals, however, only two can be off hook at one
time. The ZTN78 CP does not support bridged terminals. In addition, the TN742
supports out-of-building, extended, and off-premises stations, the ZTN78 does not.
3-29
DID Trunk (TN753)
The DID Trunk Circuit Pack interfaces eight central office trunks arranged for Direct
Inwrd Dialing ((DID) and the TDM bus. Figure 3-15 shows the following DID Trunk unique
circuitry:
●
Port Input/output (1/0) circuit
●
Eight port circuits.
Port I/O Circuit: This circuit consists of bus expanders for communication between the
on-board microprocessor and the port circuits. It receives commands from the on-board
microprocessor and distributes them to the individual port circuits. It also accesses the port
circuit scan points and passes the information to the on-board microprocessor.
Port Circuits: The eight port circuits are identical. Each port circuit consists of a codec,
balance network, trunk interface unit, and loop termination circuit.
The codec is a 4-wire circuit that converts the NPEs output to an analog signal. Likewise, it
converts the analog signal from the Central Office (CO) to a PCM signal to the NPE.
The trunk interface unit contains a hybrid, a 2-wire interface circuit, and control circuitry.
The hybrid circuit converts the 4-wire analog signal from the codec to a 2-wire analog signal
that is connected to the analog line by the 2-wire interface circuit. The control circuitry
controls loop current, internal signal gain, terminating resistance, battery feed shutdown,
and battery reversal. The circuit pack accepts both dial pulse and touch-tone signaling.
The loop termination circuit provides a fixed impedance to the DID trunk.
3-30
PORT
CIRCUIT
0
TRUNK
INTERFACE
UNIT
T.O
CODEC
NPE O
\
HYBRID
R.O
●
NPE 1
❘
❘
❘
ON-BOARD
MICROPROCESSOR
PORT
I/O
CIRCUIT
●
●
●
PORT
CIRCUIT
3
T.3
R.3
TO
CENTRAL
OFFICE
●
●
●
●
PORT
CIRCUIT
4
❘
❘
❘
PORT
CIRCUIT
7
T.4
R.4
T.7
R.7
/
Figure 3-15. Unique DID Trunk (TN753) Circuitry
3-31
Tie Trunk (TN760B)
The Tie Trunk Circuit Pack interfaces four 6-wire tie trunks and the TDM bus. Two tip and
ring pairs form a 4-wire analog transmission line. An E and M pair are used for signaling.
The T and R pair transmit analog signals from the circuit pack. The T1 and RI pair receive
analog signals from the tie trunk. The E and M pair are dc signaling leads used for call
setup handshaking. The E lead receives signals from the tie trunk and the M lead provides
signals from the circuit pack. The TN760Bs four port circuits support Type I, Type I
Cornpatible, or Type V signaling. Incoming and outgoing trunks can be either automatic,
immediate start, wink start, or delay dial. Figure 3-16 shows the following Tie Trunk unique
circuitry:
●
Ground
detector
circuit
●
Port Input/Output (1/0) circuit
●
Four port circuits.
Ground Detector Circuit: This circuit determines if a ground has been applied to the E
lead. Ground detector inputs come from the port circuits as an analog current to the -48 volt
dc supply. Its output is a port control point to the port I/O circuit.
Port I/O Circuit: This circuit consists of bus expanders for communication between the
on-board microprocessor and the port circuits. It receives commands from the on-board
microprocessor and distributes them to the individual port circuits. It also accesses the port
circuit scan points and passes the information to the on-board microprocessor.
Port Circuits: The port circuits are identical, except for port 3 where part of the E-lead
maintenance circuit is located. Each port circuit consists of a codec with associated input
and output line transformers, analog operational amplifiers, a power filter, loop-around
transistors, port control comparators, a relay driver, an electronic power feed device, an Elead test maintenance circuit, and surge protection circuits.
The codec converts the incoming 4-wire analog signal from the tie trunk to a PCM data
signal. The codec converts the incoming PCM data signal from the NPE to an analog signal.
Outgoing and incoming line transformers provide de isolation to the tip and ring leads.
Analog operational amplifiers provide amplification and buffering for the codec and network
and loop-around gain compensation. Filtered power is provided to the codec and amplifiers.
The loop-around transistors are under control of the port control comparators and provide a
loop-around path for the signal for testing purposes. The relay driver buffers and inverts
the relay drive signals from the port 1/0 circuit so that a logic high input operates the
appropriate relay. The relays and electronic power feed device control the M-lead circuitry
to provide the proper signaling handshake for call progress tones and dial pulse dialing.
The electronic feed device provides a -48 volt de current to the M-lead circuits. It also tests
the M-1ead circuits for opens or shorts and prevents uncontrolled operation during power-up.
The E-lead test circuit provides a ground to the ground detector circuit for testing purposes.
The surge protection circuitry provides lightning surge and power cross protection for the
circuit pack. For each port circuit, E&M/Simplex and surge protection are selected by
switch settings as shown on Figure 3-17.
The signaling type is administrable for each port. Table 3-B summarizes the conditions
present as the transmit and receive control signals for each signaling type. Table 3-C lists
the preferred TN760B tie trunk signaling format to be used in the likely-to-be-encountered
installation situations.
3-32
\
NPE
●
ON-BOARD
MICROPROCESSOR
PORT
I/O
CIRCUIT
T.O
R.O
T1.0
R1.O
E.O
M.0
PORT
CIRCUIT
0
●
❘
❘
GROUND
DETECTOR
❘
❘
●
TO
TIE
TRUNKS
T.3
R.3
T1.3
R1.3
E.3
M.3
PORT
CIRCUIT
3
/
Figure 3-16. Unique Tie Trunk (TN760B) Circuitry
UNPROT .
0 0 0 0
0 0 0 0
PORT: 4 3 2 1
PROT .
/
SMPLX
SMPLX
E&M
SMPLX
PORT 2
PORT 3
PORT 4
E&M
SMPLX
E&M
PORT 1
E&M
Figure 3-17. Tie Trunk (TN760B) Circuit Pack Option Switches
3-33
TABLE 3-B. Signaling Type Summary
SIGNALING
TYPE
I Std.
I Compat.
V
RECEIVE
OFF-HOOK
ON-HOOK
grd
open/bat (*)
grd
open/bat (*)
grd
open
TRANSMIT
ON-HOOK
OFF-HOOK
bat
grd
grd
open/bat (*)
grd
open
* An open circuit is preferred over voltage.
TABLE 3-C. TN760B Tie Trunk Preferred Signaling Formats
INSTALLATION
PREFERRED
SITUATION
SYSTEM
FROM
S25
S25
S25
S25
S25
S25
S25
S25
S25
CIRCUMSTANCES
CoLocated
InterBuilding
CoLocated
InterBuilding
CoBuilding
InterBuilding
CoLocated
InterBuilding
CoLocated
TO
SIMPLEX
SIGNALING
FORMAT
FAR END
25
PROTECTED
SIMPLEX
PROTECTED
OR
SIGNAL
OR
OR
SIGNAL
OR
E&M
TYPE
UNPROTECTED
E&M
TYPE
UNPROTECTED
S25/S75
Simplex Type V
(Either)
Simplex
Type V
(Either)
S25/S75
Simplex Type V
(Either)
Simplex
Type V
(Either)
S85
Simplex Type V
(Either)
Simplex
Type V
(Either)
S85
Simplex Type V
(Either)
Simplex
Type V
(Either)
Unprotected
E&M
Protected
E&M
Unprotected
E&M
Protected
E&M
Unprotected
(Don't
Care)
Dim.
E&M
Dim.
E&M
Other
E&M
Other
E&M
Network
Interface
E&M
Type I
Compatible
Type I
Compatible
Type I
Compatible
Type I
Compatible
Type I
Standard
* Requires a protection unit.
3-34
Type I
Standard
Type I
Standard
Type I
Standard
Type I
Standard
(Don't
Care)
Unprotected
Protected
Unprotected
*
(Don’t
Care)
Auxiliary Trunk (TN763)
The Auxiliary Trunk Circuit Pack interfaces four ports provided for client-provided
equipment (CPE) and the TDM bus. It is connected to the CPE by up to three pairs of wires.
The transmission pair (T and R) carry voice signals and touch-tone control signals. The T
and R also provide a loop start seizure indication to the CPE. The seizure pair (SZ and SZ1)
provide seizure indication to the CPE. The signal pair (S and S1) provide answer supervision
and/or make-busy information from the CPE. Depending on the application, either the
transmission pair only or all three pairs are connected to the CPE.
Figure 3-18 shows the following Auxiliary Trunk unique circuitry:
●
Ground detector circuit
●
Port Input/Output (I/O) circuit
●
Four port circuits.
Ground Detector Circuit: This circuit determines if an answer-supervision or make-busy
signal from the CPE is present. The inputs of the ground detector come from the port
circuits as an analog current to the -48 volt de supply. Its output is a port control point to
the port I/O circuit.
Port I/O Circuit: This circuit consists of bus expanders for communication between the
on-board microprocessor and the port circuits. It receives commands from the on-board
microprocessor and distributes them to the individual port circuits. It also accesses the port
circuit scan points and passes the information to the on-board microprocessor.
Port Circuits: The four port circuits are identical. Each port circuit consists of a codec,
hybrid circuit, line transformer, relay driver, battery polarity sensor, and surge protection
circuit.
The codec is a 4-wire circuit that converts the analog signal from the CPE to a PCM data
signal. It converts an incoming PCM data signal from the NPE to an analog signal. The
hybrid circuit converts the 4-wire analog signal from the codec to a 2-wire analog signal that
is connected to the CPE by a line transformer.
The relay driver buffers and inverts the relay drive signals from the port I/O circuit so that
a logic high input operates the appropriate relay. The relays control circuitry that provide
the proper interfaces for CPE.
The surge protection circuit provides lightning surge protection for the circuit pack.
The circuit pack supports both touch-tone and dial pulse signaling. Longitudinal surges are
isolated from the hybrid and codec by the line transformer.
3-35
\
ON-BOARD
MICROPROCESSOR
PORT
CIRCUIT
0
●
NPE
PORT
I/O
CIRCUIT
●
GROUND
DETECTOR
●
I
I
I
I
I
❘
PORT
CIRCUIT
3
T.O
R.O
S.0
S1.0
SZ.0
SZ1.0
TO
AUXILIARY
EQUIPMENT
T.3
R.3
S.3
S1.3
SZ.3
SZ1.3
/
Figure 3-18. Unique Auxiliary Trunk (TN763) Circuitry
3-36
STARLAN Interface (ZTN84) (V2)
The STARLAN Interface (ZTN84) is a System 25 circuit pack (CP) designed to function as
either a gateway or a bridge between the PBX and the AT&T STARLAN NETWORK. The
ZTN84 contains much of the circuitry common to the other CPS in the system, that is a
Sanity and Control Interface (SAKI), a Network Processing Element (NPE), and a 8031
microprocessor. The CP also contains the circuitry required to perform the protocol
conversion on the data as it travels from one system to the other. These devices include a
80186 microprocessor, 82586 coprocessor, four Octal Asynchronous Terminal Mode 2 to EIA
Asynchronous LSI (OATMEAL) devices, and a logic sequencer. The 80186, the 82586, and the
logic sequencer (PLS105N) work together to add and delete the protocol used by the Local
Area Network (LAN), while the 80186 and the OATMEALS work together to add and delete
the protocol used by the PBX.
The ZTN84 can support up to four circuit switch connections between the Private Branch
Exchange (PBX) and the Local Area Network (LAN); this capability is provided by the four
OATMEALS and the NPE, the latter being a four channel device. In providing a connection
between the PBX and the LAN, capabilities such as file sharing, printer services, connections
to hosts, and modem pooling may be accessible across systems.
The OATMEAL devices on the ZTN84 are used in such a way as to support asynchronous
data communication at any of the standard rates ranging from 300 bps to 19.2 Kbps. The
asynchronous protocol that is used is a subset of Digital Communications Protocol (DCP)
Mode 2, as only “I” channel information is transmitted, where the data is formatted in
High-Level Data Link Control (HDLC) frames.
The ZTN84 has been designed with a hardware interface that allows the CP to be connected
to a STARLAN NETWORK as an OUT connection. This can be connected to a STARLAN
NETWORK Extension Unit (NEU) IN connection, in a star configuration.
The design of the ZTN84 is not fully compatible with the daisy-chain arrangement of the
STARLAN NETWORK, since much of the daisy-chain circuitry was left off of the card. For
testing purposes, the card can be used in a limited daisy-chain arrangement, where the
ZTN84 is connected to a personal computer (PC) that possesses an Network Access Unit
(NAU). The ZTN84 and the PC should be the only two devices forming the LAN. The
daisy-chain circuitry was omitted in order to reduce cost and save board space. It is also the
architectural design of the system that the PBX be connected to the LAN by a NEU. The
NEU can either be local, in the telephone room with the switch, or in a remote office.
3-37
System Resources
The System Resource Circuit Packs (CP) are as follows:
●
Service Circuit (ZTN85)
●
Tone Detector (TN748)
●
Pooled Modem (TN758).
Service Circuit (ZTN85)
The Service Circuit CP provides the clock signals of the system. It also generates and
receives tones. The Service Circuit CP (Figure 3-19) consists of the following:
●
Bus buffers
●
Sanity and Control Interface (SAKI)
●
On-board microprocessor with external RAM
●
Clock circuit
●
Tone generator
●
Time slot table and counter
●
Tone detector ports
●
Port I/O and Sanity Check circuit.
The ZTN85 provides four touch-tone receivers, generates all tones for the system, and
supplies the system clocks. The ZTN85 can support up to 75 Dual Tone Multifrequency
(DTMF) dialers depending on call traffic; the TN748s might be required in heavy traffic
situations, even with less than 75 DTMF dialers. Each System 25 must contain one Service
Circuit CP. Power for the circuit pack (+5 volts dc) is provided on the backplane.
Bus Buffers: There are four bus buffers on the circuit pack. The clock
buffers interface three system clock signals (2.048 MHz, 8 kHz, and 160
bus. Two buffers interface the system tones (see Table 3-A) between the
Service Circuit CP. Music is not provided by the Service Circuit but can
port interface on a Tip Ring Line CP (ZTN78).
driver and receive
kHz) to the TDM
TDM bus and the
be provided via a
SAKI: This circuit functions the same as in the SAKI in the common circuitry for the
intelligent port circuits.
On-Board Microprocessor
microprocessor in the common
the dual-port RAM in the time
a tone. The external RAM also
with time).
With External RAM: This circuit functions the same as the
circuitry for the intelligent port circuits. In addition, it tells
slot table circuit the appropriate time slots in which to place
has work space for complex tones (i.e., those tones that vary
Clock Circuit: The clock circuit consists of a 20.48-MHz oscillator, various dividers, and
shift registers. The clock circuit runs independently from the rest of the Service Circuit
circuitry. The clock circuits start running when the circuit pack is first powered up and is
not controlled by the on-board microprocessor.
The output of the 20.48-MHz oscillator is fed to the clock divider. The divider divides by 10,
2560, and 128. These circuits produce the 2.048-MHz, 8-kHz, and 160-kHz clock signals,
respectively. The clock generator feeds these signals to the clock driver/receiver bus buffer
and the tone clock. The tone clock uses these signals to synchronize the counters in the tone
3-38
generator and time slot table circuits with the TDM bus.
Tone Generator: The tone generator consists of a digital signal processor (DSP), a
counter, and a dual-port tone RAM. The DSP operates at 10 MHz and produces .24 different
tones. The dual-port tone RAM stores these tones in 24 different addresses. The counter
under control of the tone clock causes the DSP to transmit one sample of each tone every 8kHz. The counter is synchronized to the TDM bus and is offset to provide delay needed for
access time.
Time Slot Table and Counter: The time slot table consists of a dual-port time slot table
RAM and a counter. The dual-port RAM (DPRAM) contains 256 different addresses. These
addresses correspond to the time slots on the TDM bus. The counter sequences through the
time slot table addresses in the dual-port RAM and causes the proper tone(s) to be output by
the dual-port tone RAM on TDM bus time slots.
Tone Detector Ports: The Service Circuit CP provides four Dual Tone Multifrequency
(DTMF) detector port circuit interfaces via the TDM bus. Each port circuit is connected to
an NPE serial input and output. Ports 0, 1, 2, and 3 are DTMF tone detectors with NPE
loop-around paths.
The four port circuits contain a DSP, NPE to DSP interface circuitry, a DSP restart circuit,
and an interrupt generator. One DSP implements two tone receivers.
The TDM bus signals are connected to the DSP in serial form from the NPEs by the DSP
interface circuit. The DSP controls the output clocking of the NPE. The system framing
signal is synchronized and connects to the DSP.
Port I/O and Sanity Check Circuit: This circuit interfaces the on-board microprocessor
to the port circuits and checks the sanity status of the DSPs of the port circuit.
3-39
TONE
GENERATOR
(DSP)
TIME
SLOT
TABLE
(DPRAM)
TONE
TABLE
(DPRAM)
TDM
BUS
LEADS
BUS
BUFFER
●
CIRCUIT
PACK
ADDRESS
LEADS
LEDs
SAKI
●
●
RAM
●
●
U-CONTROL
NETWORK
BUS
RED
YELLOW
GREEN
ADDRESS & DATA BUS
●
●
TONE
DETECTORS
NPE
SYSTEM
CLOCK
Figure 3-19. Service Circuit (ZTN85)
3-40
OUTPUT
REGISTERS
Tone Detector (TN748)
The Tone Detector Circuit Pack provides four touch-tone receivers and two general purpose
tone receivers that detect appropriate system and network tones on the TDM bus.
The Tone Detector CP consists of the same common circuitry as the intelligent port circuits
and the following unique circuits (see Figure 3-20):
●
Port I/O circuit
●
Port or DSP Sanity check circuit
●
Four touch-tone port circuits
●
Two general purpose tone detector ports
●
Two NPE loop-around test ports.
Up to a maximum of two Tone Detector CPs can be provided in the system.
Port I/O and Sanity Check Circuit: This circuit interfaces the on-board microprocessor
to the port circuits and checks the sanity status of the port circuits Digital Signal Processors
(DSPs).
Port Circuits: There are eight port circuits. Six port circuits are connected to Network
Processing Elements (NPEs). Port circuits 0,1, 4, and 5 are DTMF tone detector ports. Each
of the six port circuits has an associated Digital Signal Processor (DSP), NPE to DSP
interface circuitry, a DSP restart circuit and an interrupt filter. Port circuits 2 and 6 are
general purpose tone detector ports. Port circuits 3 and 7 provide digital Ioop-back testing of
each NPE on the circuit pack.
The NPE serializes TDM bus signals that are connected to the DSP in serial form from the
NPEs by the DSP interface circuit. Serial clock and data signals connect directly from the
NPE to the DSP. The system framing signal is synchronized and connects to the DSP.
The DSP restart circuit controls the DSPs. When the on-board microprocessor is not
functioning properly, the DSP restart circuit takes all of the DSPs out of service. It restarts
each individual DSP under control of the port 1/0 and sanity check circuit.
The touch-tone DSPs, under control of the on-board microprocessor, write data
synchronously to the NPEs. The interrupt filter detects valid touch-tone signals and allows
end-to-end transmission while blocking end-to-end touch-tone signaling.
3-41
\
/
●
❘
TDM
BUS
LEADS
●
NPE
0
BUS
BUFFERS
❘
●
❘
\
●
PORT CIRCUIT
0
PORT
CIRCUIT 3
NPE
1
PORT
CIRCUIT 7
●
PORT CIRCUIT
1
●
PORT CIRCUIT
4
●
PORT CIRCUIT
5
●
RAM
●
●
/
❘
❘
CIRCUIT
PACK
ADDRESS
LEADS
SAKI
❘
/
\
●
ON-BOARD
MICROPROCESSOR
PORT
I/O
CIRCUIT
/
●
\
●
●
RED
SANITY
CHECK
CIRCUIT
GREEN
LEDS
TOUCH-TONE
PORTS
YELLOW
●
PORT CIRCUIT
2
GENERAL
PURPOSE
TONE
DETECTOR
PORTS
PORT CIRCUIT
6
\
/
Figure 3-20. Tone Detector (TN748) Circuit
3-42
Pooled Modem (TN758)
The Pooled Modem Circuit Pack supports 0-300 and 1200 bits per second (bps) data speeds
and provides the following:
●
Circuitry to provide a signal compatible with the modulation formats of the 212series modems
●
Modem emulation (see below)
Capability
0-300 Asynchronous
300 Asynchronous
1200 Asynchronous
●
Data Module Mode
Low
300 Asynchronous
1200 Asynchronous
Modem control functions corresponding to 212A-series modem operations.
A maximum of two Pooled Modem CPs are allowed in a single cabinet (six in a 3-cabinet
system).
The Pooled Modem CP (Figure 3-21) consists of common circuitry and two conversion
resources. The conversion resource (port) allows communications between two dissimilar
endpoints. For example, the Pooled Modem CP enables a digital data endpoint linked to an
ADU connected to a port on the Data Line CP (TN726) to communicate with either a local
analog data endpoint, such as a personal computer with a modem, or a remote host via a CO
trunk connection. Each port has two connections to the TDM bus: one to the digital data
endpoint via an ADU data module, and the other to an analog endpoint.
TDM
BUS
LEADS
❘
❘
NPE
●
●
CONVERSION
RESOURCE
0
BUS
BUFFERS
❘
RAM
CIRCUIT
PACK
ADDRESS
LEADS
❘
❘
❘
SAKI
●
ON-BOARD
MICROPROCESSOR
CONVERSION
RESOURCE
1
RED
LEDS
●
GREEN
YELLOW
Figure 3-21. Pooled Modem (TN758) Circuit
3-43
Common Circuitry: The Pooled Modem common circuitry that includes all circuitry shown
on Figure 3-21 except the Conversion Resource circuitry provides the same general function
as the intelligent port common circuitry.
Conversion Resources: The two conversion resources (port circuits) are identical and
each contain the following:
●
Microprocessor
●
Transmit and Receive I-channel Controller (TRIC)
●
Universal Synchronous/Asynchronous Receiver and Transmitter (USART)
●
Data USART Clock (DUCK)
●
Digital Signal Processor (DSP).
The microprocessor controls an on-board data module and modem. This microprocessor
communicates with the port circuit microprocessor over a serial control channel. This
channel allows the on-board microprocessor to send messages to the port circuit
microprocessor specifying call startup information, option settings, information requests,
various test modes, and call termination information. It also allows the port circuit
microprocessor to inform the on-board microprocessor of various port circuit status
information.
The DUCK and TRIC interface I-channel information between the port circuit and the
remote data module. The microprocessor controls the operation of the DUCK and the TRIC
by programming their internal registers. The DUCK and TRIC together recreate the clock
and serial data stream from the remote data module, and process an on-board clock and
serial data stream for delivery to the remote data module. Control information,
handshaking, and RS-232 control leads is passed between the port circuit microprocessor and
the remote data module by the TRIC.
The USART interfaces the serial data stream of the DUCK to the conversion microprocessor.
The USART can be programmed by the microprocessor to operate synchronously or
asynchronously. T h e U S A R T a l s o d o e s t h e f o l l o w i n g t a s k s f o r t h e p o r t c i r c u i t
microprocessor:
●
Appends start and stop bits to parallel data received from the microprocessor in the
asynchronous mode
●
Converts serial data received from the DUCK to parallel data
●
Buffers data in both directions
●
Detects and generates break characters.
The DSP provides modem emulation. It interfaces the port circuit signal and the remote
modem. The microprocessor directs the DSP to execute one of many programs. The DSP
produces data, carrier detection, and timing information for the port circuit microprocessor.
3-44
Software
The System software consists of switched services, administrative, and maintenance
software. This software runs on top of the real-time operating system software.
Switched Services Software
The switched services software provides voice and data call processing. This software
resides in the Call Processor and Memory Circuit Packs (that are collectively referred to as
the Common Control circuitry), and in the 8-bit on-board microprocessors located in the port
and service circuits.
The switched services software uses the operating system to provide a process based,
message passing, execution environment. The operating system scheduler provides scheduling
for the software according to process priority.
Administrative Software
The administrative software provides the control for system rearrangement and change via
the System Administration Terminal (SAT). This software resides in the Memory Circuit
Pack and does the following functions:
●
Organizes the translation data for administrable entities in the system in a form
that can be viewed and changed at the System Administration Terminal.
●
Tests entered data for consistency with data previously entered in order to avoid
such errors as the assignment of the same extension number to two voice terminals.
An erroneous or inconsistent data entry is disallowed and an error message is
provided.
●
Causes the translation data to be downloaded, on command, to an optional Digital
Tape Unit (DTU).
Maintenance Software
The maintenance software provides automatic periodic testing of maintenance objects within
the system as well as consistency tests among the call status tables within the system. In
addition, demand testing is initiated when the system detects a condition requiring a need
for testing. Software tables are provided for storing error records. The records can be
accessed by maintenance personnel via the SAT. A Permanent System Alarm (a serious
error) causes an alarm indicator on the attendant console to light and an error record to be
stored in the error table.
Memory Allocation
The system software, like the hardware, is identified by release and version number. Each
version identifies a particular memory configuration for the release number. Main memory
is located in the Common Control circuitry. The operating system and error log software
resides on the Call Processor circuit pack and the remaining administration and call
processing software is on the Memory circuit pack.
Real-Time Constraints
Real-time constraints are a function of the speed of the common control circuitry and the
traffic load. The switch is designed so that many time-consuming and repetitious functions
are performed by processors in the port and service circuit packs, thus relieving the common
control circuits.
3-45
Traffic load, defined as the sum of static and dynamic loads, is a function of the number of
features that are executed, the frequency with which they are executed, the system
configuration, and the instantaneous (peak) call processing load. The configuration
contribution to load is known as dynamic load. The static load consists of maintenance and
audit routines.
Software Partitioning
System 25 software is comprised of various modules, each supporting a particular process.
Typical modules (referred to as tasks) include the following:
●
Administration
●
Station Call Processing
●
Station Message Detail Recording (SMDR) Call Record Processing
●
Trunk Call Processing
●
Dial Plan Manager
●
Event Timer
●
Save/Restore
●
Maintenance and Audit Functions.
(Administration
function)
As shown on Figure 3-22, software tasks associated with the Memory Circuit Pack are
Administration and Feature Code Modules, which includes Station Call Processing. Each
task controls the storage and movement of data and messages between associated elements
within the system.
Memory Circuit Pack
Administration: Provides for administration of station and system features. This software
also supports maintenance procedures related to error checking and diagnosing trouble.
Feature Code Modules: Includes the software that receives and sends data to/from the
Operating System, as well as control of all voice and data features supported by the system.
Station Call Processing includes the processing of messages and data associated with voice
terminal on-hook/off-hook indications, associated port identifications, button and LED
operations, etc. The SMDR software generates SMDR records associated with a particular
call. The records are then sent to the System RAM for storage and then to the SMDR
output channel.
Call Processor Circuit Pack
System RAM: Provides for the storage of the following:
●
Variables for the various software tasks
●
System
●
Error Records
●
Feature Code Data
●
Stack.
translations
3-46
Error Logger: Prioritizes and stores system errors. The errors are stored in three error
records (located in System RAM), that are:
Permanent System Alarms
●
●
Transient System Errors
●
Most Recent System Errors.
The Error Logger lights the Alarm LED (located on the Attendant Console) when a serious
error is detected.
Operating System (OS): Controls all message and data flow to/from the Memory Circuit
Pack, the Arch Angel Driver Interface, to the microprocessors on the port circuit packs, and
to the RS-232 driver interfaces. Messages destined for a particular task are queued until the
associated task can receive them. When a task has completed a particular process, the next
message is obtained from the message queue of the tasks. The OS provides an interval timer
that is used to time tasks. Processes that exceed the set interval (about 60 seconds) are
terminated by the OS.
Arch Angel Driver Interface: Provides an interface between the OS and Network
Control.
RS-232 Driver Interface: Handles the flow of information between the Call Processor
Circuit Pack and the peripheral equipment of the system (i.e., System Administration
Terminator Advanced Administration PC, Digital Tape Unit, and SMDR Output Device.)
TDM BUS
Provides an electronic link between the system port circuits (including System Resources)
and between the Call Processor Circuit Pack and port circuits.
Port Circuit Packs
Each port circuit pack has on-board software that provides for the sending/receiving of
Network Control messages and data. Circuit pack status messages are also sent to the
Network Control software.
Step-By-Step Call Description
The following is a description of a call originated between two multilane voice terminals.
1.
A microprocessor on a station port circuit pack (port controller) continually
monitors associated port circuits for switchhook status/change and button presses.
2.
When a user goes off-hook, the port controller detects the change.
3.
The port controller s e n d s a n o f f - h o o k u p - l i n k m e s s a g e a l o n g w i t h p o r t
identification to the Call Processor Network Controller (CPNC) via the TDM bus.
4.
The CPNC accepts the message and forwards it to the Operating System (OS) via
the Arch Angel Driver Interface.
5.
The OS checks a message directory to determine which task (i.e., software module)
is to receive the message. A function of the OS referred to as the “transformer”,
determines it has a message for the Station Call Processing task and queues the
message in RAM.
3-47
`
FEATURE CODE MODULES
/
MEMORY
CIRCUIT
PACK
—
—
—
—
ADMINISTRATION
(TASK)
—
—
—
—
—
▲
— —
—
▼
STATION
CALL
PROCESSING
(TASK)
▲
— — — — —
▼
SYSTEM RAM
CALL PROCESSOR
CIRCUIT PACK
▲
▼
OPERATING
●
●
—
\
●
SMDR
PROCESSING
(TASK)
—
—
—
—
—
—
—
ERROR
LOGGER
MESSAGE
SEND/RECEIVE
SYSTEM
▲
▼
▼
ARCH ANGEL
RS-232C
DRIVER
DRIVER
INTERFACE
INTERFACE
▲
▲
▼
NETWORK
CONTROL
SOFTWARE
▲
— — — — — —
CONTROL ❘
CHANNEL ❘
MESSAGES
— — — — — —❘
▲
—
—
—
—
—
—
—
—
—
—
—
—
TDM BUS
—
—
PORT
CIRCUIT
PACKS
▼
ON-BOARD
SOFTWARE
▲
❘
❘
❘
PORT
❘
STIMULI
— — — — — — — — — — — — — — —❘ — — — — — — — — — — — — —
▼
▼
\
\
/
/
SAT,
VOICE TERMINALS,
SYSTEM I/O
DTU, SMDR
CO FACILITIES,
RS-232C DEVICES
Figure 3-22. System Software Partitioning
3-48
6. The Station Call Processing task retrieves its message and interprets it as a call
origination. The task determines whether there is an idle call appearance button
(System Access button) on the called voice terminal. If so, two available time slots
are reserved for the connection.
7. The task sends downlink messages to the port circuit via the OS. The messages
instruct the port circuit to listen for dial tone on a specified time slot and to light
the call appearance status LED on the terminal.
8. When the user dials the first digit, the port circuit determines the digit dialed. It
then listens to appropriate time slots on the TDM bus for the two tones used to
generate an equivalent DTMF signal. It then removes dial tone and feeds the
DTMF signal back to the user until the user releases the button.
9. The port circuit sends an up-link message with each digit dialed to the OS that
routes them to the Dial Plan Manager (DPM).
10. The DPM collects the dialed digits and determines that the call is a station-tostation call.
11. When the DPM collects enough digits to identify an extension number it stops
collecting digits.
Note: If the extension number dialed is invalid, the DPM sends a down-link
message to the port circuit instructing it to listen to time slot 07 (Reorder
Tone) that is then heard by the user. Go to Step 18.
12.
A down-link message is sent to the originating port instructing it to listen to time
slot 06 (busy) or 08 (ringing), as appropriate. Go to Step 18 for Busy Tone or an
unanswered call.
13.
Station Call Processing sends a down-link rnessage to the station port circuit pack
associated with the called extension to turn on the ringer of the terminal, and to
flash the call appearance LED.
14.
When the called party lifts the receiver, the associated port circuit pack controller
sends a off-hook message to the OS as before.
15.
The Station Call Processing task, when it receives the message interprets the offhook message as an answer.
16.
The task sends a down-link message to the called port circuit to turn off the ringer
and to change the flashing LED to steadily lighted.
17.
Down-link messages are sent to the port circuits assigning talk and listen time
slots for the connection.
18.
When either of the parties hangs up, the associated port circuit controller sends an
up-link message to the Station Call Processing task.
19.
Station Call Processing interprets the on-hook message as the end of the call
20.
The task then sends a down-link message to the port circuit pack controllers to
disconnect the time slot connections and turn off the LEDs associated with the
calls.
3-49
LIST OF FIGURES
Figure 4-1.
System
25
Figure
System
Cabinet
Figure 4-3.
System
Circuit
4-2.
Cabinets
(J58901A)—Three
.
.
.
.
4-2
.
.
.
.
.
.
4-3
.
.
.
.
.
.
.
.
4-5
Figure 4-4.
Model 2500 Series (Analog) Voice Terminals . . . . . . . . . .
4-14
Figure 4-5.
Model
Figure 4-6.
420
Figure 4-7.
5-Button
Figure
10-Button
Voice
Terminal
(7303H01D)
34-Button
Voice
Terminal
(7305H01D)
7101A
(Analog)
Speakerphone
Voice
Voice
Voice
Terminal
.
.
.
.
(7302H01D)
.
.
Terminal
Terminal
.
.
.
Configurations
.
.
.
Pack
View
System
.
4-8.
(J58901A)—Rear
Cabinet
.
.
.
.
.
.
.
.
.
.
.
.
.
.
4-15
.
.
.
.
.
.
.
.
.
.
.
.
4-17
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
4-21
.
.
.
.
.
.
4-23
.
.
.
.
.
.
.
.
.
.
4-25
(7305H02D)
.
.
.
.
.
.
.
.
.
4-27
.
.
.
.
.
.
.
.
.
4-29
Figure 4-12. BIS Voice Terminal with Display (7305H04B) . . . . . . . . . .
4-31
Figure 4-13. HFAI Voice Terminal (7309H01B) . . . . . . . . . . . . . .
4-33
Figure 4-14. Ten Button MET Set (2991C05) . . . . . . . . . . . . . . .
4-35
Figure 4-15. Ten Button MET With Built-In Speakerphone (2993C04) . . . . . .
4-37
Figure 4-16. Twelve Button MET Set (7203M) . . . . . . . . . . . . . .
4-39
Figure
.
4-43
Figure 4-18. Trunk Access Equipment (TAE) Connections . . . . . . . . . .
4-47
Figure
4-19.
617A
Figure
4-20.
Typical
Figure 4-9.
Figure
4-10.
34-Button
Figure
4-11.
BIS
4-17.
Deluxe
Voice
Voice
Terminal
Asynchronous
Data
Terminal
(7305H03B)
Unit
(ADU)
.
.
(MD)
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
4-49
.
.
.
.
.
.
.
.
.
.
.
.
.
.
4-50
Figure 4-21. On-Premises Single-Line Voice Terminal Connections . . . . . . .
4-52
Figure 4-22. Out-Of-Building Single-Line Voice Terminal Connections . . . . . .
4-53
Figure 4-23. Off-Premises Station Single-Line Voice Terminal Connections . . . .
4-54
Figure 4-24. On-Premises 7300H Series Multiline Voice Terminal
Connections . . . . . . . . . . . . . .
Station
SIP
Interconnect
Connections
Panel
.
.
.
.
.
.
.
.
.
.
4-55
Figure 4-25. Out-Of-Building 7300H Series Multiline Voice Terminal
Connections . . . . . . . . . . . . . . .
.
.
.
.
.
.
4-56
Figure 4-26. Ten Button MET Set Connections . . . . . . . . . . . . . .
4-57
Figure 4-27. Stand-Alone Remotely Powered Voice Terminal Connections . . . .
4-58
Figure 4-28. Typical ADU Connections–Supporting Data Terminal And Single-Line
Voice Terminal . . . . . . . . . . . . . . . . . . . .
4-59
-ii-
Figure 4-29. Typical ADU Connections—Supporting Data Terminal And 7300H Series
Multiline Voice Terminal . . . . . . . . . . . . . . . . .
4-60
Figure
.
4-61
Figure 4-31. Z3A1/2/4 ADU Local Power Connections . . . . . . . . . . .
4-62
4-30.
Typical
MADU
Connections
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
LIST OF TABLES
TABLE
4-A.
Total
Port
TABLE
4-B.
System
TABLE
4-C.
Summary
Circuit
Circuit
Packs
.
.
.
.
.
.
.
.
.
.
.
.
.
4-6
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
4-11
Terminals
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
4-13
TABLE 4-D. Model 2500 Series Voice Terminal Adjuncts . . . . . . . . . .
4-18
TABLE
4-E.
Supplemental
Voice
TABLE
4-F.
Asynchronous
Data
Voice
.
System
.
of
Packs
Per
Terminal
Power
Supplies
Units
.
.
.
.
-
iii
-
.
.
.
.
.
.
.
.
.
.
.
.
.
.
4-41
.
.
.
.
.
.
.
4-44
SECTION 4—HARDWARE DESCRIPTION
This Section provides descriptions of System 25 hardware components and their functions.
The hardware is covered under the following major headings:
●
System Cabinets: Includes Circuit Pack (CP) carriers and CPs.
●
Terminal Equipment: Equipment that can be connected to voice or data station
ports.
●
Peripheral Equipment: Equipment that can be connected to the Call Processor
CP.
●
Auxiliary Equipment: Service- and feature-related supporting equipment.
●
Connectivity: E q u i p m e n t a n d a r r a n g e m e n t s f o r i n t e r c o n n e c t i n g t h e v a r i o u s
elements of System 25 hardware.
Note: Equipment that indirectly associated with a specific feature or service covered
in Section 2 is also described there; such equipment is noted in this Hardware
Description section, with a reference to the appropriate heading in Section 2.
All system hardware, except Cabinet 1 equipped with Call Processor, Memory, and Service
Circuit CPs, and associated cables is optional.
A listing of Product Element Codes (PECs), Apparatus Codes, and Comcodes is provided in
Section 5, “Parts Information.”
System Cabinets (J58901A1)
The system can consist of one, two, or three cabinets (Figure 4-1). Each cabinet contains its
own power supply and cooling system. A CP carrier frame is integrated into each cabinet.
Depending on the circuit pack complement, the cabinet/circuit pack configuration is as
follows:
●
Cabinet 1 (always required)—Contains the Call Processor, Memory, and Service CPs
of the system and can also contain up to nine port CPs.
●
Cabinet 2 or 3 (optional)—Provides mounting for up to 12 port CPs each.
The CPs receive power, control signals, and data via the backplane bus of the carrier and
associated 25-pair connector interfaces. In multiple cabinet systems, the backplane buses are
linked with a bus extender cable (J58901A1, L3).
The Call Processor and Memory CPs are also connected via a ribbon connector cable referred
to as the “Front Plane Bus.’’ This cable connects the front edges of the two CPs.
The cabinets have a brown front cover with beige top and sides. The front cover has a
system identification stripe across the top.
Each cabinet is constructed of sheet metal and is 13 inches high, 17 inches wide, and 21
inches deep and weighs about 75 pounds fully loaded. A three cabinet system occupies a
vertical space of about 40 inches. It is recommended that the cabinets be placed on a deskor table-top. They must not be placed on a floor where cleaning solutions and dirt can get
into them. Refer to Section 6, “Environmental Requirements” for equipment area
considerations and associated floor plan recommendations.
4-1
The front cover of the cabinet is secured by four screws. These screws must be loosened
slightly before the cover can be removed. When removed, access is provided to the CPs, a
replaceable air filter mounted just under the CP carrier frame, and two cooling fans. The
cooling fans are mounted on an assembly that, when unscrewed, provides access to the power
supply. Air intake is at the bottom of the cabinet and exhaust is vented at the left side of
the front cover.
— CABINET 3
(12 PORT CIRCUIT PACKS)
— CABINET 2
(12 PORT CIRCUIT PACKS)
— CABINET 1
( CALL PROCESSOR, MEMORY,
SERVICE CIRCUIT, 9 PORT
CIRCUIT PACKS)
Figure 4-1. System 25 Cabinets (J58901A)—Three Cabinet System
4-2
Each cabinet has its own power supply mounted to the left of the CP carrier. The power
supply is 3 inches wide and weighs about 9 pounds. Voltage and current supplied to the
carrier are: +5 V dc at 35A, -5 V dc at 3A, -48 V dc at 3A, and 90 V ac at 0.16A.
On the front of the supply is a green Light Emitting Diode (LED) that, when lighted,
indicates that the +5 V de is available and within limits. The LED can be viewed through
the slotted area on the front cover, and is just behind the fan located at the top left edge of
the cabinet.
Mounted on the back of the cabinet (Figure 4-2) is the copper grounding block with four
terminating positions, an AC input power receptacle, a power On/Off switch [(l)=ON,
(0)=OFF] and twelve 25-pair connectors. The ground block is connected to DC ground on the
carrier backplane at a location near the power supply. The 25-pair connectors provide an
interface between cross-connect wiring and the CPs immediately behind each connector.
Two slots are provided in the rear cover just above the 25-pair connectors for the Time
Division Multiplex (TDM) bus extender cable. The TDM cable is used to connect 2- or 3cabinet systems together in a daisy-chain configuration and provides control and data
signals between Cabinets. The Cabinet 1 ground block is connected to the single point
ground of the system using 6 AWG wire. Separate 6 AWG wires are then connected from
the Cabinet 1 ground block to Cabinet 2 and 3 ground blocks. The Cabinet 1 ground block is
also connected to the Coupled Bonding Conductor. An information label is provided across
the top portion of the rear panel on each cabinet. The label provides cabinet identification,
input electrical requirements, caution and warning notes, and FCC and UL labels.
CAUTION & WARNING
LABELS
SYSTEM 25
J58901A
GROUNDING
BLOCK
120 VOLTS AC
❑
6 A M P S
❑
60
❑
HZ
❘
12
FCC LABEL
11
❘
10
9
8
7
6
5
❘
4
3
2
1
❍
❍
INPUT
AC POWER
RECEPTACLE
25-PAIR
CONNECTORS
Figure 4-2. System Cabinet (J58901A)—Rear View
4-3
❍
❍
ON/OFF
SWITCH
Cabinet 1 (Control and Port Circuits)
Cabinet 1 (Figure 4-3) is always required. It provides mounting space for 12 CPs and can
support a small telecommunications system (eg., 50 to 60 stations and 10 to 15 trunks). It
contains a Memory and Call Processor that together are referred to as the Common Control
(CC), a Service Circuit, and up to 9 port CPs. The Memory, Call Processor, and Service
Circuit must be mounted in CP slots 1, 2, and 3, respectively. Slots 4 through 12 (9 total)
provide mounting for the various port CPs that can be used. Any port CP can be mounted in
any of these 9 slots. The Memory and Call Processor are electrically linked by a ribbon cable
(Front Plane B U S) that loops between their front edges. The Tone Detector and Pooled
Modem CPs of the system (referred to as System Resource Cl’s ) can also be mounted in the
port CP slots. Circuit packs are described in this Section under the heading “Circuit Packs”.
Cabinet Address Plug
An address plug is provided on the middle of the backplane of each cabinet (accessible after
removing the top rear cover) and is used to designate the cabinet number to the software.
When plugged into the designated area at CP slot 5, the cabinet is identified as Cabinet 1; at
slot 6 as Cabinet 2, and at slot 7 as Cabinet 3.
Cabinets 2 and 3 (Port Circuits)
Cabinet 2 and Cabinet 3 (Figure 4-3) can be provided. The cabinets provide mounting space
for additional port CPs (12 maximum each) required for larger systems. The Tone Detector
and Pooled Modem CPs can also be mounted in these cabinets. These cabinets are simply
stacked on top of Cabinet 1.
Table 4-A summarizes port CP capacity of 1-, 2-, or 3-cabinet systems.
4-4
CIRCUIT PACK SLOTS
1
2
3
5
4
6
7
8
9
10
11
12
PORT CIRCUITS
POWER
SERVICE CIRCUIT
CALL PROCESSOR
(a) CABINET 1 - (MOUNTING FOR 3 CONTROL & 9 PORT CIRCUIT PACKS)
CIRCUIT PACK SLOTS
1
2
3
4
POWER
SUPPLY
5
6
7
8
9
10
11 12
PORT CIRCUITS
(b) CABINET 2 OR 3 - (MOUNTING FOR 12 PORT CIRCUIT PACKS)
NOTES :
1. REFER TO TECHNICAL SPECIFICATIONS, SECTION 8 FOR CIRCUIT PACK UNIT LOAD INFORMATION.
2. DIVIDE THE TOTAL NUMBER OF VOICE TERMINAL AND TRUNK CIRCUIT PACKS BETWEEN THE
CABINETS USED.
3. MOUNT VOICE TERMINAL CIRCUIT PACKS FROM THE RIGHT, TRUNK CIRCUIT PACKS FROM THE LEFT.
4. COMMON CONTROL CIRCUIT PACKS *
● CALL PROCESSOR (ZTN82 IN R1V1)
● MEMORY (ZTN81 IN R1V1)
(ZTN127 IN R1V2)
(ZTN128 IN R1V2)
5. SYSTEM RESOURCE CIRCUIT PACKS #
● POOLED MODEM (TN758)
● SERVICE CIRCUIT (ZTN85)
● TONE DETECTOR (TN748)
(MAXIMUM 2-PER CABINET)
6. PORT CIRCUIT PACKS †
● DATA LINE (TN726)
● TIP RING LINE (ZTN78)
● GROUND START TRUNK (ZTN76)
● ATL LINE (ZTN79)
● MET LINE (TN735)
● LOOP START TRUNK (ZTN77)
● ANALOG LINE (TN742)
● DID TRUNK (TN753)
● AUXILIARY TRUNK (TN763)
● TIE TRUNK (TN760B)
● STARLAN INTERFACE (ZTN84)
* CIRCUIT PACKS MUST BE MOUNTED IN CABINET 1, IN THE SLOTS INDICATED
† UNIVERSAL PORT CIRCUIT PACKS CAN BE MOUNTED IN ANY AVAILABLE PORT SLOT
# SERVICE CIRCUIT MUST BE MOUNTED IN SLOT 3 AS SHOWN. POOLED MODEM AND
TONE DETECTOR MAY BE MOUNTED IN ANY PORT CIRCUIT SLOT.
Figure 4-3. System Circuit Pack Configurations
4-5
TABLE 4-A. Total Port Circuit Packs Per System
NUMBER
OF
CABINETS
CABINET
NUMBER *
PORT
CIRCUIT
PACKS
TOTAL PORT
CIRCUIT
PACKS †
1
Cabinet 1
9
9
2
Cabinet 1
9
21
Cabinet 2
12
Cabinet 1
9
Cabinet 2
12
Cabinet 3
12
3
33
* Cabinet 1 (always required)—Provides mounting for Memory,
Call Processor, Service Circuit, and Port CPs including
Tone Detectors and Pooled Modems.
Cabinet 2 and 3 (Optional)–Provides mounting for Port CPs
including Tone Detectors and Pooled Modems.
† The Number of Ports per CP is specified in the CP descriptions.
4-6
Circuit Packs
Required Circuit Packs:
The following CPs are provided with all systems and must be mounted in Cabinet 1:
●
ZTN82 (V1) or ZTN128 (V2) Call Processor
The ZTN82 or ZTN128 (one per system) provides a central processing unit, RandomAccess Memory (RAM) (memory) for call and feature processing, interrupt controller,
programmable timers, real time clock, status display, processor bus interface, and
four interface ports. The ports provide the following interfaces:
—Port l—System Administration Terminal (SAT)
—Port 2—Station Message Detail Recording (SMDR) equipment
—Port 3—Digital Tape Unit
—Port 4—Reserved for future use.
The Call Processor also provides -48 V de control on ports 7 and 8 for Emergency
Transfer Units.
The Call Processor CP must be mounted in slot 2 of Cabinet 1.
●
ZTN81 (Vl) or ZTN127 (V2) Memory
The ZTN81 or ZTN127 (one per system) provides 512K of read-only memory. The
Memory CP provides for the software associated with system operation, including
call processing, administration, and maintenance.
The ZTN81 or ZTN127 provides a built-in TDM bus terminator; an earlier version
(ZTN81 ) did not.
The Memory CP must be mounted in slot 1 of Cabinet 1.
●
ZTN85 Service Circuit
The ZTN85 (one per system) provides four touch-tone receivers, generates all system
tones, and supplies the system clocks. The ZTN85 can support up to 75 Dual Tone
Multifrequency (DTMF) dialers such as 2500-type voice terminals and touch-tone
(incoming) tie trunks.
The Service Circuit CP must be mounted in slot 3 of Cabinet 1.
Optional Circuit Packs:
The following CPs are optional and can be mounted in any other CP slot.
System Resource Circuit Packs
●
TN748 Tone Detector
Provides four touch-tone receivers. The TN748 is required in addition to the ZTN85
Service Circuit when more than 75 Dual Tone Multifrequency (DTMF) dialers are to
be provided in a system. It might be required in high traffic situations if a system
has less than 75 DTMF dialers. Up to two TN748 Tone Detectors can be provided in a
system.
4-7
●
TN758 Pooled Modern
Provides two integrated 212-modem compatible conversion resources for switched
connections between analog endpoints (modems), or a digital endpoint and an analog
endpoint. A maximum of two TN758s (4 conversion resources) is permitted in each
cabinet.
Station Port Circuit Packs
●
TN726 Data Line
Provides eight ports for Asynchronous Data Units (ADUs). Used for in-building
service within 2000 feet of the system cabinets. Data speeds from 300 bps to 19.2
Kbps are supported. Service beyond 2000 feet at less than 19.2 Kbps is supported; see
Section 5 “Technical Specifications.” Extends a serial communications link from the
system to data equipment over standard station wiring.
●
TN735 MET Line
Provides four ports for Multibutton Electronic Telephone (MET) sets. Used for inbuilding service within 1000 feet of the system cabinets.
●
TN742 Analog Line
Provides eight ports for single-line voice terminals with or without a message
waiting lamp. Also supports Off Premises Stations (OPS) and out-of-building
service. Auxiliary equipment interfaces are also supported. Used for service within
24,000 feet of the system cabinets. Five voice terminals can be bridged onto each
port. Only two terminals can be off-hook simultaneously on each port, otherwise
transmission can be degraded.
Note: The Off-Premises-Stations must be FCC registered.
●
ZTN78 Tip Ring Line
Provides eight ports for single-line sets with or without message waiting lamps.
Used for in-building nonbridged voice terminal service within 2000 feet of the system
cabinets.
Note: Equipment connected to the ZTN78 Tip Ring Line CP must meet the
following requirements:
●
●
AC impedance: 600 ohms
●
DC current: Less than 30ma at 48 volts
●
Ringer Equivalence Number (REN): Less than 1.15 (Set plus adjuncts)
ZTN79 ATL Line
Provides eight ports for MERLIN® voice terminals. Used for service within 1000 feet
(305 m) of the system cabinets. Off-premises extensions are not supported. Out-OfBuilding stations require In-Building and In-Range Out-of-Building (IROB) units.
●
ZTN84 STARLAN Interface (V2)
Provides four ports for interface with AT&T STARLAN NETWORKs. It supports
four connections between data endpoints connected to the PBX and data endpoints
connected to the STARLAN NETWORK. The Network Extension Unit must be
colocated with the System 25 cabinets.
4-8
Trunk Port Circuit Packs
●
TN753 DID Trunks
Provides eight ports for immediate-start or wink-start Direct Inward Dialing (DID)
trunks.
●
TN760B Tie Trunks
Provides four ports for Type 1 E&M, Type 1 E&M Compatible, or Type 5 Simplex tie
trunks. Operating protocols include automatic, immediate-start, wink-start, or delay
dial. The TN760B contains option switches for supporting the following signaling
formats:
– Type 1 E&M Standard (Unprotected)
– Type 1 E&M Compatible ([Unprotected)
– Type 1 E&M Compatible (Protected)
– Type 5 Simplex
●
TN763
Auxiliary
Trunk
Provides four ports for on-premises auxiliary equipment (paging systems and
dictation systems).
●
ZTN76 Ground Start Trunk
Provides eight ports for ground start Central Office (CO), Foreign Exchange (FX), or
Wide Area Telephone Service (WATS) trunks
●
ZTN77 Loop Start Trunk
Provides eight ports for loop-start CO, FX, or WATS trunks.
Refer to Section 9, “Glossary” for Ground Start and Loop Start definitions. Ground Start
trunks are recommended for use where possible.
Trunk specifications are provided in Section 5, “Technical Specifications.”
Circuit Pack Compatibility
The following System 75 CPs can be used in System 25, if required:
●
The TN742 Analog Line can be used instead of the ZTN78 Tip Ring CP. The TN742
supports bridged stations and out-of-building or Off-Premises Stations (OPS), the
ZTN78 does not.
●
The TN762B Hybrid Line (Version 4 or later) can be used instead of the ZTN79.
●
The TN747 CO Trunk can be used instead of the ZTN76 (Ground Start Trunk) or the
ZTN77 (Loop Start Trunk).
4-9
Circuit Pack Features
All system CPs have the following features:
●
Solid-state circuitry mounted on 7.6 by 14.1-inch printed wiring board (TN-type)
●
Color coded face plate labels identify the CP type and function (White = Control,
Purple = Port or System Resource)
●
Individual circuit functions all contained on one CP
●
Metal tab for grounding
●
Locking tab-type handle provides easy insertion or removal of a CP
●
Port CPs can be inserted or removed with power “On” and the system processing
calls. Only the calls utilizing circuits on a removed CP will be affected.
Note: Power must be turned off when replacing the following CPs:
— Memory (ZTN81 or ZTN127)
— Call Processor (ZTN82 or ZTN128)
— Service Circuit (ZTN85)
●
Status
LEDs
— Port CPs:
●
Red—“On” several seconds during power up and test, “Off” with test
pass. After test pass, “On” if fault in CP is subsequently detected.
●
Green—“On” indicates resource available (port is translated)
●
Yellow—“On” indicates a call in progress. “Off” when not in use.
●
All LEDs “Off’’—CP is not translated.
— Common Control CPs:
●
Memory CP: Red status LED only.
“On” several seconds during power up and test, “Off” with test pass.
After test pass, “On” if fault in CP is detected.
●
Call Processor CP: Green Status LED only.
“Off” for several seconds during power up and test, then lamp flashes
to indicate an “OK” state. Steady “Off” or “On” indicates a problem.
— System Resource CPs:
●
Service Circuit CP:
Similar to port CPs except yellow LED flashes to show system clock is
active and is steadily “On” when a tone receiver is in use. “Off”
indicates a problem.
●
Modem Pool and Tone Detector CPs:
Same as Port CPs.
4-10
Table 4-B lists CPs that can be used with System 25. A description of each CP is provided in
Section 3, Functional Description.
TABLE 4-B. System Circuit Packs
CIRCUIT PACK
TN726
TN735
TN742
TN748
TN753
TN758
TN760B
TN763
ZTN76
ZTN77
ZTN78
ZTN79
ZTN81 or ZTN127
ZTN82 or ZTN128
ZTN84
ZTN85
TITLE
CIRCUIT
PACK
TYPE *
NUMBER
OF
PORTS
P
P
P
R
P
R
P
P
P
P
P
8
4
8
4
8
2
4
4
8
8
8
8
Data Line
MET Line
Analog Line
Tone Detector †
DID Trunk
Pooled Modem
Tie Trunk
Auxiliary Trunk
Ground Start Trunk
Loop Start Trunk
Tip Ring Line
ATL Line
Memory
Call Processor ‡
STARLAN Interface
Service Circuit
P
C
C
P
R
4
4
* P = Port, C = Control, R = System Resource.
† System Resource Circuits (Tone Detector, Pooled Modem, Service
Circuit) ports are internal to the system. These ports are not
connected to external equipment via 25-pair connectors.
‡ Provides four channels for the peripheral equipment of the
system:
(1) System Administration Terminal
(2) SMDR Output Device
(3) Digital Tape Unit
(4) reserved for future use.
4-11
Terminal Equipment
Terminal equipment is connected to System 25 station (voice or data) ports. It is made up of
the following groups:
●
Voice Terminals
— Single-Line
— Multiline (MERLIN sets and MET sets)
●
Voice Terminal Adjuncts
●
Attendant Consoles
●
Asynchronous Data Units (for interface with data terminals).
This subsection provides information on all components in each group or contains references
to the Section where information can be found.
Voice Terminals
System 25 supports a wide range of voice terminals, including industry standard touch-tone
single-line sets and MERLIN rnultiline sets.
In addition to providing basic telephone service (placing and answering calls), voice
terminals can also be used to activate many system features. The voice terminals supported
by System 25 are listed in Table 4-C and described in individual subsections.
Single-Line Voice Terminals
Single-1ine voice terminals can have only one call appearing at the terminal at a time.
All information (voice and control signals) transmitted to and from a single-line voice
terminal is in analog form over a single pair of wires (called tip and ring). Power for these
voice terminals is also provided over this pair. The ZTN78 Tip Ring Line CP or TN742
Analog Line CP converts the analog signals to digital format before placing them on the
TDM bus. The dial pad on the single-line voice terminals is a touch-tone pad that generates
Dual Tone Multifrequeney (DTMF) signals.
The following subsections provide descriptions and illustrations of the single-line voice
terminals supported by System 25.
Model 500 Series (V2)
The Model 500 Series consists of conventional rotary dial telephones. They are recommended
for use as a Power Failure Transfer (PFT) stations if the PFT trunk does not support
touch-tone dialing. A 55A1 Ground Start button must be used with these sets if the PFT
trunk is ground start. Rotary set users cannot do any procedures that require pressing the *
or # buttons. The following 500 Series sets are supported by System 25; these sets are
similar in appearance to the 2500 sets shown in Figure 4-4 except for their rotary dials.
●
Model 500MM-Basic desk set
●
Model 500SM—Desk set with 4A Speakerphone compatibility
●
Model 554BMPA—Basic wall set.
4-12
TABLE 4-C. Summary of Voice Terminals
TERMINAL
TYPE *
Single-Line
Tip Ring
(Analog)
†
420
500MM
500SM
554BMPA
2500MMGB
2500DMGC
2500SM
2514 BMW
2554BM
7101A
MERLIN
Multiline
7300H Series
(Hybrid)
MET ‡
Multiline
(Hybrid)
DESCRIPTION
MODEL
Memory Set W/Built-In Speakerphone
Rotary Desk Set
Rotary Set Compatible
with 4A Speakerphone
Rotary Wall Set
Basic Touch-Tone Desk Set
Basic Desk with Message Waiting
Basic Desk Compatible
with 4A Speakerphone
Basic Desk with Headset Jack
Basic Wall
Desk or Wall
CIRCUIT PACK
INTERFACE
ZTN78
TN742
7302H01D
7303H01D
7305H01D
7305H02D
7305H03B
7305H04C
7309H01B
5-Button
10-Button
34-Button
34-Button Deluxe
BIS Set (Built-In Speakerphone)
BIS Set with Display
HFAI Set (Hands-Free-Answer
on Intercom)
ZTN79
2991C05
2991D05
2993C04
7302M
10-Button
10-Button
10-Button
12-Button
TN735
(Desk)
(Wall)
with BIS
(Desk)
* System 25 supports several voice terminals that are no longer orderable. These
include MET sets and the 34-button (basic) MERLIN set.
† The system supports equivalent industry standard touch-tone single-line sets. Voice
terminals connected via the ZTN78 Tip Ring Line CP must have a REN less than or
equal to 1.20 A/B.
‡ The 2991C04 set [with Busy Lamp Field (BLF)] will not operate with System 25
unless specially modified.
4-13
Model 2500 Series
Each of the following Model 2500 voice terminals come equipped with:
●
Handset
●
Touch-Tone Dial (Dual Tone Multifrequency - DTMF)
●
Ringer Volume Control.
Several 2500 series voice terminals are shown in Figure 4-4. Supported 2500 series sets
include the following:
●
Model 2500 DMGC—Desk Set with message waiting indicator and Recall button for
timed switchhook flash
●
Model 2500MMGB—Basic desk set
●
Model 2500 SM—Basic desk set that can be used with a 4A Speakerphone.
●
Model 2514 BMW—Basic desk set equipped with built-in headset speaker jack.
●
Model 2554 BM—Basic wall-mounted set.
Adjuncts: Refer to Table 4-D.
HANDSET
TOUCH-TONE
TELEPHONE
DIAL
2500MMGB BASIC DESK SET
2554BM
BASIC
WALL SET
RINGER
VOLUME
MESSAGE
INDICATOR
RECALL
2500DMGC DESK SET WITH
EXTRA FEATURES
Figure 4-4. Mode1 2500 Series (Analog) Voice Terminals
4-14
Model 7101A (MD) (PEC 3170-00M)
The Model 7101A (Figure 4-5) is about 2-3/4 inches wide, 3-1/2 inches high, and 8-1/2 inches
long. The set comes equipped with the following:
●
Handset
●
Touch-Tone Dial
●
Message Indicator
●
Tone Ringer with Volume Control
●
Two Fixed Feature Buttons
— Recall—Used to place a call on hold and to obtain recall dial tone for
Conference, Transfer, and other features accessible by feature access code.
— Disconnect—Used to disconnect one call and immediately obtain dial tone for
another call.
Adjuncts: None
Figure 4-5. Mode1 7101A (Analog) Voice Terminal
4-15
420 Speakerphone (V2)
The 420 Speakerphone (Figure 4-6) is a single-line voice terminal that can be desk or wall
mounted. This set comes equipped with the following:
●
Handset
●
Touch-Tone
●
Built-In Speakerphone
●
●
Dial
Twelve Memory Buttons—Where emergency numbers and frequently called numbers
can be stored for quick calling.
Six Fixed Feature Buttons
— Program—For entering the memory button programming mode.
— Redial—For recalling the last number dialed.
— Flash—For generating a timed switchhook flash.
— Mute—For turning off the speakerphone microphone temporarily for privacy
— Hold (with status lamp)—For putting calls on hold.
— Speaker (with status lamp)—For making speakerphone calls and for turning
on the speaker during handset calls.
●
Tone
Ringer
●
Three Volume Controls
— Tone ringer
— Speaker
— Handset receiver.
Adjuncts: None
4-16
MEMORY
BUTTONS
PROGRAM BUTTON
REDIAL
BUTTON
FLASH BUTTON
HOLD BUTTON AND LAMP
SPEAKERPHONE BUTTON
AND LAMP
MUTE BUTTON
VOLUME
CONTROL
SPEAKER
VOLUME
CONTROL
RECEIVER
VOLUME
CONTROL
Figure 4-6. 420 Speakerphone Voice Terminal
4-17
TABLE 4-D. Model 2500 Series Voice Terminal Adjuncts
Adjunct
2500DMGC 2500MMGB 2500SM 2514 BMW 2554BM
x
4A Speakerphone
55A1 Ground
Start Key
x
x
x
x
x
AT&T Answer-Record
2500 or
Code-A-Phone 2540
(Answering Machine)
x
x
x
x
x
x
x
x
x
Z34A Message
Waiting Indicator
Single-Line Voice Terminal Connection Information
Single-line voice terminal connection information is provided in the following figures:
●
Figure 4-21—On-Premises Single-Line Voice Terminal Connections
●
Figure 4-22—Out-Of-Building Single-Line Voice Terminal Connections
●
Figure 4-23—Off-Premises Station Single-Line Voice Terminal Connections.
Descriptions of the Station Interconnect Panel (SIP), Trunk Access Equipment (TAE), and
associated cables and adapters, as shown on the figures, are provided under the heading
“connectivity” later in this Section.
Maximum cabling distances from the system cabinets to single-line voice terminals is
provided in Section 5, “Technical Specifications.”
Feature Operations
Refer to Single-Line Terminal User Guide (555-520-702) for information about feature
operation.
4-18
Multiline Voice Terminals
The system supports MET (10- and 12-button only) and MERLIN (7300H Series) multiline
voice terminals.
Multiline voice terminals have two LEDs located beside each assignable button (except for
the 34-button basic set). The LEDs are referred toas I-Use (red) and Status (green) LEDs.
Additional information on the LEDs is provided in the “Line Status and I-Use Indications”
feature description in Section 2.
Most multiline voice terminals support adjuncts.
following the voice terminal descriptions.
The supported adjuncts are described
Multilane voice terminals can have more than one call appearing at the terminal at onetime.
Each multiline terminal has two System Access buttons on which calls can be made or
received. System Access buttons are, essentially, inside line buttons.
Multilane voice terminals transmit voice signals in analog form and control signals in digital
form. The terminals operate over Premises Distribution System 4-pair wiring.
7300H Series Voice Terminals: These voice terminals are the same as those used
MERLIN Communications Systems. They connect to ports on the Analog Terminal
(ATL) Line (ZTN79) CP. The ATL Line CP converts the analog voice signals to digital
before placing them on the TDM bus. Three-wire pairs connect these sets to the port
see “Port Specifications “ in Section 5 for details.
with
Link
form
CPs;
The dial pad on 7300H series terminals is not a touch-tone pad in that the signals generated
are digital, not DTMF signals. A digital signal is sent to the switch with each button press.
Reference to the dial pad on these sets throughout this manual is in the form, “touch dial
pad.”
MET Sets: The MET Sets are not orderable as part of the system equipment but can be
used in System 25 installations, where appropriate (i.e., in reuse situations). Three-wire
pairs connect these sets to the port CPs; see “Port Specifications” in Section 5 for details.
The TN735 MET Line CP converts the analog voice signals to digital form before placing
them on the TDM bus. The touch-tone dial pad on MET voice terminals generates DTMF
signals.
The following subsections provide descriptions and illustrations of the multiline voice
terminals supported by System 25.
4-19
5-Button Voice Terminal (7302H01D) PEC 3160-111
The 5-Button Terminal (Figure 4-7) can be desk or wall mounted and is about 5-3/4 inches
wide, 5-1/4 inches high, and 8-1/2 inches long. The set comes equipped with the following:
●
Handset
●
Touch Dial Pad (not DTMF)
●
I-Use and Status LEDs
●
Test/Program Switch
— T—Used to test the operation of LEDs and ringer
— P—Used to program features
●
Built-in
Speaker
●
Speaker/Ringer Volume Control
●
Six Predefine Buttons
— Conference
— Drop
— Transfer
— Hold
— Speaker
— Recall
●
Five Feature Buttons (each equipped with I-Use and Status LEDs)
— Two predefine as System Access
— Three programmable (default assignments for V1 are all Repertory Dial; in
V2, one Repertory Dial button is replaced by Last Number Dialed).
Note: This set does not have a Message button or Message LED.
4-20
I-USE/STATUS LEDs
HANDSET
SYSTEM ACCESS
TEST/PROGRAM
SWITCH
PROGRAMMABLE
FEATURE
BUTTONS
SPEAKER/RING
VOLUME CONTROL
SWTICH
❑ CONFERENCE
❑ TRANSFER
❑ DROP
❑ HOLD
TOUCH
DIAL
PAD
RECALL
SPEAKER
Figure 4-7. 5-Button Voice Terminal (7302 HOlD)
4-21
10-Button Voice Terminal (7303H01D) PEC 3161-172
The 10-Button Terminal (Figure 4-8) can be desk or wall mounted and is about 7 inches wide,
5-1/4 inches high, and 8-1/2 inches long. The set comes equipped with the following:
●
Handset
●
Touch Dial Pad (not DTMF)
●
I-Use and Status LEDs
●
Test/Program
Switch
(two
positions)
— T–Used to test the operation of LEDs and ringer
— P–Used to program features
●
Built-in
Speaker
●
Speaker/Ringer
●
Seven
Volume
Predefined
Control
Buttons
— Conference
— Drop
— Transfer
— Hold
— Speaker
— Message
— Recall
●
Ten Feature Buttons (each with I-Use and Status LEDs)
— Two predefined as System Access
— Eight programmable (default assignments for V1 are Repertory Dial [3], Flex
DSS [3], Send All Calls [1], and Account Code Entry [l]; in V2, one Repertory
Dial button is replaced by Last Number Dialed).
Adjuncts:
●
502A Headset Adapter (PEC 3164-HFA)
●
S102A Speakerphone (PEC 3163-HFU).
4-22
I-USE/STATUS LEDs
SYSTEM ACCESS (2)
PROGRAMMABLE
FEATURE BUTTONS (3)
TEST/PROGRAM
SWITCH
PROGRAMMABLE
FEATURE
BUTTONS (5)
HANDSET
TOUCH DIAL PAD
❑ CONFERENCE
❑ TRANSFER
❑ DROP
❑ HOLD
SPEAKER/RING
VOLUME CONTROL
SWITCH
SPEAKER
RECALL
MESSAGE
Figure 4-8. 10-Button Voice Terminal (7303HO1D)
4-23
34-Button Voice Terminal (7305H01D) (MD) PEC 3162-412
The 34-Button Terminal (Figure 4-9) can be desk or wall mounted and is about 10-1/4 inches,
5-1/2 inches high, and 8-1/2 inches long. This set is available only on a reuse basis and is not
orderable via the Delivery Operation Support System (DOSS) Configurator. The set comes
equipped with the following:
●
Handset
●
Touch Dial Pad (not DTMF)
●
I-Use and Status LEDs
●
Test/Program Switch (two positions)
— T—Used to test the operation of LEDs and ringer
— P—Used to program features
●
Built-in Speaker
●
Speaker/Ringer Volume Control
●
Seven Predefined Buttons
–– Conference
—–
Drop
— Transfer
— Hold
— Speaker
— Message
— Recall
●
34 Feature Buttons (only ten with I-Use and Status LEDs)
— Two predefined as System Access
— 32 programmable (default assignments for V1 are Repertory Dial [3], Flex
DSS [27], Send All Calls [1], and Account Code Entry [1]; in V2, one
Repertory Dial button is replaced by Last Number Dialed).
Note: The programmable buttons without LEDs should not be programmed
for features that require I-Use and Status LEDs. Only the Repertory Dialing,
Manual Signaling, and Account Code Entry features should be assigned to
these buttons.
Adjuncts:
●
502A Headset Adapter (PEC 3164-HFA)
●
S102A Speakerphone (PEC3163-HFU).
4-24
I-USE/STATUS LEDs
HANDSET
SYSTEM ACCESS (2)
PROGRAMMABLE FEATURE
BUTTONS (3)
TEST/PROGRAM
SWITCH
PROGRAMMABLE
FEATURE
BUTTONS (29)
TOUCH
DIAL
PAD
SPEAKER/RING
VOLUME CONTROL
SWITCH
MESSAGE
SPEAKER
RECALL
❑ CONFERENCE
❑ TRANSFER
❑ DROP
❑ HOLD
Figure 4-9. 34-Button Voice Terminal (7305HO1D) (MD)
4-25
34-Button Deluxe Voice Terminal (7305H02D) PEC 3162-417
The 34-Button Deluxe terminal (Figure 4-10) is available for general use and as a Direct
Trunk Attendant Console. The voice terminal is about 10-1/4 inches wide, 5-l/2 inches high,
and 8-1/2 inches long. It comes equipped with the following:
●
Handset
●
Touch Dial Pad (not DTMF)
●
I-Use and Status LEDs
●
Test/Program Switch (two positions)
— T—Used to test the operation of LEDs and ringer
— P—Used to program features on the voice terminal
●
●
●
Built-in
Speaker
Speaker/Ringer Volume Control
Seven Predefined Buttons
— Conference
— Drop
— Transfer
— Hold
— Speaker
— Message
— Recall
●
34 Feature Buttons (each equipped with I-Use and Status LEDs)
For General Use:
— Two predefined as System Access
— 32 Programmable (default assignments for V1 are, Repertory Dial [3], Send
All Calls [1], Account Code Entry [1], and Flex DSS [27]; in V2, one
Repertory Dial button is replaced by Last Number Dialed)
For Use as a Direct Trunk Attendant Console:
— Two predefined as System Access
— Six other predefined (Start, Cancel, Release, Return-On-Don’t-Answer,
Return-On-Busy, and Alarm)
— 26 programmable (default assignments for V1 are Repertory Dial [3], Flex
DSS, Account Code Entry [1], Attendant Message Waiting [1], Night Service
[l], trunk appearances [15 as Personal Lines*], Group Call Coverage [1], and
Direct Facility Access† [3]; in V2, one Repertory Dial button is replaced by
Last Number Dialed)
* On the primary Attendant Console, the first 15 trunks in the system are assigned
button appearances on the console. If there are fewer than 15 trunks, the remaining
buttons are not assigned. On the secondary Console these trunks do not receive
4-26
default assignments.
† On the primary Attendant Console, the first of the Direct Facility (Pooled) Access
buttons defaults to loop-start trunks, the second to ground-start trunks, and the
third to tie trunks. For any trunk type not assigned in the system, the associated
button does not receive a default assignment. On the secondary Console, these
buttons do not receive default assignments.
Adjuncts:
●
502A
●
S102A
Headset
Adapter
Speakerphone
(PEC3164-HFA)
(PEC3163-HFU).
HANDSET
SYSTEM ACCESS (2)
PROGRAMMABLE FEATURE
BUTTONS (3)
I-USE/STATUS LEDs
TEST/PROGRAM
SWITCH
PROGRAMMABLE
FEATURE
BUTTONS (29)
TOUCH
DIAL
PAD
SPEAKER/RING
VOLUME CONTROL
SWITCH
MESSAGE
SPEAKER
RECALL
❑ CONFERENCE
❑ TRANSFER
❑ DROP
❑ HOLD
Figure 4-10. 34-Button Deluxe Voice Terminal (7305H02D)
4-27
Built-In Speakerphone (BIS) Voice Terminal (7305H03B) PEC 3162-BIS
The BIS voice terminal (Figure 4-10) can be desk or wall mounted and is about 9-1/2 inches
wide, 5-1/4 inches high, and 9-1/4 inches long. The set comes equipped with the following:
●
Handset
●
Touch Dial Pad (not DTMF)
●
I-Use and Status LEDs
●
Test/Program Switch (two positions)
— T—Used to test the operation of LEDs and ringer
P—Used to program features
●
Built-In Speakerphone
●
Speakerphone Volume Control
●
Built-In HFAI Microphone
●
Speaker/Ringer Volume Control
●
Nine Predefined Buttons
— Conference
— Drop
— Transfer
— Hold
— Speakerphone
— Recall
— Message
— Microphone
— HFAI
●
34 Feature Buttons (only 12 have associated LEDs)
— Two predefined as System Access
— 32 programmable (default assignments for V1 are Repertory Dial [3], Flex
DSS [27], Send All Calls [1], and Account Code Entry [1]; in V2, one
Repertory Dial button is replaced by Last Number Dialed)
Note: The programmable buttons without LEDs should not be programmed
for features that require I-Use and Status LEDs. Only the Repertory Dialing,
Manual Signaling, and Account Code Entry features should be assigned to
these buttons.
Adjunct:
●
502A Headset Adapter (PEC 3164-HFA).
4-28
I-USE/STATUS LEDs
HANDSET
SYSTEM ACCESS (2)
PROGRAMMABLE
FEATURE
BUTTONS (3)
PROGRAMMABLE
FEATURE
BUTTONS (29)
TEST/PROGRAM
SWITCH
TOUCH DIAL
PAD
SPEAKER/RING
VOLUME CONTROL
SWITCH
RECALL
SPEAKERPHONE
VOLUME CONTROL
HFAI
MESSAGE
SPEAKERPHONE
MICROPHONE
CONFERENCE
DROP
❑ TRANSFER ❑
HOLD ❑
❑
Figure 4-11. BIS Voice Terminal (7305H03B)
4-29
Built-In Speakerphone (BIS) Voice Terminal With Display (7305H04C) PEC 3162DIS (V2 only)
The BIS Voice Terminal with Display (Figure 4-12) is available for general use and as a
Switched Looped Attendant Console (SLAC). In general use, it can be desk or wall mounted;
as a console it is normally desk mounted. The terminal is about 9-1/2 inches wide, 5-1/4
inches high, and 9-1/4 inches long. The set comes equipped with the following:
●
Handset
●
Touch Dial Pad (not DTMF)
●
I-Use and Status LEDs
●
Test/Program
Switch
(two
positions)
— T—Used to test the operation of LEDs and ringer
— P—Inoperative on this terminal
●
Built-In Speakerphone
●
Built-In Display Module
— Screen for call information and time displays
— Clock/Timer controls
— Contrast control for screen display
●
Speakerphone
Volume
●
Built-In
●
Speaker/Ringer
●
Nine Predefined Buttons
HFAI
Control
Microphone
Volume
Control
— Conference
— Drop
— Transfer
— Hold
— Speakerphone
— Recall
— Message
— Microphone
— HFAI
●
34 Feature Buttons (only 12 have associated LEDs)
For General Use:
— Two predefined as System Access
— 32 programmable (default assignments are Repertory Dial [2], Last Number
Dialed [1] Flex DSS [27], Send All Calls [1], and Account Code Entry [1])
4-30
For Use as a Switched Loop Attendant Console:
— Five predefined as loop buttons
— Twelve other predefined (Inspect, Attendant Message Waiting, Alarm, Local,
Scroll, Forced Release, Start, Source, Release, Destination, Cancel, Join)
— Seventeen programmable (defaulted as Flex DSS [15], Position Busy [1], and
Last Number Dialed [1])
Note: The programmable buttons without LEDs should not be programmed
for features that require I-Use and Status LEDs. Only the Repertory Dialing,
Manual Signaling, and Account Code Entry features should be assigned to
these buttons.
Adjunct:
●
502A Headset Adapter (PEC 3164-HFA).
SYSTEM ACCESS (2)
PROGRAMMABLE
CLOCK
FEATURE
CONTROLS
BUTTONS (3)
I-USE/STATUS LEDs
PROGRAMMABLE
FEATURE
BUTTONS (29)
CONTRAST CONTROL
DISPLAY SCREEN
HANDSET
TEST/PROGRAM
SWITCH
TOUCH DIAL
PAD
SPEAKER/RING
VOLUME CONTROL
SWITCH
RECALL
SPEAKERPHONE
VOLUME CONTROL
MESSAGE
HFAI
SPEAKERPHONE
MICROPHONE
CONFERENCE ❑
TRANSFER ❑
DROP ❑
HOLD ❑
Figure 4-12. BIS Voice Terminal with Display (7305H04B)
4-31
Hands-Free-Answer On Intercom (HFAI) Voice Terminal (7309H01B) PEC
3161-161
The HFAI voice terminal (Figure 4-13) can be desk or wall mounted and is about 6-1/4
inches wide, 5-1/4 inches high, and 9 inches long. The set comes equipped with the
following:
●
Handset
●
Touch dial pad (not DTMF)
●
I-Use and Status LEDs
●
Test/Program
Switch
(two
positions)
— T—Used to test the operation of LEDs and ringer
— P—Used to program features on the voice terminal
●
Built-In HFAI Microphone
●
Built-In
●
Speaker/Ringer
●
Eight Predefine Buttons
Speaker
Volume
Control
— Conference
— Drop
— Transfer
— Hold
— Speaker
— Recall
— Message
— HFAI Microphone
●
Ten Programmable Buttons (each equipped with I-Use and Status LEDs)
— Two predefine as System Access
— Eight programmable (default assignment for V1 is Repertory Dial [3], Flex
DSS [3], Send All Calls [1], and Account Code Entry [1]; in V2, one Repertory
Dial button is replaced by Last Number Dialed).
Adjuncts: None
4-32
I-USE/STATUS LEDs
HANDSET
SYSTEM ACCESS (2)
PROGRAMMABLE
FEATURE BUTTONS (3)
TEST/PROGRAM
SWITCH
TOUCH DIAL
PAD
PROGRAMMABLE
FEATURE BUTTONS (5)
SPEAKER/RING
VOLUME CONTROL
SWITCH
CONFERENCE
DROP
MESSAGE
RECALL
❑
TRANSFER
HOLD
❑
❑
❑ HFAI MICROPHONE ❑
SPEAKER ❑
❑
Figure 4-13. HFAI Voice Terminal (7309HO1B)
4-33
❑
Ten Button Multibutton Electronic Telephone (MET) (2991 C/D05)
The Ten Button MET set (Figure 4-14) may be desk or wall mounted. This set is available
only on a reuse basis and is not orderable via the Delivery Operation Support System (DOSS)
Configurator. The set comes equipped with the following:
●
Handset
●
Touch-Tone Dial Pad (DTMF)
●
I-Use and Status LEDs
●
Tone Ringer Volume Control
●
Six
Predefined
Buttons
— Recall
— Conference
— Drop
— Transfer
— Hold
— Message
●
Five Feature Buttons (each equipped with I-Use and Status LEDs)
— Two predefined as System Access
— Three programmable (default assignments for V1 are Repertory Dial; in V2,
one Repertory Dial button is replaced by Last Number Dialed).
Adjuncts:
●
4A Speakerphone
●
MET
Headset
Adapter.
Note: The BLF version of this set, unless modified, will not work on System 25.
4-34
TOUCH TONE
DIAL
PAD
HANDSET
SYSTEM ACCESS
PROGRAMMABLE
FEATURE BUTTONS
MESSAGE WAITING
DROP
CONFERENCE
TRANSFER
HOLD
RECALL
BUTTON
INDICATOR
LAMPS
Figure 4-14. Ten Button MET Set (2991C05)
4-35
Ten Button MET With Built-In Speakerphone (2993C04)
The 10-Button MET set with BIS (Figure 4-15) can be desk or wall mounted. This set is
available only on a reuse basis and is not orderable via the Delivery Operation Support
Systcrn (DOSS) Configurator. The set comes equipped with the following:
●
Handset
●
Touch dial pad (DTMF)
●
I-Use and Status LEDs
●
Built-In Speakerphone
●
Tone Ringer Volume Control
●
Speakerphone Volume Control
●
On/Quiet and Off Speakerphone Control Buttons
●
Speakerphone Indicator Lamp
●
Six Predefined Buttons
— Recall Button
— Conference
— Drop
— Transfer
— Hold
— Message
●
Five Feature Buttons (each equipped with I-Use and Status LEDs)
— Two predefined as System Access
— Three programmable (default assignments for V1 are Repertory Dial; in V2,
one Repertory Dial button is replaced by Last Number Dialed).
Adjuncts: None
4-36
TOUCH TONE
DIAL
PAD
SYSTEM ACCESS
PROGRAMMABLE
FEATURE BUTTONS
HANDSET
ON/QUIET
BUTTON
ON LAMP
SPEAKERPHONE
OFF BUTTON
RECALL
BUTTON
INDICATOR
LAMPS
VOLUME
CONTROL
/
❑ MESSAGE WAITING
❑ DROP
❑ CONFERENCE
❑ TRANSFER
❑ HOLD
\
Figure 4-15. Ten Button MET With Built-In Speakerphone (2993C04)
4-37
Twelve Button MET Set (7203M)
The 12-Button MET set (Figure 4-16) is a freestanding voice terminal. This set is available
only on a reuse basis and is not orderable via the Delivery Operation Support System (DOSS)
Configuator. The set comes equipped with the following:
●
Handset
●
Touch-Tone Dial Pad (DTMF)
●
I-Use and Status LEDs
●
Tone Ringer Volume Control
●
Message Waiting LED
●
Seven Predefined Buttons
— Recall
— Conference
— Drop
— Transfer
— Hold
— Message
— Disconnect
●
Five Feature Buttons (each equipped with I-Use and Status LEDs)
— Three programmable (default assignments for V1 are Repertory Dial; in V2,
one Repertory Dial button is replaced by Last Number Dialed).
Adjuncts:
●
500A Headset Adapter
●
S101A
Speakerphone.
4-38
I-USE/STATUS
LEDS
❑ HOLD
❑ TRANSFER
❑ CONFERENCE
❑ DROP
❑ MESSAGE
❑ (PROGRAMMABLE)
❑ (PROGRAMMABLE)
❑ (PROGRAMMABLE)
❑ SYSTEM ACCESS
❑ SYSTEM ACCESS
HANDSET
TOUCH TONE
DIAL PAD
TONE RINGER
VOLUME
CONTROL
DISCONNECT
RECALL
Figure 4-16. Twelve Button MET Set (7203M)
4-39
Multiline Voice Terminal Connection Information
Detailed connection information is provided in the following figures:
●
Figure 4-24—On-Premises 7300H Series Multiline Voice Terminal Connections
●
Figure 4-25—Off-Premises 7300H Series Multiline Voice Terminal Connections
●
●
Figure 4-26—Ten Button MET Set Connections
Figure 4-27—Stand-Alone Remotely Powered Voice Terminal Connections
Descriptions of the Station Interconnect Panel (SIP), Trunk Access Equipment (TAE), and
associated cables and adapters, as shown on the figures, are provided under the heading
“Connectivity” later in this Section.
Maximum cabling distances from the system cabinets to multiline voice terminals is
provided in Section 5, “Technical Specifications.”
Feature Operations
Refer to Multiline Terminal User Guide (555-520-703) for information about feature
operation.
Voice Terminal Adjuncts
The following adjuncts and associated power supplies are supported:
●
MET Headset Adapter (for 10-Button MET Sets); refer to “Headset Adapter
Adjunct” in Section 2.
●
500A/502A Headset Adapter (for 12-Button MET Sets and MERLIN voice terminals,
respectively); refer to “Headset Adapter Adjunct” in Section 2.
●
4A Speakerphone System (for 10-Button MET Sets); refer to “Speakerphone Adjunct”
in Section 2.
●
S101A/S102A Speakerphone (for 12-Button MET Sets and MERLIN voice terminals,
respectively); refer to “Speakerphone Adjunct” in Section 2.
●
55A1 Key (Ground Start Button)
A 55A1 Ground Start Button is required to obtain dial tone for each Power Failure
Transfer (PFT) station that is connected to a ground start trunk.
●
Acoustic Coupler
An Acoustic Coupler (349A Adapter) can be used with the 7300H series voice
terminals and MET sets. The coupler provides acoustic coupling between the handset
and acoustic modems.
●
AT&T Answer-Record 2500
The Answer-Record 2500 (PEC 3121-050) or Code-A-Phone 2540 (PEC 3121-040) can
be used as an adjunct (using a 267A “T” Adapter) to single-line voice terminals.
Note, that when an answering device is bridged on to a ZTN78 Tip Ring Line CP port,
the combined adjunct/terminal REN must not exceed 1.20 A/B.
●
Z34A Message Waiting Indicator
The Z34A Message Waiting Indicator provides a message waiting indication at
single-line sets. The indicator can be used with 2500 series voice terminals that do
not have message waiting lamps.
4-40
Voice Terminal Adjunct Connection Information
Detailed adjunct connection information is provided in Section 2 with the detailed feature
descriptions of the headset adapter and speakerphone adjuncts.
Descriptions of the Station Interconnect Panel (SIP), Trunk Access Equipment (TAE), and
associated cables and adapters, as shown on the figures, are provided under the heading
“Connectivity’’ later in this Section.
Voice Terminal Adjunct Power Supplies
Table 4-E provides a summary of the supplemental power supplies and their applications.
TABLE 4-E. Supplemental Voice Terminal Power Supplies
UNIT
OUTPUT
FOR USE WITH
2012D Transformer
18 V ac
ADUs (except Z3A5) if
required;
MET auxiliary power.
KS22911-L1 Power
Unit
48 V dc
Selector Console;
Z3A5 ADUs;
MERLIN sets that require
local power.
85B1 Power Unit
18 V ac
4A Speakerphone System
Note: The S102A Speakerphone does not require supplemental power except when
connected to a 34-Button Deluxe Voice Terminal. The 502A Headset Adapter does not
require supplemental power except when connected to a 34-Button Deluxe, BIS, or BIS
with Display voice terminal.
Attendant Consoles
System 25 can have one of the following attendant console configurations:
●
One or two Direct Trunk Attendant Consoles (DTAC)–34-Button Deluxe Voice
Terminals (7305 H02B) administered for attendant service. Either or both positions
can have a Direct Extension Selector Console associated with it.
●
R1V2 Only: One or two Switched Loop Attendant Consoles (SLAC)–34-Button
BIS/Display Voice Terminals administered for attendant service. Either or both
positions can have a Selector Console.
Complete information on the Attendant Console feature can be found in Section 2.
4-41
Asynchronous Data Units (AD US)
Asynchronous Data Units, and Multiple Asynchronous Data Units (MADUs), provide an
interface between ports on the TN726 Data Line CP and RS-232C Data Terminal Equipment
(DTE) or Data Communications Equipment (DCE). The DTE is equipment that provides a
data source, termination, or both—a host computer, printer, or a data terminal are examples
of DTE. The DCE is equipment that provides the functions required to establish, maintain,
and terminate data communications—modems are the most common DCE.
The Z3A series of Asynchronous Data Units (ADUs) (Figure 4-17) are DCE that allow a
direct connection between DTE and port circuits on the Data Line CP (TN726). To connect
an ADU to DCE equipment, a cross-over cable (“null modem”) is required.
The modular jack labeled “Wall” connects the ADU to the building wiring with a standard
4-pair modular cord. The 400B2 Adapter is required to provide supplemental AC power and
can be bridged at the wall jack if required (Z3A1, 2, and 4 units only).
The modular jack labeled “Telephone” allows a voice terminal to be attached to the ADU.
Separate wire pairs from the telephone to the system cabinets are provided in a single 4-pair
cable run back to the SIP. The pairs separate at the SIP for connection to voice and data
ports.
The Z3A series of ADUs measure about 4.5 inches in length, 2 inches wide, and 1 inch high.
The ADUs available are shown in Table 4-F.
The Z3A ADUs should be installed only on lightning-protected facilities; they are not
designed to be used with CO cables or with exposed outside wiring (such as aerial cables).
The Z3A series of ADUs offer the following features:
●
Provide an interface to the digital switch from RS-232C device.
●
Increases the distance RS-232C signals can travel over standard twisted-pair wiring.
Refer to Section 5, “Technical Specification” for distance limitations.
●
Data and control signals can be transmitted 2,000 feet in asynchronous full-duplex
mode at speeds up to 19,200 bps. The transmission speed automatically matches that
of the attached RS-232C device.
●
The DC isolation via opto-couplers ensures high noise immunity, resulting in very
low error rates.
●
A variety of Z3AS with different connectors allows easy connection to RS-232C
terminals, printers, and host computers (see Table 4-F).
●
Most Z3AS can be powered from the RS-232C interface. The ADU requires 7 volts on
pin 20 (DTR) to operate properly. If the RS-232C equipment cannot meet this
requirement, a low-voltage power transformer and adapter(s) must be connected.
Z3A5 ADUs always require supplemental power.
●
An analog single-line voice terminal (2500 or 7100 series) or a 7300H series multiline
voice terminal (Z3A5 ADU required) can be connected to the ADU, allowing the voice
terminal and DTE to share a common wall jack and 4-pair cable run back to the SIP.
Note: Neither off-premises nor out-of-building service can be provided with ADUs.
For additional information on ADUs, see Z3A Asynchronous Data Unit User Manual (555401-701).
4-42
RS-232C
CONNECTOR
INTERFACE
TELEPHONE
JACK
WALL
JACK
OPTIONAL
ORIGINATE/DISCONNECT
SWITCH
Figure 4-17. Asynchronous Data Unit (ADU)
ADU Connection Information
Detailed connection information is provided in the following figures:
●
Figure 4-28—Typical ADU Connections Supporting Data Terminal And Single-Line
Voice Terminal
●
Figure 4-29—Typical ADU Connections Supporting Data Terminal And 7300H Series
Multiline Voice Terminal
●
Figure 4-30—Typical MADU Connection
●
Figure 4-31—Typical Z3A1/2/4 ADU Local Power Connections.
As shown in Figure 4-31, local external power can be provided to the Z3Al, Z3A2, and Z3A4
ADUs via a 2012D power transformer and a 248B Adapter attached to a 400B2 Adapter with
a D6AP-87 modular cord.
Descriptions of the Station Interconnect Panel (SIP), Trunk Access Equipment (TAE), and
associated cables and adapters, as shown on the figures, are provided under the heading
“Connectivity’’ later in this Section.
4-43
TABLE 4-F. Asynchronous Data Units
PEC
FEATURE
Z3A1
2169-001
3-foot plug-ended EIA connector and
mod jack for single-line set.
Z3A2
2169-002
EIA plug and mod jack for single-line
set.
Z3A4
2169-004
3-foot receptacle-ended EIA cord and
mod jack for single-line set.
Z3A5
62506
3-foot plug-ended EIA connector and
m o d j a c k f o r h y b r i d s e t . Requires
KS-22911-L1 power unit and D6AP
cord.
MADU
2169-005
Self-powered.
Used for host or
protocol converter connections where
voice terminals are not required. No
sets can be connected directly to the
MADU.
UNIT
4-44
Peripheral Equipment
Peripheral Equipment is equipment that connects to the Call Processor (ZTN82 or ZTN128)
CP, including:
●
System Administration Terminal (SAT); refer to “System Administration” in section
2.
●
Digital Tape Unit (DTU); refer to “Digital Tape Unit” in Section 2.
●
Station Message Detail Recording (SMDR) printer or Call Accounting System (CAS);
refer to “Station Message Detail Recording” in Section 2.
Auxiliary Equipment
Auxiliary equipment supports System 25 features and services. The following equipment is
supported:
●
Dictation Equipment; refer to “Dictation System Access” in Section 2.
●
External Alerting Equipment; refer to “External Alerts” in Section 2.
●
Music Source (Music-On-Hold); refer to “Music-On-Hold” in Section 2.
●
Paging Equipment; refer to “Paging System Access” in Section 2.
●
Recorded Delay Announcement Equipment; refer to “Direct Group Calling Delay
Announcement” and “Night Service-Delay Announcements” in Section 2.
●
Optional Power Equipment.
Notes:
1. Auxiliary equipment connected to the Ztn78 Tip Ring Line CP must meet the
following requirements:
— AC impedance: 600 ohms
— DC current less than 30 ma at 24V dc
– Ringer Equivalent Number (REN) less than 1.20
— Distance must not exceed 2000 feet
2. Off-premises auxiliary equipment must be connected to the TN742 Analog Line CP.
If the auxiliary equipment requires a contact closure, the TN763 Auxiliary Trunk
CP must be used (on-premises service only).
Optional Power Equipment
In addition to the power supplies already mentioned, the following equipment can be used
with System 25.
Uninterruptible Power Supply
The AT&T lKVA Uninterruptible Power Supply (UPS) Model 010U111 PEC 2403-004 is
recommended. At maximum load the UPS will bridge a 5 minute power outage. The UPS
must be connected to the common System 25 power outlet. One UPS will support a two
cabinet system.
4-45
AC Power Line Surge Suppressor
The TII Model 428 Self-Restoring Powerline Surge Suppressor (PEC 8310-001, Comcode
402988950) protects against electrical surges, spikes, and transients that can cause damage to
the System 25 power supply. A pilot light indicates that full protection is present. The unit
plugs directly into a standard 120-volt 15-amp grounded outlet, providing a dual outlet to
protected equipment.
346 Modular Bulk Power Supply
The 346 Modular Bulk Power Supply (346 MBPS) is a cost effective and flexible alternative
to the KS-22911 power supply. The 346 MBPS can be used where the wall outlet mounted
KS-22911 cannot be used (Canada) or where multiples units are required. The 346 MBPS
consists of the 346A Power Unit (346A PU) and the 346A1 Power Panel, that is the sole
method of mounting the power units. Up to three 346A PUS can be mounted per power
panel. Each PU is capable of powering four terminals with adjuncts, for a total of twelve
terminals per full MBPS. The 346 MBPS is intended to be installed in a closet and should be
near the SIP. All connections are modular and are made with cords and adapters at the
858A Adapter of the SIP. Terminals and the Selector Console can be powered by the 346
MBPS. Refer to Figure 4-27 for a typical use of the MBPS.
Connectivity
System 25 requires four-pair building wiring that conforms to AT&T Premises Distribution
System (PDS) specifications. Various cords, cables, adapters, and connecting blocks are used
to facilitate the connection of equipment and associated cable and wire.
Major points of connectivity include the following:
●
The system cross-connect field located on a wall adjacent to the system cabinets.
The field provides mounting space for the Trunk Access Equipment (TAE), Station
Interconnect Panels (SIPS), and Emergency Transfer Units (ETUs).
Refer to Section 5, “Environmental Requirements” for a typical System 25 layout
including cross-connect field and associated equipment layout.
●
25-pair connectors located on the rear of each system cabinet
●
Modular jacks located at each work station provide modular connections for
terminals and associated adjuncts and auxiliary equipment. These jacks are
connected by building wiring to the SIP. Several wiring options are described below.
Wiring Options: There are three basic PECs under which building (station) wiring is
ordered:
●
The 2772 (-JA1, JA2, or JC1) PECs cover new, reuse and (new) plenum wiring,
respectively. These PECs apply on a per wire-run basis.
●
PEC 2771-JDX covers wiring on a time and materials.
●
PEC 2773-JDX covers wiring based on a PCS quote.
Consult the 2770-section of the Sales Manual for restrictions and requirements before
ordering.
Trunk Access Equipment (TAE)
The TAE (Figure 4-18) provides for the connection of communications facilities such as Tie,
Ground Start, Loop Start, and DID trunks, to the trunk ports of the system. Up to three
4-46
trunk CPs (except Tie Trunk CPs) can be connected to a three-way splitter cable (OR6016)
that concentrates the CP interfaces into one 25-pair cable. Up to two Tie Trunk CPs can be
connected to a two-way splitter cable (OR6015) that concentrates the CP interfaces into one
25-pair cable. Each splitter cable connects to an interface block at the TAE.
Cables are cut down or plug into the TAE blocks and plug into the telephone company
provided network interface (RJ21X or RJ2GX). Trunks and tie lines are cut down by the
Telco at the interface.
700A or 157B Blocks are usually used for the TAE connections (furnished by the installer).
700A NETWORK
INTERFACE BLOCKS
(11O-TYPE OR
66-TYPE)
DIGITAL SWITCH
CO NETWORK INTERFACE
FACILITIES
(LOOP AND GRD. START TRUNKS,
OPS, DID TRUNKS)
CO TRUNK
FACILITIES
PORT CP(S)
RJ21X
CO TRUNK
FACILITIES
PORT CP(S)
RJ21X
CENTRAL
OFFICE
ANALOG
LINE CP
(TN-742)
TIE TRUNK
FACILITIES
PORT CP(S)
RJ2GX
TIE LINE
LEGEND:
A - SINGLE-ENDED 25-PAIR CONNECTOR CABLE (A25D)*
B - 3 TO 1 SPLITTER CONNECTORIZED CABLE (OR6016) - PEC 2720-06X
c - 2 TO 1 SPLITTER CONNECTORIZED CABLE (OR6015) - PEC 2720-05X
OPS - OFF-PREMISES STATION
SIP - STATION INTERCONNECT PANEL*
D - OCTOPUS CABLE (WP90780) - PEC 2720-05P
E - INSIDE WIRE*
* - FURNISHED BY INSTALLER
Figure 4-18. Trunk Access Equipment (TAE) Connections
4-47
Station Interconnect Panel (SIP)
The Station Interconnect Panel (SIP) provides for the connection of the terminals (voice and
data), peripheral equipment, and some auxiliary equipment of the system to station port
CPs. This equipment includes voice terminals, attendant consoles, data terminals, System
Administration Terminal, Digital Tape Unit, and Call Accounting System. The SIP is made
up of 617A Panels and associated adapters.
617A Panel
The 617A Panel (Figure 4-19) is a metal plate with key slot holes on each side for mounting
on a backboard. Each 617A Panel can hold eight Z21OA1 or 858A Adapters, each of which
can accommodate six connections to the port circuits in the cabinets. As many as five 617A
Panels can be required for a maximum size system. The adapters snap into prepunched
holes on the 617A Panels. (Reattached spacer buttons keep adapters from touching the
metal panels. )
The cable rings located at the top of the 617A Panel route the building wiring cables to the
adapters. Purse lock clips hold the building wiring cables in place. The white posts at the
bottom of the 617A Panel guide the wiring from the 50A Fanning Strip to each column of
adapters.
Preprinted boxes and numbers on the panel identify modular jacks for recordkeeping
purposes. Letters are marked on the boxes at the top of each column by the installer. The
letter (A-J) and the corresponding preprinted row number (l-24) identify the port jacks. For
example, Al identifies the modular jack located in column A row 1.
Adapters
Adapters that mount on the panel connect the following:
●
Building wire runs terminated in modular jacks, 25-pair connectors, or unterminated.
●
Cables from the system cabinets terminated in modular jacks or 25-pair connectors.
The following adapters can be mounted on the 617A panel:
●
Z210A—Six 4-pair modular jacks to six 4-pair modular jacks. One required per six
voice terminals. Connects to building wiring terminated in modular jacks.
(Equivalent to six Z600A Adapters.)
●
858A—Six 4-pair modular jacks to six 110-type cut-down blocks. One required per
six voice terminals. Connects to unterminated building wiring. (Equivalent to six
Z601As.)
The SIP equipment is furnished by the installer.
4-48
NOTE
BUILDING
WIRING RING
MODULAR
JACK ROW
NUMBER
COLUMN
LETTER
BOX
28 1/16”
PURSE
LOCK
CLIP
ADAPTER
MOUNTING
CLIP
OCTOPUS CABLE
DISTRIBUTION POST
NOTE 10” WIDE, 9 1/2” MOUNTING
CENTERS WHEN FLANGES
ARE OVERLAPPED
Figure 4-19. 617A Station Interconnect Panel
4-49
Figure 4-20 shows voice terminal connections to the system cabinets via the SIP. Typically,
voice terminals are plugged into modular wall jacks that provide a cut-down block for
building wiring. At the SIP, 858A Adapters provide a cut-down point for 4-pair wire runs.
An octopus cable (WP90780) from a station CP provides 25-pair connectorized cabling to
eight 4-pair modular jacks. Each jack is terminated on the SIP by a 858A Adapter. An
octopus cable connects a maximum of eight voice terminals to a port CP.
DIGITAL SWITCH
STATION
CIRCUIT
PACK
C2
— — — (1)
— — —
— — — (8)
PART OF SIP
UNTERMINATED
4-PAIR
BUILDING
WIRE
❘ ❘ ❘ ❘ ❘ ❘
❘ ❘ ❘ ❘ ❘ ❘
❘ ❘ ❘ ❘ ❘ ❘
❘ ❘ ❘ ❘ ❘ ❘
858A
ADAPT
❘ ❘ ❘ ❘ ❘ ❘
❘ ❘ ❘ ❘ ❘ ❘
❘ ❘ ❘ ❘ ❘ ❘
❘ ❘ ❘ ❘ ❘ ❘
❘ ❘ ❘ ❘ ❘ ❘
❘ ❘ ❘ ❘ ❘ ❘
858A
ADAPT
❘ ❘ ❘ ❘ ❘ ❘
❘ ❘ ❘ ❘ ❘ ❘
MODULAR
JACKS
VOICE TERMINALS,
ADJUNCTS,
POWER UNITS
LEGEND
:
C2 - OCTOPUS CABLE (WP90780) - PEC 2720-05P
858A ADAPTER (FURNISHED BY INSTALLER) SIX 4-PAIR MODULAR JACKS TO
SIX 11O-TYPE CUTDOWN BLOCKS
Figure 4-20. Typical SIP Connections
4-50
Connectivity Figures
Figures 4-21 through 4-31 provide connection information for various equipment. These
figures have been included as an aid to understanding how equipment can be connected to
System 25 and to indicate required connecting and supporting equipment. Other
arrangements are possible; these figures can be useful in developing connecting
arrangements for new or customer-provided equipment.
The PEC codes have been noted on the figures, as have indications of the source for
obtaining non-PEC equipment (eg, from installer or furnished with other equipment). This
information can be of use to Account Executives and Technical Consultants who are adding
equipment to existing installations. For new installations, the DOSS Configurator must be
used to select equipment requirements. For existing installations, you will need to determine
what equipment is already installed. Y OU should not order equipment directly using the
PECs in these figures. The octopus cable (PEC 2720-05P), for example, supports eight
terminals; you do not order one per terminal.
A list of related PECs, Apparatus, and Comcodes is provided in Section 7. Be sure to check
the Sales Manual and/or DOSS before ordering since this information changes frequently.
Symbols Used in Figures: Modular jacks are shown by the triangle symbol. The 25-pair
connectors are indicated by shaded blocks. Generally, only one leg of an octopus cable is
shown. Unterminated wiring requiring cut down or other termination do not have symbol
designations. The 103A Connecting Block is a typical modular wall jack that provides cutdown connections for building (station) wiring.
Voice Terminal and Adjunct Connections
Figures 4-21 through Figure 4-27 provide connection information for single-line and
multiline voice terminals. The single-line terminals can be located on-premises, offpremises, or out-of-building. The 7300H series multiline voice terminals can be used for outof-building service but must be within 2000 feet of the system cabinets (local power is
required beyond 1000 feet). Off-premises service is not available.
Diagrams for voice terminal adjunct connections are integrated with the specific feature
descriptions in Section 2.
Attendant Console Connections
Diagrams for attendant console connections are included in the “Attendant Console”
descriptions in Section 2.
Peripheral Equipment Connections
Diagrams for peripheral equipment connections are integrated with the specific feature
descriptions in Section 2.
ADU Connections
Figures 4-28 and 4-29 provide connection information for data terminals and associated
single-line or multiline voice terminals. The voice terminal and data terminal leads are
separated at the SIP with a Y-adapter and are connected to their respective station ports.
Figure 4-30 presents a typical Multiple Asynchronous Data Unit (MADU) connection.
Figure 4-31 shows local power connections for Z3A1, Z3A2, and Z3A4 ADUs.
Auxiliary Equipment Connections
Diagrams for auxiliary equipment connections are integrated with the specific feature
descriptions in Section 2.
4-51
SYSTEM 25
CABINET
ZTN78
OR
TN742
PART OF
OCTOPUS
CABLE
— — —
C2
●
—
—
—
—
—
—
—
—
—
PART OF
SIP
SIP
ADAPT.
W1
B1
C5
SINGLE-LINE
VOICE
TERMINAL
(NOTE)
LEGEND:
ZTN78 –
TN742 –
B1–
C2 C5 –
W1 –
TIP RING CP
ANALOG LINE CP
TYPICAL-103A CONNECTING BLOCK*
OCTOPUS CABLE (WP90780) - PEC 2720-05P
MODULAR CORD (D4BU-87) FURNISHED WITH SET
4 PAIR INSIDE WIRING CABLE*
NOTE : RANGE LESS THAN 2000 FEET FROM SYSTEM CABINET, USE ZTN-78 CP.
RANGE MORE THAN 2000 FEET BUT LESS THAN 1300 OHMS † (LOOP RESISTANCE)
FROM SYSTEM CABINET, USE TN742 CP. FIVE SINGLE-LINE VOICE
TERMINALS CAN BE BRIDGED WHEN USING A TN742 CP, HOWEVER, ONLY TWO
MAY BE OFF-HOOK AT ONE TIME.
* FURNISHED BY INSTALLER
† - INCLUDES TELEPHONE/TERMINAL
Figure 4-21. On-Premises Single-Line Voice Terminal Connections
4-52
SYSTEM 25
CABINET
TN742
ANALOG
LINE CP
PART OF
OCTOPUS
CABLE
— — —
— — —
C2
●
— — —
EXPOSED
CABLE
SIP
ADAPT.
W1
D1
D1
G
G
———
———
SINGLE-LINE
VOICE
TERMINAL
(NOTE)
C5
B1
LEGEND:
B1 – TYPICAL – 103A CONNECTING BLOCK
C2 - OCTOPUS CABLE (WP90780)
C 5 - MODULAR CORD (D4BU-87) - FURNISHED WITH SET
D1 - STANDARD GAS TUBE/FUSE PROTECTION PER BSP 46O-1OO-4OO*
G - APPROVED BUILDING GROUND
WI – INSIDE WIRING CABLE
NOTE:
MAXIMUM LOOP RESISTANCE FROM SYSTEM CABINET <1300 OHMS †
FIVE SINGLE-LINE VOICE TERMINALS CAN BE BRIDGED; ONLY
TWO MAY BE OFF-HOOK AT ONE TIME.
* – FURNISHED BY INSTALLER
INCLUDES VOICE TERMINAL
†–
Figure 4-22. Out-Of-Building Single-Line Voice Terminal Connections
4-53
SYSTEM 25
CABINET
TN742
ANALOG
LINE CP
PART OF
OCTOPUS
CABLE
— — —
— — —
C2
●
PART OF TAE
700A NETWORK
INTERFACE BLOCK
(110- OR 66-TYPE)
PART OF
SIP
SIP
ADAPT.
NETWORK
INTERFACE
W1
A
RJ21X
— — —
— — —
OPS
SINGLE-LINE
VOICE TERMINAL
CENTRAL
OFFICE
LEGEND:
A – SINGLE-ENDED 25 PAIR CABLE (A25D)*
C2 – OCTOPUS CABLE (WP90780) - PEC 2720-05P
WI -4 PAIR INSIDE WIRING CABLE*
* - FURNISHED BY INSTALLER
Figure 4-23. Off-Premises Station Single-Line Voice Terminal Connections
4-54
PART OF
OCTOPUS
CABLE
— — —
— — —
SYSTEM 25
CABINET
ZTN79
ATL
LINE CP
●
LEGEND
E1
C1
C2
T1
W1
C2
_
_
_
—
—
—
PART OF
SIP
SIP
ADAPT .
W1
B1
VOICE
TERMINAL
T1
:
– TYPICAL-103A CONNECTING BLOCK*
– MODULAR CORD (D8W-87) - FURNISHED WITH SET
– OCTOPUS CABLE (WP90780) - PEC 2720-05P
– 7300H TYPE VOICE TERMINAL
– 4 PAIR INSIDE WIRING CABLE*
* - FURNISHED BY INSTALLER
NOTE: RANGE WITHIN 2000 FEET OF SYSTEM CABINET (LOCAL POWER
REQUIRED >1000 FEET).
Figure 4-24. On-Premises 7300H Series Multiline Voice Terminal Connections
4-55
SYSTEM 25
CABINET
ZTN79
ATL
LINE CP
EXPOSED CABLE
PART OF
OCTOPUS CABLE
— — — —
—
—
—
—
I
PART OF SIP
SIP
ADAPT
C2
●
—
—
—
—
—
—
—
—
—
—
—
—
C1
IROB
PROTECTION
TII 341
IROB
PROTECTION
TII 341
(NOTE 1)
(NOTE 1)
W1
B1
C1
LEGEND :
B1
C1
C2
D
G
T1
WI
– TYPICAL-103A CONNECTING BLOCK*
–MODULAR CORD (D8W-87) - FURNISHED WITH SET
–OCTOPUS CABLE (WP90780) - PEC 2720-05P
– # 10 AWG GROUND WIRE*
– APPROVED BUILDING GROUND
– 7300H-TYPE VOICE TERMINAL
– 4 PAIR INSIDE WIRING CABLE*
VOICE
TERMINAL
T1
(NOTE 2)
NOTES :
1. IN-RANGE-OUT-OF-BLDG UNIT MODEL 341 IROB’S
PROTECTORS.
REQUIRED - PRIMARY/SECONDARY
2. RANGE: WITHIN 2000 FEET OF SYSTEM CABINET (LOCAL POWER REQUIRED >1000 FEET).
*
FURNISHED
BY
INSTALLER
Figure 4-25. Out-Of-Building
Connections
7300H
Series
4-56
Multiline
Voice
Terminal
SYSTEM 25
CABINET
TN735
MET
LINE CP
PART OF
OCTOPUS
CABLE
— — —
C2
●
—
—
—
—
—
—
—
—
—
PART OF SIP
SIP
ADAPT.
W1
B1
400B2
ADAPT
C1/C9
MET
SET
C7
248B
ADAPT
2012D
TRANS
LEGEND
:
B1 –
TYPICAL-103A
CONNECTING
BLOCK*
C1 – MODULAR CORD (D8W-87) - FURNISHED WITH SET
C2 - OCTOPUS CABLE CABLE (WP90780) - PEC 2720-05P
C9 – MODULAR CORD -(ZD8AJ-87) -COMES WITH DOSS ORDER
MET SETS – 2991C05, 2993C04 AND 2991D05 TEN BUTTON MET SETS,
USE ZD8AJ-87 ADAPTER CORD (C9) AND Cl
MET SETS – 7203M SET - 12-BUTTON MET SET - USES Cl ONLY
W1 – INSIDE WIRING CABLE*
248B ADAPTER – MODULARIZES 2012D TRANSFORMER
400B2 ADAPTER - POWER ADAPTER
PEC 21691
2012D TRANSFORMER – 15-18V AC TRANSFORMER
C7 -MODULAR CORD (D6AP-87)
1
NOTE: ONLY MET SET WITH BUILT-IN SPEAKERPHONE (2993C04) REQUIRES
TRANSFORMER AND ADAPTERS. OTHERWISE, PLUG C1 INTO B1 DIRECTLY.
* FURNISHED BY INSTALLER
Figure 4-26. Ten Button MET Set Connections
4-57
SYSTEM 25
CABINET
ZTN79
HYBRID
LINE CP
PART OF
OCTOPUS
CABLE
— — —
— — —
C2
PART OF
SIP
451A
●
— — —
— — —
ADAPT.
Z400F
ADAPT.
C1
SIP
ADAPT.
C7
W1
MBPS
P2
7300H
SERIES
VOICE
TERMINAL
LEGEND:
B1–
C1–
C 2C7 –
P 2–
TYPICAL- 103A CONNECTING BLOCK
MODULAR CORD (D8W-87)
OCTOPUS CABLE (WP90780)
MODULAR CORD (D6AP-87)
MODULAR BULK POWER SUPPLY
POWER UNIT (346A) - PEC 31760
POWER PANEL (346A-1) - PEC 31761
W1 – INSIDE WIRING CABLE
4-58
C1
B1
SYSTEM 25
CABINET
PART OF
OCTOPUS
CABLE
— — —
C2
ZTN78
OR
TN742
PART OF SIP
●
— — —
— — —
ADAPT
(WP90851-L1)
SIP
ADAPT
M1
Z3A1
ADU
B1
— — —
C2
TN726
●
— — —
RSS232
TERMINAL
— — —
LEGEND:
ZTN78
TN742
TN726
B1
C1
C2
C5
W1
WP90851-L1
Z3A1 ADU
-
TIP RING LINE CP
ANALOG LINE CP
DATA LINE CP
TYPICAL -103A CONNECTING BLOCK *
MODULAR CORD (D8W-87) - PEC 2725-07G
OCTOPUS CABLE (WP90780) - PEC 2720-05P
MODULAR CORD (D4BU-87) - FURNISHED WITH SET
INSIDE WIRING CABLE *
MODULAR Y ADAPTER - PEC 2750-T05
E/W 3 FT PLUG ENDED EIA CORD PEC 2169-001
C5
SINGLELINE SET
(NOTES)
NOTES :
1. IF RANGE IS GREATER THAN 2000 FT FROM SYSTEM CABINETTERMINAL DATA RATE (SPEED) WILL BE LIMITED
2. IF RANGE IS LESS THAN 2000 FEET FROM SYSTEM CABINET USE ZTN-78.
IF RANGE IS MORE THAN 2000 FEET BUT LESS THAN 1300 OHM †
(LOOP RESISTANCE) FROM CABINET USE TN-742
* FURNISHED BY INSTALLER
† INCLUDES
TELEPHONE/TERMINAL
Figure 4-28. Typical ADU Connections— Supporting Data Terminal And SingleLine Voice Terminal
4-59
SYSTEM 25
CABINET
ZTN79
ATL
LINE CP
TN726
DATA
LINE CP
PART OF
OCTOPUS
CABLE
— — —
C2
PART OF SIP
●
—
—
—
—
—
—
—
— —
C2
●
—
—
—
—
Y ADAPT
(WP90851-L1)
SIP
ADAPT.
W1
B1
Z3A5
ADU
—
—
-48V DC
P1
C1
LEGEND
:
WP90851-L1
E1
C1
C2
PI
WI
Z3A5 ADU
PEC
RSS232
TERMINAL
- MODULAR Y ADAPTER - PEC 2750-T05
- TYPICAL - 103A CONNECTING BLOCK *
- MODULAR CORD (D8W-87) - PEC 2725-07G
7300H
- OCTOPUS CABLE (WP90780) - PEC 2720-05P
SERIES
- KS22911 POWER SUPPLY- INCLUDED IN PEC 62506
VOICE
- 4-PAIR INSIDE WIRING CABLE
TERMINAL
- E/W 3 FT PLUG ENDED EIA CORD
62506 INCLUDES Z3A5 ADU, ONE D8W CORD (Cl), AND P1.
Figure 4-29. Typical ADU Connections—Supporting Data Terminal And 7300H
Series Multilane Voice Terminal
4-60
SYSTEM 25
CABINET
MADU
TN726
C10
W2
C11/C12
HOST
COMPUTER
OR
TERMINALS
LEGEND
:
TN726 – DATA LINE CP
MADU – MULTIPLE ASYNCHRONOUS DATA UNIT (PEC 2169-005)
SINGLE UNIT ASSEMBLY (8 ADU PORTS)
W2 – BUILDING WIRING (25-PAIR CABLE)
C1O – 25-PAIR CENTERFEED-TO-ENDFEED CABLE (PEC 2724-78B)
(ALWAYS REQUIRED)
Cll – M48C OCTOPUS CABLE (PEC 2724-29G) (7-FOOT CORD WITH
EIGHT 6-INCH ARMS FOR "DTE" HOST INTERFACE
C 1 2– M48G OCTOPUS CABLE (PEC 2724-98G) (7-FOOT CORD WITH
EIGHT 6-INCH ARMS FOR "DCE" HOST INTERFACE
Figure 4-30. Typical MADU Connections
4-61
TO RS-232C DEVICE
OPTIONAL SINGLE-LINE
AC POWER
OUTLET
2012D POWER
TRANSFORMER
248B ADAPTER
400B2 ADAPTER
D6AP-87 CORD
4-PAIR
D8W CORD
(PEC 2725-07G)
WALL JACK
D8AM-87 CROSSOVER CORD
(REQUIRED IF THIS ADU IS CONNECTED
TO ANOTHER ADU OR ANY OTHER DCE DEVICE, RATHER THAN DLC)
NOTE :
PEC 21691 includes 2012D transformer, 248B and 400B2
adapters and D6AP cord.
Figure 4-31. Z3A1/2/4 ADU Local Power Connections
4-62
SECTION 5—TECHNICAL SPECIFICATIONS
This section provides information on the technical characteristics and capacities of the
system. Some items covered here are discussed elsewhere in the manual but are repeated
here for ease of reference.
Technical specifications are provided for:
●
Hardware and Software Parameters
●
Unit Loads
●
Cable
●
Call Progress Tones
●
Indicator Lamp Signals
●
Port Specifications
●
Recommended Network Facilities (Trunks)
●
Analog Transmission Characteristics.
Distance
Limitations
5-1
Hardware And Software Parameters
The following is a listing of maximums for hardware and software parameters.
TOTAL
ITEM
Attendant Consoles
Direct Trunk Termination
Direct Extension Selector
2
2
Automatic Route Selection (ARS):
Patterns
Subpatterns Per Pattern
Routes Per Subpattern
Facility Restriction Levels
8
2
3
4
3
Cabinets
Circuit Packs:
Common Control
Circuit Pack Slots Per Cabinet
3
12
Call Coverage
Individual Covered Stations (VI )
Individual Covered Stations (V2)
Call Coverage Groups
Receivers per covered station or group
31
No limit
32
8
Call Pickup
16
16
Groups
Members per group
5
Conference Members
600
Dial Codes
Personal Dial Codes (PDCs)
Floating Personal Dial Codes (FPDCs)
Data Dial Codes (DDCs)
200
300
104
10
Virtual Facilities
5-2
Hardware And Software Parameters (Contd)
TOTAL
ITEM
Direct Group Calling (DGC)
32
10
Groups
Members per group
Emergency Transfer Units (ETUs)
Voice Terminals per ETU
4
5
Modem Pool Circuit Packs per cabinet
Conversion Resources per circuit pack
2
2
Paging Zones
3
Parked Calls (System )
24
1
8
Per Voice Terminal
Attendant DXS Console
16
Trunk Groups
System Delay Announcements:
Direct Group Calling Delay Announcement
Directed Night Service Delay Announcement
Account Code Digits
SMDR or Call Accounting System
(Models 100, 200, 300, or 500)
1
2
15
Speed Dialing Numbers
System Speed Dialing Numbers (#100-#189)
Personal Speed Dialing Numbers (#20-#26)
per station
Repertory Dialing plus Speed Dialing Digits
90
7
5000
System Administration Terminal
1
Toll Call Allowed Lists
Total Entries (all lists)
4
64
5-3
Hardware And Software Parameters (Contd)
TOTAL
ITEM
Traffic Data
● Simultaneous 2-Party Conversations
● Call Capacity
–CCS/Hour
–Busy Hour Call Capacity
4140
2500
Reliability
● Mean Time Between Outages (MTBO)
4 Years
Power Consumption
● Per Cabinet, Maximum
● Thermal Dissipation
115
500 Watts
1700 BTU/Hour
256
Total Ports - also includes trunk and station ports
[Software Limits; hardware maximum = 36 CPs]
System Resources
● One Service Circuit CP
(includes 4 TT Receivers)
(8 ports allocated/CP)
● TWO Touch-Tone Receiver CPs
(4 TT Receivers/CP)
(8 ports allocated/CP
● Pooled Modem
(Max 6 CPs, 4 ports allocated/CP
(two modems per CP)
● STARLAN Interface CP (V2 only)
(4 ports per CP)
Trunks
● Trunk Ports
–Tie Trunks
● Auxiliary
Trunk Ports
—Paging Access
—Dictation Access
8
16
24
4
104
32
3
8
Station Ports
● Data Ports
● Voice Ports
–Single-Line Voice Terminals (200)
–Multiline Voice Terminals (V2-111; V1-96)
● Attendant Consoles (2)
● Selector Consoles (2)
● 34-Button Sets (non-att) (V2-53; V1-38)
● Non 34-Button Sets (V2-109; V1-94)
5-4
240
104
200
Unit Loads
A cabinet can supply no more than 80 unit loads of 48 volt power (a unit load is defined as
44 mA). Unit loading is determined by the terminal connected to the port circuits. The
following table lists unit loads for various terminals.
UNIT LOADS (Note)
EQUIPMENT
2500 Voice Terminals
7101A Voice Terminal
CIRCUIT
PACK
UNIT LOAD
PER PORT
ZTN78
0.5
TN742
1.0
MET Sets
TN735
1.0
5-Btn. (7302H01D)
10-Btn. (7303 H01D)
34-Btn. (7305 H01D)
HFAI (7309 H01B)
ZTN79
1.0
1.0
1.0
1.0
BIS (7305 H03B)
BIS w/display (7305H04C)
ZTN79
ZTN79
1.50
2.00
34-Btn. Deluxe
(7305 H02D)
ZTN79
2.0
S102A Speakerphone
ZTN79
1.0
502A Headset Adapter
ZTN79
0.75
Asynchronous Data Units
TN726
0.0
Tie Trunks
TN760B
2.0
DID Trunks
TN753
0.5
Emergency Transfer Unit
ZTN82/128
Pooled Modem
TN758
2 per ETU
Zero Unit Loads but
maximum of two
TN758s/cabinet
allowed.
Locally Powered
DXS Selector Console
Note: Equipment not listed above (i.e., TN763, ZTN76, ZTN77) does not affect unit
loading. Any voice terminal/adjunct combination requiring more than 2 Unit Loads
must be locally powered. When a voice terminal is locally powered, it places no unit
load on the cabinet. Specifically, a 34-Button Deluxe voice terminal equipped for
speakerphone operation requires auxiliary power. In addition, any 34-Button Deluxe,
BIS, or BIS with Display voice terminal equipped for headset operation requires
auxiliary power.
5-5
Cable Distance Limitations
The following specifications provide allowable cabling distances for the following devices:
●
Single-Line Voice Terminals
●
Multiline Voice Terminals
●
Data Terminals (RS-232) Connected to Asynchronous Data Units (ADUs).
Single-Line Voice Terminals
SUPPORTING
CIRCUIT
PACK
24-GAUGE WIRE
(0.5106 mm)
FEET
ZTN78
METERS
2,000
610
TN742 *
24,000
7,320
TN742 †
15,000
4,500
* 2500-type voice terminals
† 7101-type voice terminals
IN BUILDING
SINGLE-LINE
VOICE
TERMINAL
EXTENDED STATION
OR
OUT-OF-BUILDING
OR
OFF-PREMISES
SINGLE-LINE
VOICE
TERMINAL
< 2000 FEET
ZTN78
TIP RING LINE
CP
> 2000 FEET *
TN742
ANALOG LINE +
CP
* UP TO 24,000 FEET. TIP/RING LOOP RESISTANCE FROM
SYSTEM CABINETS (INCLUDING VOICE TERMINAL) MUST
NOT EXCEED 1300 OHMS.
+ FIVE SINGLE-LINE VOICE TERMINALS
WHEN USING THE TN-742, HOWEVER, ONLY TWO MAY BE
OFF-HOOK AT ONE TIME.
Figure 5-1. Single-Line Voice Terminal Allowable Cable Distances
5-6
Multiline Voice Terminals
SUPPORTING
CIRCUIT
PACK
ZTN79
(7300H Series–
in-building or
out-of-building,
no off-premises)
TN735
(in-building
MET Sets only )
24-GAUGE WIRE
(0.5106mm)
FEET
METERS
2,000
(Note)
610
1,000
305
Note: Requires local power beyond 1,000 feet.
IN-BUILDING
7300H SERIES
VOICE TERMINALS
< 2000 FEET
ZTN79
ATL LINE CP
< 1000 FEET
OUT-OF-BUILDING
7300H SERIES
VOICE TERMINALS
341
IROB
341
IROB
ZTN79
ATL LINE CP
LEGEND: IROB - IN RANGE OUT-OF-BUILDING
(2 IROB PROTECTION DEVICES REQUIRED)
IN-BUILDING
2991C05, 2991D05,
2993C04, AND
7320M
VOICE TERMINALS
*
< 1000 FEET
TN735
MET LINE
CP
* MODEL 2993C04 (MET EQUIPPED WITH BIS) REQUIRES
A 2012D POWER UNIT (15 - 18V AC)
Figure 5-2. Multilane Voice Terminal Allowable Cable Distances
5-7
Data Terminals (RS-232) Connected To Asynchronous Data Units
DATA RATE
300 bps
● 1,200 bps
● 2,000 bps
● 4,800 bps
● 9600 bps
● 19,200 bps
●
24-GAUGE WIRE
(0.5106 mm)
FEET
METERS
40,000
20,000
12,000
7,000
5,000
2,000
12,200
6,096
3,657
2,133
1,524
610
DISTANCE
(SEE TABLE)
RS-232C
DEVICE
(ASYNCHRONOUS)
ADU
ADU
ZTN82 OR
ZTN128
CALL
PROCESSOR CP
OR
DISTANCE
(SEE TABLE)
RS-232C
DEVICE
(ASYNCHRONOUS)
TN726
DATA
LINE CP
ADU
Figure 5-3. Asynchronous Data Unit Allowable Cable Distances, In-Building Only
5-8
Tones
The following call progress tones are generated by the system:
FREQUENCY
PATTERN (In Milliseconds)
Busy Tone
480 Hz + 620 Hz
500 on, 500 off; repeated
Confirmation Tone
350 HZ + 440 HZ
100 on, 100 off, l00 on, l00 off,
l00 on followed by silence
Dial Tone
350 HZ + 440 HZ
Continuous
Reorder Tone
480 Hz + 620 Hz
250 on, 250 off; repeated
Ringback Tone
440 Hz + 480 Hz
1200 on, 4000 off; repeated
Call Waiting
440 Hz
200 on; not repeated
Recall Dial Tone
350 Hz + 440 Hz
100 on, 100 off, 100 on, 100 off,
100 on, 100 off, followed by
continuous tone
Preferred Route
(Queuing) Tone (ARS)
440 Hz
Five 50 ms tones, 50 ms apart,
not repeated
TONE
Indicator Lamp Signals
The following lamp signals are provided at multiline voice terminal line appearances:
MEANING
LAMP SIGNAL
PATTERN (In Milliseconds)
Dark
Off
Lighted
On
Flashing
500 on, 500 off; repeated
Ringing
Broken
Fluttering
50 on, 50 off; repeated,
(gated on/off every 500 ms)
Transfer/Conference
in progress
Wink
350 on, 50 off; repeated
Hold
Inactive
5-9
Port Specifications
The following tables provides interface specifications for System 25 line and trunk port
circuits: supported by System 25:
DATA TERMINAL PORTS (Note)
STATION TYPE
EIA RS-232 Device Via
ADU
CIRCUIT PACK
DATA TERMINAL SPECIFICATIONS
Data Line
(TN726)
RS-232 device must furnish signals on ADU
pins 2 (TD) and 20 (DTR) and ground on
either pin 1 or 7. The ADU furnishes
signals on pins 3 (RD) and 8 (CD). The CD
signal is also tied to pins 5 (CTS) and 6
(DSR). Some data terminals may require
auxiliary power when used with a Z13A1,
Z3A2, or Z3A4 ADU (the Z3A5 always
requires local power.) ADUs require 7 volts
on pin 20 (DTR) to operate properly. The
following table lists data terminals that
have been tested and are known to operate
properly without auxiliary power.
Note: Refer to Cable Distance Limitations for supported data rates.
5-10
Port Specifications (Contd)
DATA TERMINALS
(Do Not Require Local Power)
DATA
TERMINAL
REQUIRES Z3A1
or Z3A2 ADU
AT&T
4410
4415
51OA
X
X
X
ADMs
3A
31
X
X
ADDS
Viewpoint *
X
ConCept
HDS 108
REQUIRES
Z3A4 ADU
X
Datamedia
Elite 1521
X
Hazeltine
1510
X
Hewlett Packard
2621A
2623A
2640
2645
2645A
X
X
X
X
X
Teletype
BLIT/1 (68000 based)
5620 (MAC-80 based)
5420
X
X
X
* Requires Originate/Disconnect Switch.
5-11
Port Specifications (Contd)
VOICE TERMINAL PORTS
STATION TYPE
Tip and Ring
Single-Line Sets
(Analog)
Tip and Ring
Single-Line Sets
(Analog)
SPECIFICATIONS
CIRCUIT PACK
Tip Ring Line
(ZTN78)
Analog Line
(TN742)
5-12
●
l-Pair Interface (Tip and Ring)
●
Analog signals modulated over DC
loop
●
Loop Voltage: 24 V dc
●
Signaling: Dual Tone Multifrequency
(DTMF) only
●
REN (max.): 1.2
●
DC Current (max.): 35 mA
●
Loop Range (24 AWG ): 2,000 feet
●
In-building service only
●
I-Pair Interface (Tip and Ring)
●
Analog signals modulated over DC
loop
●
Loop Voltage: 48 V dc
●
Signaling: Dual Tone Multifrequency
(DTMF) only
●
REN (max.): 5.0
●
DC Current (max.): 40 mA
●
Loop Range (24 AWG): 24,000 feet
●
Supports Out-of-Building, Extended
(greater than 2000 feet), Off-Premises
and Bridged Station services
(maximum of five bridged stations,
and two off-hook simultaneously).
Port Specifications (Contd)
VOICE TERMINAL PORTS (Contd)
STATION TYPE
MET Sets
(Hybrid)
MERLIN® Sets
(7300H Series)
(Hybrid)
SPECIFICATIONS
CIRCUIT PACK
MET Line
(TN735)
ATL Line
(ZTN79)
5-13
●
3-Pair Interface
l-Voice pair
2-Control pairs
●
Analog Voice, Digital
Control/Signaling
●
Power: Phantom Power Over Data
Pairs
●
Bipolar Signaling With O V de Offset
●
1 MHz Nominal Signaling Rate
●
Loop Range: 1000 feet
(In-Building service only)
●
3-Pair Interface
l-Voice pair
l-Control pair
l-Power pair
●
Analog Voice, Digital
Control/Signaling
●
Bipolar non-return to zero line-coding
●
40 kHz Nominal signaling rate
●
Loop Range: 1000 feet
●
In-Building and In-Range Out-ofBuilding (IROB) services only
Port Specifications (Contd)
TRUNK PORTS
TRUNK TYPE
CIRCUIT PACK
Auxiliary Trunk
TN763
Direct Inward
Dialing (DID)
Trunk
Tie Trunk
TN753
TN760B
5-14
SPECIFICATIONS
●
Capacity: 4 Circuits
●
3-pair interface:
Voice (T,R),
Signaling (S, S1),
Status (SZ, SZ1)
●
Capacity: 8 Circuits
●
2-Wire (600 Ohm Fixed
Impedance) Transmission
●
Signaling: Wink Start,
Delay Dial, or Immediate
Dial.
Accepts Dial Pulse Signals
●
Incoming Service Only
●
Capacity: 4 Circuits
●
Supports Type I E&M,
Type I Compatible E&M or
Type V Simplex Signaling
●
4-Wire
●
3-Pair Interface
Transmit
Receive
Signaling (E&M)
Transmission
Port Specifications (Contd)
TRUNK PORTS
TRUNK TYPE
CIRCUIT PACK
Ground Start
Trunk
ZTN76
Loop Start
Trunk
ZTN77
5-15
SPECIFICATIONS
●
Capacity: 8 Circuits
●
2-Wire (600 Ohms or RC
Balance) Transmission
●
Network Signaling: Ground
Start
●
Two-way or Incoming only
Service
●
Capacity: 8 Circuits
●
2-Wire (600 Ohms or RC
Balance) Transmission
●
Network Signaling: Loop
Start
●
Two-way or Incoming only
Service
Recommended Central Office Trunk Facilities
The following table provides recommendations for CO trunks based on the number of voice
terminals in the system and the calling traffic.
VOICE
TERMINALS
CALLING TRAFFIC
LIGHT
MEDIUM
HEAVY
TRAFFIC
TRAFFIC
TRAFFIC
20
3/3
4/4
5/5
25
3/4
5/5
6/6
30
4/4
6/5
8/7
40
4/4
6/6
9/8
50
5/4
‘7/6
10/8
60
5/4
8/7
11/9
70
5/5
8/7
12/10
80
6/5
9/7
12/10
100
6/5
10/8
14/12
120
7/6
11/9
16/13
140
7/6
12/10
17/14
160
8/7
13/10
19/15
180
8/7
13/11
20/16
200
9/8
14/12
22/18
Notes:
1. For systems with both one-way and two-way
trunks, t h e f i r s t n u m b e r l i s t e d u n d e r “Calling
Traffic” is the number of two-way trunks required,
the second number is the number of one-way trunks
required.
2. For systems with just two-way trunks, add the two
numbers listed under Calling Traffic to determine the
number of trunks required.
5-16
Analog Transmission Characteristics
Frequency Response:
(Station-To-Station or Station-To-CO-Trunk, relative to loss at 1 kHz)
FREQUENCY
60 Hz
200 Hz
300-3000 Hz
3200 Hz
3400 Hz
LOSS
>20 dB
<5 dB
<1 dB
<1.5 dB
<3 dB
Insertion Loss:
CONNECTION TYPE
Standard Station to Standard Station
Standard Station to Extended/Off-Premises Station
Extended/Off-Prem Station to Extended/Off-Prem Station
Station-to-Trunk
Trunk-to-Trunk
Overload Level:
+3 dBm0
Crosstalk:
< -70 dB
Intermodulation
Distortion:
FOUR TONE METHOD
>45 dB
2nd Order
Tone Products
>53 dB
3rd Order
Tone Products
Quantization Distortion:
SIGNAL LEVEL
+2 to -30 dBm0
-40 dBm0
-45 dBm0
DISTORTION LEVEL
35 dB
29 dB
25 dB
5-17
LOSS
6 dB
3 dB
O dB
O dB
O dB
Analog Transmission Characteristics (Contd)
Sampling Rate:
8 kHz
Terminating Impedance:
600 ohms
Trunk Balance Impedance:
600 ohms or Complex Z (selectable)
Echo Return Loss:
The echo return loss of the switching equipment is infinite. The echo return loss of the
station equipment can be engineered for greater than 18 db over the range of 500 Hz to
2500 HZ.
Loop Resistance:
●
TN74.2—Loop resistance of up to 1300 ohms, including the station
●
ZTN78—Loop resistance of up to 100 ohms not including the station
(2000 feet with No. 24 AWG.)
Connection Bandwidth: 64 Kbits
Steady State Noise Level:
The steady state noise level presented to any busy path does not exceed 23 dBrnc during
the busy hour.
Impulse Noise:
The impulse noise is 0 count (hits) in five minutes at +55 dBrnc during the busy hour.
Single Frequency Return Loss (Talking State):
Station to station—exceeds 12 db
Station to 4 wire trunk connection—exceeds 14 db
Station to 2 wire trunk connection—exceeds 12 db
Peak Noise Level:
Analog to analog—20 dBrnc
Analog to digital—19 dBrnc
Digital to analog—13 dBrnc
5-18
SECTION 6—ENVIRONMENTAL REQUIREMENTS
This section provides information on floor and wall space requirements for System 25
Also included are specifications for
cabinets and associated peripheral equipment.
temperature, humidity, air purity, lighting, electrical noise (RFI) suppression, power,
grounding and lightning protection.
Floor Plans And Layouts
Floor plan arrangements will vary depending on the available equipment area and
anticipated system growrth. A typical floor plan is shown in Figure 6-1.
The floor must be tiled or suitably sealed, level, and free from vibration. Allow for a
minimum unobstructed clearance of seven feet above the floor throughout the equipment
area.
Do not locate the equipment in areas:
●
Where it might be subjected to excessive vibrations or disturbed by moving equipment
such as hand trucks and transporters.
●
Where noise levels may exceed 90 dB.
●
Susceptible to flooding.
Maintain clear access to the equipment area for both installation and maintenance purposes.
The wall behind the system cabinet must be clear of all objects (pictures, shelves, or
windows) that might interfere with system installation. The entire area behind the cabinet
and to the side as shown on Figure 6-2 must be reserved for the cross-connect field and cable
access. Also, room for system .growth should be considered.
6-1
BACKBOARD
TERMINATION
FIELD
(NOTE 4)
SYSTEM
CABINETS
(FOOTPRINT)
TABLE
FRONT
NOTE 3
NOTES:
1. 115V AC, 60 Hz, 15 AMP OUTLETS
(HUBBELL 5262 OR EQUIVALENT)
MUST BE LOCATED WITHIN SIX FEET
(1.8 m) OF SYSTEM CABINETS.
2.
MULTIPLE CABINET SYSTEMS
TWO QUAD OUTLETS, SINGLE
SYSTEMS REQUIRE ONE QUAD
REQUIRE
CABINET
OUTLET.
3. ALLOW AT LEAST 24 INCHES OF SPACE
IN FRONT OF CABINETS. TABLE MUST
BE ABLE TO SUPPORT 250 POUNDS.
4.
BACKBOARD IS 3/4 INCHES THICK BY
48 INCHES WIDE BY 72 INCHES LONG
Figure 6-1. Typical System 25 Equipment Area Floor Plan
6 - 2
QUAD AC OUTLETS
Figure 6-2. Typical System 25 Equipment Area Elevation Plan
6-3
Table Top Space
The following system equipment requires (customer provided) table top space in the
equipment area:
●
System Cabinets - Each cabinet is 13 inches high, 17 inches wide, and 21 inches deep.
A three cabinet system requires a vertical space of approximately 40 inches and a 17
inch by 21 inch table top space. Each cabinet weighs approximately 75 pounds. Place
the cabinets on a desk or table-top that is about 18 inches high and capable of
supporting at least 250 pounds. The cabinets rnust not be placed on the floor.
●
System Administration Terminal (SAT) Model 703 - The SAT should also be located
near the system cabinets and plugged into the same AC outlet. It is 12 inches wide, 10
inches long, and 3 inches high.
●
Digital Tape Unit - The Tape Unit (Model DC5 Digital Data Recorder) should also be
located near the system cabinets. It is 5 inches wide, 2 inches high, and lo inches long.
●
SMDR or Call Accounting System (CAS) - T h e A T & T M o d e l 4 7 5 p r i n t e r i s
approximately 16 inches wide, 12 inches long, and 6 inches high. The CAS runs on the
AT&T PC 6300. They should also be located near the system cabinets.
Wall Space Requirements
The customer provided backboard for the cross-connect field shall be 3/4 inch thick, 4 feet
high, and 8 feet wide. Mount the board 30 inches above the floor. The board must conform
to national and local fire safety codes.
If existing cross-connect hardware is reused, the space requirements and hardware
requirements must be shown on the floor plan. Contact your AT&T Technical Consultant
for assistance in planning for reuse of existing equipment.
Temperature and Humidity
The System 25 equipment should be installed in a well-ventilated area. The equipment m u s t
be located in an area with an ambient temperature between 40 degrees and 104 degrees
Fahrenheit (5 and 40 degrees Celsius). The relative humidity must be less than 95%,
noncondensing. These parameters shall be maintained 24 hours a day, seven days a week.
6-4
Air Purity
The cabinet should not be installed in an area where the air may be contaminated with any
of the following:
●
Excessive dust, lint, carbon particles, paper fiber contaminants, or metallic
contaminants
●
Contaminants expelled by office copying machines
●
Highly corrosive atmosphere within an enclosed area or atmosphere containing
vaporized chemical compounds that may condense on the equipment
●
Explosive or flammable atmosphere
Lighting
I.ighting should be adequate to allow administration and maintenance personnel to perform
their tasks. The recommended light intensity level is 50 to 70 footcandles. This level
complies with the Occupational Safety and Health Act (OSHA) standards.
Electrical Noise (RFI)
In most cases, electrical noise is introduced to the system through trunk or voice terminal
cables. However, electromagnetic fields near the system cabinets may also induce noise in
the system. Therefore, the system cabinets 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 volt per meter are unlikely to cause interference. Field strength
can be measured by a tunable meter such as the Model R-70 meter manufactured by
Electro-Metrics Division or broadband meters such as the HOLADAY* HI-3001 meter or
Model EFS-1 meter manufactured by Instruments for Industry, Inc.
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 KHz).
* Trademark of Holaday Industries
6-5
AC Power Requirements
●
The System 25 power service shall
equipment served (See Figure 6-3. )
be
a
dedicated
branch
circuit
with
no
other
●
Each cabinet requires 500 Watts at 115V ac (maximum)
●
Provide a load center of appropriate current rating (ITE EQ4 typical) equipped with
120V ac, 15 ampere (AMP), single pole magnetic circuit breaker(s) (ITE QP1-B015
typical ). Each breaker is to protect 2 associated wall mounted 115 V ac, 15 AMP,
receptacles (HUBBELL 5262 typical). Grounding of this load center is to be provided
by a “Green Wire” ground extended from the grounding electrode conductor at the AC
service entrance to the load center.
The following materials are required:
A.
Single Cabinet System
1
1
1
1
2
B.
-
15 AMP 3 Wire Dedicated Branch Service
4“ Box (RACO 230 or Equiv.)
4“ Cover (RACO 807 or Equiv.)
Ground Bar (Square D PK9GTA or Equiv.)
Recpt. (Hubbell 5262 15 AMP or Equiv.)
Multiple Cabinet System
2
2
2
1
4
-
15 AMP 3 Wire Dedicated Branch Service
4“ Box (RACO 230 or Equiv.)
4“ Cover (RACO 807 or Equiv.)
Ground Bar (Square D PK9GTA or Equiv.)
Recpt. (Hubbell 526215 AMP or Equiv.)
C.
Typically, multiple cabinet systems can be powered from a single phase 120V
ac, 60 Hertz service (two 15 amp circuits required. ) There are no phase
restrictions between cabinets. Therefore the two 15 amp circuits required
may be derived from single or three-phase service.
D.
The receptacles shall be located at least 1 foot above the floor. Receptacles
shall not be located further than 4 feet from the cabinets.
DANGER
Under no circumstances should this equipment
be connected to 220V ac; doing so poses a
serious fire hazard.
6-6
TWO SEPARATELY FUSEO
15 AMP CIRCUITS
GROUND (GREEN)
ITE:
QP1 - B015
TWO 15 AMP CKT
BAKA OR
APPROVED
EQUIVALENT
LOAD CENTER
4“ BOX (RACO 230
OR EQUIVALENT)
HUBBELL RECPTS.
(5262 15 AMP
OR EQUIVALENT)
NEUTRAL
INSULATED FROM
LOAD ENTER
4“ COVER (RACO 807
OR EQUIVALENT)
SINGLE POINT
GROUND
GROUND BAR MOUNTED
ON 4“ BOX (SQUARE D
PK9GTA OR APPROVED
EQUIVALENT)
\
/
TO 240 VAC 30 AMP.
SINGLE PHASE MAIN
Figure 6-3. AC Power Distribution - Multiple Cabinet System
6-7
APPROVED
BUILDING
GROUND
(#6 AWG)
Grounding
Connection of an approved ground to the system cabinets is essential. An approved ground
may consist of any of the following:
●
Grounded Building Steel - The metal frame of the building.
●
Water Pipe - A continuous metal water pipe, not less than l/2 inch in diameter, that is
connected to an underground metal water pipe that is in direct contact with earth for
10 feet or more.
●
Concrete-Encased Ground - An electrode encased by at least 2 inches of concrete and
located within and near the bottom of a concrete foundation or footing indirect contact
with the earth. The foundation must consist of at least 20 feet of one or more steel
reinforcing bars or rods of not less than 1/2 inch in diameter, or at least 20 feet of
bare, solid copper wire not smaller than No. 4 gauge.
●
Ground Ring -A ring that encircles a building or structure indirect contact with earth
at a depth of at least 2-1/2 feet. The ring must consist of at least 20 feet of bare
copper conductor not smaller than No. 2 gauge.
Lightning Protection
A Coupled Bonding Conductor (CBC) tie-wrapped to all trunks provides lightning protection.
The CBC can be any one of the following:
●
A 16 gauge ground wire
●
Continuous
●
Six unused pairs of wire
cable
sheath
The CBC should be run from the telephone company provided network interface to the
system Cabinet 1’s ground block. If the telephone company has not extended the CBC from
the facility entrance to the network interface, the System 25 installer should run the CBC
along the same route as the incoming facilities, where feasible.
6-8
SECTION 7—PARTS INFORMATION
This section contains information that may assist you in cross referencing Apparatus Codes,
Comcodes and Price Element Codes (PECs). The first table is grouped by descriptive Price
Element Codes (PEC). The second table is arranged by component codes (COMCODE).
Parts Listed by PEC
PEC
DESCRIPTION
APPARATUS CODE
6250-011
System 25 Control Unit (V1)
Carrier
includes
J58901Al L1 (e/w)
Part of:
6250-011
6250-011
6250-011
6250-011
6250-011
6250-011
6250-011
6250-011
6250-011
6250-011
6250-012
Part of:
6250-012
6250-012
6250-012
6250-012
6250-012
6250-012
6250-012
6250-012
6250-012
62501
Part of:
62501
62501
62501
62501
62501
WP90510
WP90677 L1
21985-1
------J58901A1 L2,4
ZTN81
ZTN81B
ZTN81C
ZTN82
ZTN85
Power Supply
Fan (2)
Air Filter
Address Plug
TDM Bus Term
Memory CP
Memory CP
Memory CP
Processor CP
Service Circuit CP
System 25 Control Unit (V2)
Carrier
System 25 Expansion Unit
Carrier
7-1
403954761
845416379
403957129
845416635
103810586
105212179
105211023
103965323
845875155
includes
J58901A1 L1 (e/w)
WP90510
WP90677 L1
21985-1
-------J58901A1 L3
Power Supply
Fans (2)
Air Filter
Address Plug
TDM Bus Cable
403954761
845416379
403957129
845416635
103810586
103965281
103982740
105291488
103965299
103965323
includes
J58901A1 L1 (e/w)
WP90510
WP90677 L1
21985-1
------J58901A1 L2,4
ZTN127
ZTN128
ZTN85
Power Supply
Fan (2)
Air Filter
Address Plug
TDM Bus Term
Memory CP
Processor CP
Service Circuit CP
SLAC Grap. Layer (2)
COMCODE
403954761
845416379
403957129
845416635
403961519
PEC
DESCRIPTION
APPARATUS CODE
COMCODE
62502
GS Trunk CP
ZTN76
103965232
62503
LS Trunk CP
ZTN77
103965240
62504
TR Line CP
ZTN78
103965257
62505
ATL Line CP
ZTN79
103965265
62506
Asynchronous
Data Unit
D181521 includes
105105506
Part of:
62506
62506
D8W-87 Cord (14 ft)
Z3A5 ATL ADU
103786802
103975349
62507
Digital Tape Unit
TS-555A
404079436
62508
System Admin
Terminal (SAT)
TI-703
404079428
62509
Direct Extension
Selector Console
23A1-003 e/w
103969424
Part of:
62509
62509
62509
62509
62510
D6AP Cord
D8W-87 Cord (14 ft)
KS22911 L1 Power
400B Adapter
D181522 includes
MERLIN VT
local power
Part of:
62510
62510
62510
62511
Z400F Adapter
D6AP-87 Cord
KS22911 L1 Power
D181523 includes
Auxiliary Trunk
Interface
Part of:
62511
62511
62512
Block Conn 66E3-25
Cable B25A 15/DE
D181524 includes
Paging/Dictation
(Aux Trunk Interface)
Part of:
62512
62512
62512
62512
102937620
103786802
403242639
103848859
105105514
103942857
102937620
403242639
105105522
100009968
100017334
105105530
103871844
278A Adapter
KS22911 Ll Pwr Unit 403242639
102999059
D4BY Cord (14 ft)
105031181
D181321 Kit (zener)
7-2
PEC
DESCRIPTION
APPARATUS CODE
62513
Part of:
62513
62513
62513
MOH Interface
D181575 includes
COMCODE
*
KS23395 Interface
2012D Transformer
36A Voice Coupler
105143186
102600517
103558916
62514
Part of:
62514
62514
10B ETU
62515
Peripherals Interface
for Remote Access
Emergency Trans Unit
(2) B25A Cables DE
D8W-87 Cord (7 ft)
Part of:
62515
62515
62515
62515
62515
62515
62515
62515
62515
62515
103984118
100017334
103786786
248B Adapter
355AF Adapter
248B Adapter
400B2 Adapter
D6AP-87 Cord
D8AM-87 Cord
D8W-87 Cord (7 ft)
M7U-87 Cord
Z3A4 ADU
2012D Transformer
102802113
105012645
102802113
104152558
102937620
104154430
103786786
104466616
103964185
102600517
62518
STARLAN Interface CP
ZTN84
103965315
62519
Part of:
62519
62519
62519
62519
RlV2 Upgrade Kit
D181782 includes
105335657
Memory CP
Processor CP
SLAC Grap. Layer (2)
V2 Documentation
ZTN127
ZTN128
105212179
105211023
845875155
62520
Part of:
62520
62520
62520
STARLAN/ATI. Interface
D181807 includes
105355374
Y-Adapter
Adapter
Power Unit
WP90851,L1
KS-23475
KS-22911,L1
405010620
405462904
403242639
63111†
Analog Line CP
TN742
103556957
63112†
MET Line CP
TN735
103556882
63116†
DID Trunk CP
TN753
103557062
63117†
Tie Trunk CP
TN760B
103975645
7-3
PEC
DESCRIPTION
APPARATUS CODE
COMCODE
63118†
Aux Trunk CP
TN763
103557161
63119†
Pooled Modem CP
TN758
103557112
63123†
Tone Detector CP
TN748
103976163
63130†
Data Line CP
TN726
103556791
1020-S90
ACCESS Software
*
105341218
1020-S91
CAM Software
*
105341382
1203-020
AA Software
*
105339584
2169-001
Tip Ring ADU
Z3A1
103963963
2169-004
Tip Ring ADU
Z3A4
103964185
21691
Part of:
21691
21691
21691
ADU Aux Power
248B
400B
D6AP-87
102802103
103848859
102937620
2301-SAA
Supplemental Alert
Adapter
*
*
2610-001
STARLAN NETWORK
Network Extension Unit (NEU)
527840003
2614-100
STARLAN NETWORK
Network Access Unit (NAU)
527840102
2720-05P
25-pair/8-plug 15-ft CBL
WP90780L1
405010612
2720-05X
Splitter CBL, Tie-Trunk
WP90929,L3
403864150
2720-06X
Splitter CBL, CO Trunk
WP90929,L1
403836620
2724-30C
RS232 X-Over Cable
M7U-87
104246616
2724-38X
Mod(ADU) X-Over Cord
D8AM-87
104154430
2725-07G
Mod Cord
D8W-87 Cord (7 ft)
103786786
2725-075
Cord (25 ft)
D8W-87
103786828
2750-A17
MET Adapter Cord
ZD8AJ
103881421
7-4
PEC
DESCRIPTION
APPARATUS CODE
COMCODE
2750-A24
Mod/RS232 Adapt, Male
355A
105012637
2750-A25
Mod/RS232 Adapt, Female
355AF
105012645
2750-T05
Voice/Data (Y) Adapter
WP90851L1
405010620
2781-JDX
System Wiring-Time
and Materials
*
*
2782-JA1
(New) System Wiring
Run, Firm Price
*
*
2782-JA2
(Reuse) System Wiring
Run, Firm Price
*
*
2782-JC1
(New) System Wiring
Run, Firm Price
(Plenum Wiring)
*
*
System Wiring-Based
On a PCS Quote
*
*
2783-JDX
3100-1TD
Basic TT Desk Tel
2500MMGB
*
3100-TWR
Basic TT Wall Tel
2554BM
103234472
3178-SYS
Basic TT Desk Tel
w/Message and Recall
2500DMGC
103966255
*
Basic TT Desk Tel with
Headset Jack
2514 BMW
3100-2TD
Basic TT Desk
(4A Spkphone Compatible)
2500SM
*
3100-ORD
Basic Rtry Desk Tel
500MM
103870267
3100-ORW
Basic Rtry Wall Tel
500BMPA
103823555
3100-2RD
Basic Rtry Desk Tel
(4A Spkphone Compatible)
500SM
103870416
*
Multifeature TT Desk Tel
w/ Speaker, Memory & Redial
CS6402U01A
*
1121-050
AT&T Answer-Record
2500
1140-010
10-Button MET VT
2991C05
7-5
*
*
103871018
PEC
DESCRIPTION
APPARATUS CODE
COMCODE
3141-BIS
Part of:
3141-BIS
3141-BIS
MET Set-BIS
2993C04
103942146
Transformer
Kit of Parts
2012D
102600517
*
3143-12M
12-Button MET VT
Z7203M01A-003
103963310
3160-111
(MAC30 Att)
5-Button MERLIN VT
Z7302H01D-003
*
3161-172
(MAC30 Att)
10-Button MERLIN VT
Z7303H01D-003
*
3161-161
MERLIN HFAI VT
Z7309H01B-003
103982005
3162-412
34-Button MERLIN VT
Z7305H01D-003
*
3162-417
(MAC30 Att)
34-But Dlx MERLIN VT
Z7305H02D-003
*
3162-BIS
MERLIN BIS VT
Z7305H03B-003
103981965
3162-DIS
MERLIN VT
With Display
Z7305H04C-003
103981981
3163-HFU
MERLIN Speaker Mod
S102A
103814356
3164-HFA
MERLIN Headset
Adapter
502A
*
31710
Acoustic Coupler
349A Adapter
104010061
3170-00M
Single-Line VT w/Message
Light & Recall Button
7101A01A-003
103871109
31017
Ground Start Key
55A1
100287085
31019
External Alert
*
*
:310:32
Message Adjunct
Z34A
*
31760
31761
Mod. Bulk Pwr. Supply
Power Unit
Power Panel
346A
346A-1
104174768
104174750
32918
IROB Unit
Mod 341
*
D181245
7-6
PEC
DESCRIPTION
8310-001
AC Power Surge
Suppressor
TII Model 428
APPARATUS CODE
COMCODE
402988950
Miscellaneous
No PEC
Fanning Strip
50A
105196604
No PEC
Block, Connector
110 Female
103756334
No PEC
Block, Connector
157B Female
403613003
No PEC
Cable Conn (15’ DE)
A25D
100963990
No PEC
Block, Connector
103A
103104220
No PEC
Block, Connector
104A
103116943
No PEC
Panel (Part of SIP)
617A
103982658
No PEC
ll0 to Mod Adpter
858A
405177791
No PEC
Mod to Mod Adpter
Z210A1
103972907
No PEC
CPU/MEM
interconnect cable
*
845412956
* Not Available.
† System 75 PECs are listed. System 25 PECs may now be available.
Check before ordering.
7-7
Parts Listed by COMCODE
Throughout the following table, “part of” is abbreviated as “P/O.”
COMCODE PEC
100009968
P/O 62511
P/O 62514
DESCRIPTION
APPARATUS CODE
Auxiliary Trunk
Interface
10B ETU
Block Conn 66E3-25
(2) B25A Cables DE
100017334
P/O 62511
Auxiliary Trunk
Interface
Cable B25A 15/DE
100287085
31017
Ground Start Key
55A1
100963990
No PEC
Cable Corm (15’ DE)
A25D
102600517
P/O
P/O
P/O
P/O
ADU Aux Power
Peripherals Interface
MET Set-BIS
MOH Interface
2012D
2012D Transformer
2012D Transforrner
2012D Transformer
102802103
P/O 21691
ADU Aux Power
248B
102802113
P/O 62515
Peripherals Interface
248B Adapter
102937620
P/O 62509
Direct Extension
Selector Console
MERLIN VT
local power
Peripherals Interface
for Remote Access
ADU Aux Power
D6AP Cord
21691
62515
3141-BIS
62513
P/O 62510
P/o 62515
P/O 21691
D6AP-87 Cord
D6AP-87 Cord
D6AP-87
102999059
P/O 62512
Paging/Dictation
(Aux Trunk
Interface)
D4BY Cord (14ft)
103104220
No PEC
Block, Connector
103A
103116943
No PEC
Block, Connector
104A
103234472
3100-TWR
TT Wall Tel
2554BM
103556791
63130†
Data Line CP
TN726
103556882
63112†
MET Line CP
TN735
103556957
63111†
Analog Line CP
TN742
7 - 8
COMCODE
PEC
DESCRIPTION
APPARATUS CODE
103557062
63116†
DID Trunk CP
TN753
103557112
63119†
Pooled Modem CP
TN758
103557161
63118†
Aux Trunk CP
TN763
103558916
P/O 62513
MOH Interface
36A Voice Coupler
103756334
No PEC
Block, Connector
110 Female
103786786
P/O 62515
P/O 62514
2725-07G
Peripherals Interface
10B ETU
D8W-87 Cord (7 ft)
103786802
P/O 62509
Direct Extension
Selector Console
Asynchronous
Data Unit
D8W-87Cord (14 ft)
PO 062506
D8W-87Cord (14 ft)
2725-075
103786828
No PEC
Cord
D8W-87Cord (25 ft)
103810586
P/O 6250-011
P/O 6250-012
TDM Bus Term
J58901A1 L2,4
103814356
3163-HFU
MERLIN Speaker Mod
S102A
103823555
3100-ORW
Rtry Wall Tel
500MPA
103842050
3162-412
34-Button MERLIN VT
Z7305H01B-003
103843538
3162-417
34-Button Dlx MERLIN VT
Z7305H02B-003
103848859
P/O 21691
P/O 62509
ADU Aux Power
Direct Extension
Selector Console
400B
400B Adapter
103870267
3100-ORD
Rtry Desk Tel
500MM
103870416
3100-2RD
Rtry Desk Tel
(4A Spkphone Compatible)
500SM
103871018
3140-010
10-Button MET VT
2991C05
103871109
3170-00M
Single-Line VT w/Message
7101A01A-003
103871844
P/O 52512
Paging/Dictation
278A Adapter
7-9
COMCODE P E C
DESCRIPTION
APPARATUS CODE
103881421
2750-A17
MET Adapter Cord
ZD8AJ
103942146
3141-BIS
MET Set-BIS
2993C04
103942857
P/O 62510
MERLIN VT
local power
Z400F Adapter
103963310
3143-12M
12-Button MET VT
Z7203M01A-003
103963963
2169-001
Tip Ring ADU
Z3A1
103964185
2169-004
P/O 62515
Tip Ring ADU
Peripherals Interface
Z3A4
Z3A4 ADU
103965232
62502
GS Trunk CP
ZTN76
103965240
62503
LS Trunk CP
ZTN77
103965257
62504
TR Line CP
ZTN78
103965265
62505
ATL Line CP
ZTN79
103965281
P/O 6250-011
Memory CP
ZTN81
103965299
P/O 6250-011
Processor CP
ZTN82
103965315
62518
STARLAN Interface CP
ZTN84
103965323
P/O 6250-011
P/O 6250-012
Service Circuit CP
ZTN85
103966255
3178-SYS
Basic TT Desk Tel
w/Message and Recall
2500DMGC
103969424
P/O 62509
Direct Extension
Selector Console
23A1-003
103972907
No PEC
Mod to Mod Adpter
Z21OA1
103975349
P/O 62506
Asynchronous
Data Unit
Z3A5 ATL ADU
103975645
63117†
Tie Trunk CP
TN760B
103976163
63123†
Tone Detector CP
TN748
103981965
3162-BIS
MERLIN BIS VT
Z7305H03B-003
7-10
APPARATUS CODE
COMCODE
PEC
DESCRIPTION
103981981
3162-DIS
Z7305H04C-003
103982005
3161-161
MERLIN HFAI VT
Z7309H01B-003
103982658
No PEC
Panel (Part of SIP)
617A
103982740
P/O 6250-011
Memory CP
ZTN81B
103984118
62514
10B ETU
Emergency Trans Unit
104010061
31710
Acoustic Coupler
349A Adapter
104152558
P/O 62515
Peripherals Interface
400B2 Adapter
104154430
2724-38X
P/O 62515
Mod (ADU) X-Over Cord
Peripherals Interface
D8AM-87
D8AM-87Cord
104174750
31761
Power Panel
346A-1
104174768
31760
Power Unit
346A
104246616
2724-30C
P/O 62515
RS232 X-Over Cable
M7U-87
105012637
2750-A24
Mod/RS232 Adapt, Male
355A
105012645
2750-A25
P/O 62515
Mod/RS232 Adapt, Female
355AF
105031181
P/O 62512
Paging/Dictation
D181321 Kit (zener)
105105506
P/O 62506
Asynchronous
D181521
105105514
P/O 62510
MERLIN VT
D181522
105105522
P/O 62511
Auxiliary Trunk
D181523
105105530
P/O 62512
Paging/Dictation
D181524
105291488
P/O 6250-011 Memory CP
ZTN81C
105196604
No PEC Fanning Strip
50A
105211023
P/O 6250-011 Processor CP
P/O 6250-012
ZTN128
105212179
P/O 6250-011 Memory CP
P/O 6250-012
ZTN127
7-11
COMCODE
PEC
DESCRIPTION
APPARATUS
105335657
62519
RlV2 Upgrade Kit
D181782
105339584
1203-020
AA Software
*
105341218
1020-S90
ACCESS Software
*
105341382
1020-S91
CAM Software
*
105355374
62520
STARLAN/ATL Interface
D181807
402988950
8310-001
AC Power Surge
Suppressor
*
403242639
P/O 62509
KS22911 L1 Power
P/O 62512
P/O 62520
Direct Extension
Selector Console
MERLIN VT
local power
Paging/Dictation
STARLAN/ATL Interface
403613003
No PEC
Block, Connector
157B Female
403836620
2720-06X
Splitter CBL, CO Trunk
WP9092,L1
403864150
2720-05X
Splitter CBL, Tie-Trunk
WP90929,L3
403954761
P/O 6250-011
P/O 6250-012
P/O 62501
Power Supply
WP9051O
403957129
P/O 6250-011
P/O 6250-012
P/O 62501
Air Filter
21985-1
403961519
P/O 62501
TDM Bus Cable
J58901A1 L3
404079428
62508
System Admin
TS-458A
404079436
62507
Digital Tape Unit
TS-555A
405010612
2720-05P
25-pair/8-plug 15 ft CBL
WP90780L1
405010620
2750-T05
Voice/Data (Y) Adapter
WP90851L1
405177791
No PEC
ll0 to Mod Adpter
858A
405193186
P/O 62513
MOH Interface
KS23395 Interface
P/O 62510
7-12
CODE
KS22911 L1 Power
KS22911 L1 Pwr Unit
KS22911 L1 Power
COMCODE PEC
DESCRIPTION
APPARATUS CODE
KS-23475,L1
405462904
P/O 62520
STARLAN/ATL Interface
Adapter
527840003
2610-001
AT&T STARLAN NETWORK
Network Extension Unit (NEU)
527840102
2614-100
AT&T STARLAN NETWORK
Network Access Unit (NAU)
845412956
No PEC
CPU/MEM
Interconnect Cable
*
845416379
P/O 6250-011
P/O 6250-012
P/O 62501
Fan (2)
WP90677L1
845416635
P/O 6250-011
P/O 6250-012
P/O 62501
Address Plug
845875155
P/O 6250-011
P/O 6250-012
SLAC Grap. Layer (2)
* Not Available.
† System 75 PECs are listed. System 25 PECs may now be available.
Check before ordering.
7-13
SECTION 8—REFERENCE DOCUMENTATION
System 25 is supported by a complete set of basic and supplementary documentation and
optional software. This section provides a brief summary of the available material. Manuals
not specified for Release 1 Version 2 (R1V2) cover both R1V1 and R1V2.
Basic Manuals
Administration Manual . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Administration Manual for R1V2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Implementation Manual . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Implementation Manual for R1V2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Installation and Test Manual . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Introduction to AT&T System 25 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Maintenance Manual . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
New Capabilities Manual for R1V2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Reference Manual . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Terminal Operations Manual . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
User Guides
- Data Features User Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
- Direct Trunk Attendant Console User Guide . . . . . . . .
- Multiline Terminal User Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
- Single-Line Terminal User Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . .
- Switched Loop Attendant Console User Guide . . . . .
- Multifeature Single-Line Terminal User Guide . . . .
555-500-500
555-520-500
555-500-662
555-520-650
555-520-100
555-520-021
555-520-105
555-5.20-205
555-520-200
555-520-710
555-520-704
555-520-701
555-520-703
555-520-702
555-520-706
555-520-707
Supplementary Material
Application Notes Binder . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
R1V2 Upgrade Superpac . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
AT&T Call Accounting System
- User’s Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
- Implementation Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
AT&T Model 703 SAT Supplement . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Customer Education Leader Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Customer Education Student Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Customer Training Superpac . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
System 25–Product Brochure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
System 25–Slim Jim Brochure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
System 25–Switched Loop Attendant
Console Sales Literature . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
System 25–AT&T STARLAN NETWORK Access
Feature Sales Literature . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
System 25–The Integrated Solution
Sales Literature . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
420 Speakerphone Sales Literature . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
8-1
555-509-002
555-520-013
555-006-201
999-500-247
555-500-720
555-520-016
555-520-014
555-520-011
PM-4410
PM-4409
PM-4410-F
PM-4410-H
PM-4410-E
PM-4410-G
Optional Software
Advanced Administration Software Package
Communications Access Manager Software Package
● AT&T STARLAN NETW0RK ACCESS Software Package
●
●
Descriptions of Basic Manuals
Administration
Manual
Provides the information necessary to initialize a system and to perform on-going system
administration. Explains the operation of the System Administration Terminal, the Digital
Tape Unit, and the commands that allow the System Administrator to make changes and
additions.
Implementation Manual
Describes how to plan the operating configuration of the system. Explains how to determine
customer needs and how to convert these needs into a system configuration plan. This plan
is recorded on accompanying forms that are used in conjunction with the Administration
Manual to initialize the system. The Implementation Manual and associated forms are
packaged together in the Administration Records Binder.
Installation And Test Manual
Provides step-by-step procedures for installing System 25 and associated equipment.
Includes procedures for testing equipment and trunks and for making additions and changes
to the system.
Introduction to AT&T System 25
Provides an introduction to System 25 features and services. The emphasis is on how
System 25 helps solve information management, productivity, and cost control problems.
Maintenance Manual
Provides a detailed description of system operation and procedures for isolating and clearing
customer affecting faults.
New Capabilities Manual
This manual describes the System 25 Release 1 Version 2 (R1V2) features that were not
included in Release 1 Version 1 (R1V1). In addition, R1V1 features that have been enhanced
for R1V2 and new R1V2 hardware not included in R1V1 are included.
8-2
Planning Manual
The document is a presale workbook used by the Account Team and customer to define a set
of orderable equipment that will meet the customer’s specific requirements. This workbook,
when completed, can be used in conjunction with the Quick-Quote Price Estimator or the
DOSS Configurator to obtain a price estimate or formal quote or to place an order.
Reference Manual
This document is the principal technical reference of the system. It provides reference
material for sales support, s y s t e m c o n f i g u r a t i o n a n d o p e r a t i o n a n d f o r t h e s y s t e m
technician. It contains a comprehensive description of the system, emphasizing features,
Components and overall capabilities and capacities.
Terminal Operations Manual
This manual is designed to help the System Administrator better understand System 25
voice terminal and data terminal operation. This manual contains the operating procedures
provided in the various User Guides and provides additional explanatory material as well.
User Guides (700 Series)
These guides contain step-by-step operating procedures for System 25 attendants and voice
and data terminals users.
8-3
SECTION 9—GLOSSARY
This section provides explanations for acronyms and definitions of terms used in this
manual.
ADU: (Asynchronous Data Unit)
ARS: (Automatic Route Selection)
ASCII: (American Standard Code for Information Exchange)
Administer
To access and change the parameters associated with the services or features of the
system.
Analog Data Endpoint
Data endpoints with customer provided built-in or stand-alone modems. They do not
require the use of data modules (asynchronous data units) and are addressed similar
to any voice terminal by PDC. These end-points connect to tip/ring type circuit pack
ports.
Analog Voice Terminals
Voice terminals served by a single-line tip and ring circuit (2500 series and 7101A
voice terminals or industry standard Dual Tone Multifrequency equivalent).
Appearance
See Call Appearance.
Asynchronous Data Transmission
A scheme for transmitting data where each character is preceded by a start bit and
followed by a stop bit, thus permitting data elements to occur at irregular intervals.
This type transmission is advantageous when transmission is not regular (characters
typed at a keyboard).
Asynchronous Data Unit (ADU)
A data communications equipment (DCE) type device that allows direct connection
between RS-232C equipment and the digital switch via ports on the Data Line Circuit
Pack (TN-726).
Attendant
The operator of the attendant console.
9-1
Attendant Console
●
Direct Trunk or Switched Loop Attendant Console: An electronic
call-handling position with pushbutton control. Used by attendants to
answer and place calls and to manage and monitor some system operations.
●
Direct Extension Selector Console: Provides the attendant with a visual
indication of the active or idle status of extension numbers assigned in the
system. Also allows the attendant to place calls to system users by operation
of appropriate H u n d r e d s G r o u p S e l e c t b u t t o n s a n d a s s o c i a t e d D i r e c t
Extension Selection (DXS) buttons.
Auxiliary Equipment
●
Dictation Equipment
●
Delay Announcement Devices
●
External Alerting Devices (external alerts)
●
Music-On-Hold Equipment
●
Paging Equipment
Auxiliary Trunk
A trunk circuit used to connect auxiliary equipment to the switch, for example, music
or dictation equipment.
BLF: (Busy Lamp Field)
BPS: (Bits Per Second)
Bit (Binary Digit)
One unit of information in binary notation (having two possible states or values, zero
or one).
Bridge (Bridging)
The sharing of the same extension by two or more voice terminals.
Buffer
A circuit or component that isolates one electrical circuit from another. Typically, a
buffer holds data from one circuit or process until another circuit or process is ready
to accept the data.
Bus
A multi-conductor electrical path used to transfer information over a common
connection from any of several sources to any of several destinations.
9-2
Bus, Time Division Multiplex
See Time Division Multiplex Bus.
CCS (Hundred Call Seconds)
A traffic-measuring unit that expresses the load of one or more traffic-handling
devices. A device used for 1 hour without interruption generates 36 CCS which
equals 1 erlang (see Erlang).
Call Appearance, Attendant Console
Two buttons, labeled System Access, used to originate, receive, and hold calls. Each
button has two associated LEDs to show the status of the call appearance.
Call Appearance, Voice Terminal
A button labeled with an extension number used to place outgoing calls, receive
incoming calls, or hold calls. Two LEDs next to the button show the status of the
call appearance or status of the call.
Central Office
The location housing telephone switching equipment that provides local telephone
service and access to toll facilities for long-distance calling.
Central Office Codes
The first three digits of a 7-digit public network telephone number. These codes are
numbered from 200 through 999 and are sometimes referred to as NNXs.
Central Office Trunk
A telecommunications channel that provides access from the system to the public
network through the local central office.
Channel
A communications path for transmitting voice and data.
Class of Service (COS)
Parameters used to define voice terminal, data, and trunk port capabilities and
restrictions.
Common Control Switching Arrangement (CCSA)
A private telecommunications network using dedicated trunks and a shared
switching center for interconnecting company locations.
Confirmation Tone
Three short bursts of tone followed by silence; indicates that the feature activated,
deactivated, or canceled has been accepted.
9-3
Console
See Attendant Console.
Coverage Call
A call that is redirected from the called party’s personal dial code to an alternate
answering position when certain criteria are met.
Coverage Path
The order in which calls are redirected to alternate answering positions.
Coverage Point
The attendant positions (as a group), Direct Group Call (DGC) group, Coverage
Answer Group, or a voice terminal extension designated as an alternate answering
position in a coverage path.
Covering User
The person at an alternate answering position who answers a redirected call.
DCE: (Data Communications Equipment)
DDC: (Data Dial Code)
DDD: (Direct Distance Dialing)
DID: (Direct Inward Dialing)
DGC: (Direct Group Call)
DTE: (Data Terminal Equipment)
DTU: (Digital Tape Unit)
DTMF: (Dual Tone Multifrequency)
DXS: (Direct Extension Selector)
Data Channel
A communication path between two points used to transmit digital signals.
Data Communications Equipment (DCE)
Refers to a specific RS-232C interface connector configuration. DCE devices are
designed to interface directly (pin-for-pin) to Data Terminal Equipment (DTE). The
transmit and receive pins are reversed from that of a DTE interface. A modem is an
example of a DCE device.
Data End Point
Two general groups; those having a DTE type interface, which encompasses almost
all of the data terminal devices, and the group of DCE interface devices which are
primarily modems. However, it must be noted that within each category, control
interfaces may also vary. Refer to Analog Data Endpoint and Digital Data
Endpoints for additional information.
9-4
Data Module
A data interface device (i.e., Asynchronous Data Unit) providing a standard interface
between customer provided data equipment and a data port on the switch.
Data Terminal Equipment (DTE)
DTE refers to a specific RS-232C connector termination designed to connect directly
to a DCE type connection. Typically associated with video display terminals,
printers, and computers which either originate or terminate a data transmission
path.
Refers to RS-232C compatible Data Terminal Equipment
Delay-Dial Tie Trunk
After a request for service (called a seizure) is detected on an incoming trunk, the
system sends a momentary signal followed by a steady tone over the trunk. This
informs the calling party that dialing can start. This type of trunk allows dialing
directly into the system. That is, the digits are received as they are dialed.
Digital Data Endpoints
In System 25, digital data endpoints include any digital device providing a RS-232C
connection interface to the switch. The connection is via Asynchronous Data Units
(ADUs) to the switch.
Direct Extension Selector (DXS) Console
An option at the attendant console that allows an attendant direct access to voice
terminals by pressing a Group Select button and a DXS button.
EIA: (Electronics Industries Association)
Emergency Transfer Unit
Provides direct connection of designated Power Failure Transfer (PFT) registered
voice terminals to the CO during a power failure or other service interruption.
Erlang
A traffic measuring unit that expresses the load of one or more traffic-handling
devices [36 CCS equals 1 erlang - see CCS (Hundred Call Seconds)].
Extension Number
One- through four-digit number assigned to each voice terminal and data end point
in the system. Also see “ Personal Dial Code”
External Call
A connection between a system user and a party on the public telephone network or
on a tie trunk.
FRL: (Facility Restriction Level)
9-5
Facility (physical)
A transmission channel to another switching system; to a Central Office for example.
By application, examples are:
●
CO Trunks
●
FX Trunks
●
WATS
●
OCC Trunks
●
Tie Trunks
Trunks
By technical type these include loop start, ground start, DID, automatic ringdown,
etc. These facilities may be accessed by their facility access codes (FACs).
Feature
A specifically defined function or service provided by the system.
Feature Button
A labeled button on a voice terminal or attendant console designating a specific
feature.
Foreign Exchange (FX)
A central office other than the one providing local access to the public telephone
network.
Foreign Exchange Trunk
A telecommunications channel that directly connects the system to a central office
other than its local central office.
Foreign Numbering Plan Area Code (FNPA)
An area code other than the local area code. The foreign area code must be dialed to
call outside the local geographical area.
Ground-Start Trunk
On outgoing calls, System 25 transmits a request for services to the distant switching
system by grounding the trunk ring lead. When the distant system is ready to
receive the digits of the called number, that system grounds the trunk tip lead.
When the System 25 detects this ground, the digits are sent. (Tip and ring are
common nomenclature to differentiate between ground-start trunk leads.) On
incoming calls, detection of ground on the tip lead is sufficient to cause the call to
route to a predetermined destination, normally the system attendant group. No
digits are received.
9-6
Home Numbering Plan Area Code (HNPA)
The local area code. The area code does not have to be dialed to call numbers within
the local geographical area.
Immediate-Start Tie Trunk
After establishing a connection with the distant switching system for an outgoing
call, the system waits a nominal 65 milliseconds before sending the digits of the
called number. This allows time for the distant system to prepare to receive the
digits. Similarly, on an incoming call, the system has less than 65 milliseconds to
prepare to receive the digits.
In-Use Lamp
A red lamp on a multiline voice terminal that lights to show which call appearance
will be selected when the handset is lifted or which call appearance is active when a
user is off-hook.
Intercept Tone
An alternating high and low tone; indicates a dialing error or denial of the service
requested.
Interface
A common boundary between two systems or pieces of equipment.
Internal Call
A connection between two users within the system.
LDN: (Listed Directory Number)
LED: (Light Emitting Diode)
Loop Start Trunk
After establishing a connection with the distant switching system for an outgoing
call, System 25 waits for a short period of time before sending the digits of the called
number. On incoming calls, the received request for service is sufficient to cause the
call to route to a predetermined destination, normally the system attendant group.
No digits are received.
MET: (Multibutton Electronic Telephone)
Modem
A device that modulates and demodulates signals transmitted over a communications
path. Used to connect Data Terminal Equipment to the system’s analog ports. The
system provides a pooled modem conversion resource (12 resources maximum per
system—212A compatible).
9-7
Modem Pooling
Provides shared-use conversion resources that eliminate the need for a dedicated
modem when an analog data end point accesses, or is accessed by, an analog line or
trunk.
Multifeature Single-Line Voice Terminal
A terminal served by one tip and ring voice circuit and having additional buttons for
activating features.
Multilane Voice Terminal
A terminal equipped with several call appearance buttons for the same extension
number. Allows the user to handle more than one call, on that same extension
number, at the same time.
Multiplexed
The simultaneous transmission of two or more signals over a common transmission
medium.
NPA: (Number Plan Area)
Network
An arrangement of inter and/or intra location circuits designed to perform specific
functions.
Network Interface
Provided by the CO telephone company in two forms:
(l) RJ21X for trunk facilities other than tie trunks.
(2) RJ2GX for tie trunk facilities.
An arrangement provided by the local telephone company which permits remote
Terminal Equipment to operate as though it was directly connected to the System 25.
This tariffed service can only be provided for FCC registered single-line voice
terminals.
Out-Of-Building Station
The Terminal Equipment indirectly connected to the System 25, but is not located in
the same building as the common equipment. Special arrangements are made to
protect the system and its users from lightning, power line crosses, etc. Only the
single-line and 7300H series of voice terminal may be so connected. MET Sets can
not be connected as Out-Of-Building stations.
PDC: (Personal Dial Code)
Paging Trunk
A telecommunications channel used to access an amplifier for loudspeaker paging.
9-8
Parameter
Any set of physical properties whose values determine the characteristics or
behavior of something
Peripheral Equipment
System Administration Terminal (SAT), SMDR Output device such as a SMDR
Printer or a Call Accounting System, Digital Tape Unit (DTU).
Personal Dial Code
Each system user is assigned a PDC and is allowed to “ login” the PDC at any voice
terminal in the system (optional feature) as they move about the premises. The PDC
may be a 1-, 2-, 3-, or 4-digit number. There are two types of PDCs:
●
PDCs assigned to voice terminals - Associated with each voice terminal in the
system.
●
Floating - Assigned to users and visitors who will be moving about the
premises. Floating PDCs may be associated with the attendant position or
may be “logged-in” by the user at a system voice terminal. Calls to the
floating PDC will ring at the terminal where “logged in”
Pickup Group
A group of individuals authorized to answer any call directed to an extension number
within the group.
Port
An interface circuit between System 25 and associated auxiliary and peripheral
equipment. Typical references include:
●
Terminal port (station port)
●
Facility port (trunk port)
●
Auxiliary
equipment
port
Private Branch Exchange (PBX)
A switching system that provides switched communications access amongst its
terminals and facilities (e.g., System 25)
Private Network
A network used exclusively for handling the telecommunications needs of a
particular customer.
Private Network Office Code (RNX)
The first three digits of a 7-digit private network number. These codes are
numbered 220 through 999, excluding any codes that have a O or 1 as the second digit.
9-9
Protocol
A set of conventions or rules governing the format and timing of message exchanges
to control data movement and correction of errors.
Public Network
The network that can be openly accessed by all customers for local or long-distance
calling.
Queue
An ordered sequence of calls waiting to be processed.
Queuing
The process of holding calls in order of their arrival to await connection to an
attendant, to an answering group, or to an idle trunk. Calls are automatically
connected in first-in, first-out sequence.
Random Access Memory (RAM)
A storage arrangement whereby information can be retrieved at a speed independent
of the location of the stored information.
Read Only Memory (ROM)
A storage arrangement primarily for information retrieval applications.
Recall Dial Tone
Three short bursts of tone followed by steady dial tone; indicates the system has
completed some action (such as holding a call) and is ready to accept dialing.
Redirection Criteria
The information administered for each voice terminal that determines when an
incoming call is redirected to coverage.
Reorder Tone
A fast-busy tone repeated 120 times a minute; indicates that at least one of the
facilities, such as a trunk or a digit transmitter, required for the call was not
available at the time the call was placed.
Single-Line Voice Terminal
Voice terminal served by a single-line tip and ring circuit (2500 series and 7101A
voice terminals or industry standard Dual Tone Multifrequency equivalent).
SAT: (System Administration Terminal)
SIP: (Station Interconnect Panel)
SMDR: (Station Message Detail Recording)
9-10
Software
A set of computer programs that accomplish one or more tasks.
Split
A condition whereby a caller is temporarily separated from a connection with the
attendant. This split condition automatically occurs when the attendant, active on a
call, presses the Start button.
Status LED (lamp)
A green LED or lamp that shows the status of a call appearance or a feature button
by the state of the lamp (lighted, flashing, fluttering, broken flutter, or dark).
Station
A place where terminal equipment is located or sometimes the terminal equipment
itself. Each voice terminal (station) is assigned a station (extension) number. Users
of the terminal are sometimes referred to as station users. Reference to the
extension number is sometimes in the form PDC (Personal Dial Code) rather than
station number. Though PDCs may be “logged-in” at other stations, in most
discussions, though, PDCs and station numbers are interchangeable. Analogously,
data stations are assigned DDCs (Data Dial Codes)
Switch
The software-controlled communications processor complex that interprets
pulses/tones/keyboard characters and makes the proper interconnections both
the system and external to the system. The switch itself consists of a
computer, software, storage device (memory), and associated circuit packs and
hardware necessary to perform the actual connections.
dialing
within
digital
special
Switchhook
The button(s) on a voice terminal located under the receiver.
System Manager
A person responsible for specifying and administering features and services for the
system.
System Reload
A process that allows stored data to be written from a tape into the system memory
(normally after a power outage).
TAE: (Trunk Access Equipment)
Terminal Equipment:
Equipment for changing information (sound, keystrokes) into an electrical signal
compatible with the system’s port circuits (voice and data terminals are two
subdivisions).
9-11
Tie Trunk
A telecommunications channel that directly connects two private switching systems.
Time Division Multiplex Bus
A special bus that is time shared by preallocating short time slots to each
transmitter on a regular basis. In a PBX, all port circuits are connected to the time
division multiplex bus permitting any port to send a signal to any other port.
Tone Ringer
A device with a speaker, used in electronic voice terminals to alert the user.
Translations
Specific information assigned to a terminal or to the system and customized for the
user.
Trunk
A telecommunications channel between two switching systems.
Trunk Group
Telecommunications channels assigned as a group for certain functions.
Trunk Port
The hardware providing the access point to the system switching network for each
circuit associated with a trunk.
Voice Terminal
A single-line or multiline voice instrument (e.g., telephone)
Voice Terminal Adjuncts
. 500A/502A Headset Adapter
. S101A/S102A Speakerphone
. Acoustic Coupler
(Refer to Section 4–Hardware Description for a complete list and description
of Voice Terminal Adjuncts. )
Wide Area Telecommunications Service (WATS)
A service that allows calls to a certain area or areas for a flat-rate charge based on
expected usage.
Wink-Start Tie Trunk
After establishing a connection with a distant switching system for an outgoing call,
the system waits for a momentary signal (wink) before sending the digits of the
called number. Similarly, on an incoming call, the system sends the wink signal
when ready to receive digits.
9-12
Write Operation
The processor putting information onto a storage medium such as magnetic tape.
800 Service
A service that allows incoming calls from a certain area or areas to an assigned
number for a flat-rate charge based on usage.
9-13
INDEX
Auxiliary Equipment (Contd.)
Connections, 4-51
Auxiliary Trunk (TN763), 3-35
500 Voice Terminals, 4-12
55A1 Key, 4-40
Abbreviated Alerting, 2-132
AC Power Requirements, 6-6
Access Equipment, Trunk, 4-46
Access, Dictation System, 2–118
Account Code Entry, 2-6
Accountability, Call, 2-61
Accounting, Call, 2-62
Acoustic Coupler, 4-40
Activating the Third-Party Call Setup Feature, 2-136
Adjunct, Speakerphone, 2-207
Administration
Interface, Remote, 2-199
Manual, 8-2
System, 2-225
Administrative Software, 3-45
ADU Connections, 4-51
Information, 4-43
Air Purity, 6-5
Alerting, Abbreviated, 2-132
Alerts, External, 2-139
Analog Line (TN742), 3-28
Analog Transmission Characteristics, 5-17
Announcements, Night Service Delay, 2-173
Answer-Record, 4-40
Answering a Call, 2-21
ARS Restriction, 2-93
Asynchronous Data Units (ADUs), 4-42
AT&T STARLAN NETWORK Access (V2), 2-215
ATL
Adapters, 2-221
Line (ZTN79), 3-22
Attendant
Call Extending, 2-8
Camp-on, 2-10
Cancel, 2-12
Console Connections, 4-51
Console, Direct Trunk, 2-13
Console, Switched Loop, 2-17
Direct Extension Selection, 2-25
Display, 2-29
Features, 2-5
Forced Release, 2-34
Join , 2-35
Message Waiting, 2-36, 2-162
Position Busy, 2-37
Release, 2-40
Return Coverage On Busy, 2-42
Return Coverage On Don’t Answer, 2-44
Source and Destination, 2-46
Splitting One-Way Automatic, 2-47
System Alarm Indication, 2-48
Automatic
Hold, 2-40
Intercom, 2-49
Maintenance Tests, 2-235
Route Selection, 2-51
Auxiliary Equipment, 4-45
Block Diagram of System 25, 1-3
Business Communications Needs, 1-4
Busy-to-Idle Reminder, 2-60
Cabinet 1 (Control and Port Circuits), 4-4
Cabinet Address Plug, 4-4
Cabinets 2 and 3 (Port Circuits), 4-4
Cable Distance Limitations, 5-6
Call
Accountability, 2-61
Accounting, 2-62
Accounting System (CAS), 2-65
Coverage Message Waiting, 2-162
Coverage—Group, 2-77
Coverage—Individual, 2-82
Extending, 2-8
Following (Forwarding), 2-84
Handling Capabilities, 1-4
Origination Interactions, 2-23
Park, 2-86
Pickup, 2-89
Processor Circuit Pack (ZTN82 or ZTN128), 3-2
Progress Tones, 2-91
Setup, Third-Party (V2), 2-239
Types, 2-22
Waiting, 2-10
Calling Restrictions, 2-92
Calls from
STARLAN NETWORK to System 25, 2-218
System 25 to the STARLAN NETWORK, 2-218
CAM, 2-99
Camp-on, Attendant, 2-10
Cancel, Attendant, 2-12
Capabilities, Call Handling, 1-4
CAS, 2-65
Circuit Pack
Address Leads, 3-15
Compatibility, 4-9
Features, 4-10
Circuit Packs, 4-7
Code Entry, Account, 2-6
Command Mode, 2-95
Common Control, 3-2
Communications Access Manager Program, 2-99
Conference, 2-101
Drop, 2-103
Connection Information,
ADU, 4-43
Multiline Voice Terminal, 4-40
SAT, 2-227
Single-Line Voice Terminal, 4-18
Voice Terminal Adjunct, 4-41
Connections,
Attendant Console, 4-51
Auxiliary Equipment, 4-51
10-1
Error Log, 2-235
EX RS-232 Driver, 2-100
Exclusion, 2-134
Expert Mode (V2), 2-135
Extended Stations, 2-138
Extending, Attendant Call, 2-8
External Alerts, 2-139
Connections, (Contd.)
Peripheral Equipment, 4-51
Voice Terminal and Adjuncts, 4-51
Connectivity, 4-46
Figures, 4-51
Console,
Direct Trunk Attendant, 2-13
Selector, 2-25
Switched Loop Attendant, 2–17
Consoles, Dual Attendant, 2-27
Coupler, Acoustic, 4-40
Facilities,
Recommended Central Office Trunk, 5-16
Virtual (V2), 2-258
Facility Access Restriction, 2-92
Features
and Services, 2-1
Attendant, 2-5
Data, 2-3
Network, 2-3
Station, 2-4
System, 2-2
Flex DSS, 2-130
Floor Plans And Layouts, 6-1
Forced Release, 2-34
Functional Description, 3-1
Data
Call Preindication, 2-247
Call Setup, 2-105
Features, 2-3
Line (TN726), 3-24
Services Overview, 2-106
Terminal Dialing, 2-112
Transfer to, 2-247
Delayed Access, 2-167
Description,
Functional, 3-1
System 25, 1-2
Destination, 2-46
Dial
Access, Pooled Facility, 2-186
Code, Personal, 2-181
Plan, 2-116
Dialing
Direct Inward, 2-127
Data Terminal, 2-112
Repertory, 2-200
Dictation System Access, 2-118
DID Trunk (TN753), 3-30
Digital
Switch, 3-1
Tape Unit, 2-120
Direct
Access, Pooled Facility, 2-187
Extension Selection, Attendant, 2-25
Extension Selector Console, 2-19
Group Calling (DGC), 2-122
Group Calling Delay Announcement, 2-125
Inward Dialing (DID), 2-127
Station Selection (DSS), 2-130
Trunk Attendant Console, 2-13
Display, Attendant, 2-29
Distinctive Ringing, 2-132
DSS, 2-130
Direct Station Selection, 2-l30
DTAC, 2-13
DTU, 2-120
Dual Attendant Selector Consoles, 2-27
DXS, 2-25
Glossary, 9-1
Ground Start
Button, 4-40
Trunk (ZTN76), 3-16
Grounding, 6-8
Group Calling (DGC), Direct, 2-122
Hands-Free Answer on Intercom (HFAI), 2-142
Hardware
and Software Parameters, 5-2
Description, 4-1
Headset Adapter Adjunct, 2-145
Hold, 2-150
Hunting, Station, 2-222
Implementation Manual, 8-2
Indications, Line Status And I-Use, 2-157
Indicator Lamp Signals, 5-9
Individual, Call Coverage—, 2-82
Installation And Test Manual, 8-2
Intercept Treatment With Reorder Tone, 2-151
Intercom
(HFAI), Hands-Free Answer On, 2-142
Automatic, 2-49
Interdigit Timeouts, 2-152
Introduction to AT&T System 25, 8-2
Join, 2-35
Electrical
Characteristics, 3-7
Noise (RFI), 6-5
End-To-End Signaling, 2-133
Environmental Requirements, 6-1
Equipment Configuration, Typical, 1-3
Last Number Dialed, 2-135
Lighting, 6-5
Lightning Protection, 6-8
Line
Selection, 2-155
10-2
Pooled Facility (Contd.)
- Direct Access, 2-187
Pooled Modem (TN758), 3-43
Pooling, Modem, 2-164
Port
Circuits, 3-10
Specifications, 5-l0
Position Busy, Attendant, 2-37
Power Failure Transfer (PFT), 2-189
Preference,
Prime Line, 2-155
Ringing Line, 2-155
Preindication, Data Call, 2-247
Preselection, 2-156
Prime Line Preference, 2-155
Program, 2-194
Programmable Feature Buttons, 2-18
Line (Contd.)
Status and I-Use Indications, 2-157
Lines, Personal, 2-184
Loop Start Trunk (ZTN77), 3-18
Maintenance
Manual, 8-2
Software, 3-45
System, 2-234
Making a Data Call From Expert Mode, 2-135
Manual Signaling, 2-159
Memory
Allocation, 3-45
Circuit Pack (ZTN81 or ZTN127), 3-5
Message Center Administration, 2-161
Message Center-Like Operation, 2-160
Message Waiting,
Attendant, 2-36
Station-To-Station, 2-224
Messaging Services, 2-162
MET Line (TN735), 3-26
Mode, Command, 2-95
Model 500 Voice Terminals, 4-12
Modem Pooling, 2-164
Multiline Voice Terminals, 4-19, 5-7
Connection Information, 4-40
Music-On-Ho1d, 2-167
Music Source (Music-On-Hold), 2-168
R1V1, 1-1
R1V.2, 1-1
Real-Time Constraints, 3-45
Recall, 2-198
Recommended Central Office Trunk Facilities, 5-16
Recording, Station Message Detail, 2-62
Reference
Documentation, 8-1
Manual, 8-3
Release 1
Version 1, 1-1
Version 2, 1-1
Release, Attendant, 2-40
Reminder, Busy-to-Idle, 2-60
Remote Administration Interface, 2-199
Reorder Tone, Intercept Treatment With, 2-151
Repertory Dialing, 2-200
Restrictions, Calling, 2-92
Return Coverage
On Busy, Attendant, 2-42
On Don’t Answer, Attendant, 2-44
Ringing Line Preference, 2-155
Ringing, Distinctive, 2-132
Rotary Telephone, 4-12
Route Selection, Automatic, 2-51
NAUCOM Driver, 2-100
Network Features, 2-3
New Capabilities Manual, 8-2
Night Service, 2-170
Delay Announcements, 2-173
NPEs (Network Processing Element), 3-14
Off-Premises Stations (OPS), 2-175
One-Button-Transfer To Data, 2-247
Options, User Changeable
V2, 2-253
in Expert Mode, 2-136
Organization, 1-1
Out-of-Building Stations, 2-176
Outward Restriction, 2-92
Overview, 1-1
Sl01A/S102A Speakerphone, 2-207
Safety, 1-4
SAT Connection Information, 2-227
Selector Console, 2-25
Send All Calls (V2), 2-202, 2-204
Service Circuit (ZTN85 ), 3-38
Services, Messaging, 2-162
Setup, Data Call, 2-105
Signaling,
End-To-End, 2-l33
Manual, 2-159
Single-Line Voice Terminals, 4-12, 5-6
Connection Information, 4-18
SLAC, 2-17
SMDR, 2-62
Software, 3-45
Partitioning, 3-46
Source, 2-46
Paging
Equipment, 2-178
System Access, 2-177
Parts Information, 7-1
Peripheral Equipment, 4-45
Connections, 4-51
Personal
Dial Code (PDC), 2-181
Lines, 2-184
Speed Dialing, 2-213
Physical Characteristics, 3-7
Placing a Call, 2-21
Planning Manual, 8-3
Pooled Facility
- Dial Access, 2-186
10-3
Speaker, 2-206
Speakerphone Adjunct, 2-207
Speed Dialing, 2-212
Splitting One-Way Automatic, Attendant, 2-47
STARLAN INTERFACE
(ZTN84), 3-37
Circuit Pack, 2-216
STARLAN NETWORK
Access (V2), 2-215
Administrable Parameters, 2-221
Station
Features, 2-4
Hunting, 2-222
Interconnect Panel (SIP), 4-48
Message Detail Recording (SMDR), 2-62
Station-To-Station Message Waiting, 2-163, 2-224
Stations,
Extended, 2-138
Off-Premises, 2-175
Out-of-Building, 2-176
Switch, Digital, 1-1
Switched Loop
Attendant Console, 2-17
Operation, 2-20
Switched Services Software, 3-45
Switching Network, 3-6
System 25
Block Diagram, 1-3
Description, 1-2
System
Access, Paging, 2-177
Administration, 2-225
Administration Terminal, 2-226
Cabinets (J58901Al), 4-1
Errors And Alarms, 2-234
Features, 2-2
Maintenance, 2-234
Resource Circuit Packs, 4-7
Resources, 3-38
Speed Dialing, 2-212
Transfer (Contd.)
Button, Use Not Recommended, 2-8
to Data, 2-247
Trunk Access Equipment (TAE), 4-46
Trunk-To-Trunk Transfer, 2-252
Trunking, Tandem (V2), 2-236
Trunks, Tie, 2-242
Typical Equipment Configuration, 1-3
Unit Loads, 5-5
User Changeable Options
V2, 2-253
in Expert Mode, 2-136
User Guides (700 Series), 8-3
Virtual Facilities (V2), 2-258
Voice Terminal Adjuncts, 4-40
Connection Information, 4-41
Power Supplies, 4-41
Voice Terminals, 4-12
and Adjunct Connections 4-51
Waiting Call, 2-10
Wall Space Requirements, 6-4
Z34A Message Waiting Indicator, 4-40
Table Top Space, 6-4
Tandem Trunking (V2), 2-236
Tape Unit, Digital, 2-120
TDM BUS , 3-6
Technical Specifications, 5-1
Temperature and Humidity, 6-4
Terminal
Dialing for Voice (V2), 2-239
Equipment, 4-12
Operations Manual, 8-3
Test, 2-238
Third-Party Call Setup (V2), 2-239
Tie Trunk (TN760B), 3-32
Tie Trunks, 2-242
Timeouts, Interdigit, 2-152
Tip Ring Line (ZTN78), 3-20
Toll Restriction, 2-92
Tone Detector (TN748), 3-41
Tones, 5-9
Call Progress, 2-91
Touch-Tone and Dial Pulse Services, 2-244
Transfer, 2-245
(PFT), Power Failure, 2-189
10-4